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[alsa-devel] Query regarding SND_PCM_PLUGIN_DEFINE_FUNC()
by Pharaoh . 18 Mar '08

18 Mar '08

Hi, What does the mode field in SND_PCM_PLUGIN_DEFINE_FUNC() used for? is it for determining blocking mode or non-blocking mode of opening? -pharaoh

2 1

[alsa-devel] [PATCH] Removed deprecated sound/driver.h from Freescale MPC8610 drivers
by Timur Tabi 18 Mar '08

18 Mar '08

With commit 9004acc70e8c49c50c4c7b652f906f1e0ed5709d, include/sound/driver.h is deprecated. This patch removes the #include from fsl_ssi.c and fsl_dma.c. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 1 - 2 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 652514f..78de716 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -20,7 +20,6 @@ #include <linux/interrupt.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 145ad13..b2a11b0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -15,7 +15,6 @@ #include <linux/device.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.5.4

2 1

[alsa-devel] [PATCH] usbmidi.c: Call usb_autopm functions from rawmidi open/close callbacks
by Karsten Wiese 18 Mar '08

18 Mar '08

Lets "usb auto suspend" work for devices that set their usb_driver::supports_autosuspend to 1. Has no effect for other devices. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- sound/usb/usbmidi.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6c616c7..842486f 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -860,11 +860,14 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; + usb_autopm_get_interface(umidi->iface); return 0; } static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct snd_usb_midi* umidi = substream->rmidi->private_data; + usb_autopm_put_interface(umidi->iface); return 0; } -- 1.5.3.3 -------------------------------------------------------

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[alsa-devel] /usr/share/alsa/alsa.conf :and file extention
by Alexander Indenbaum 18 Mar '08

18 Mar '08

Hello, I'm playing with embedded system without any hardware sound device. I'd like to grab sound from ALSA application, for instance Firefox displaying YouTube clip inside flash plugin and to store output PCM stream into file on local disk. I got an advice to use File extension. So I defined following /usr/share/alsa/alsa.conf pcm.default{ type file file /tmp/pcm.out } But when alsa lib still complains about "Unknown PCM plughw:0,0" "Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_file.so" I have a nasty feeling that syntax of my /usr/share/alsa/alsa.conf is broken beyond recognition. I wonder what is the correct way to define my situation: * No real audio devices * All sound is output goes to file as PCM stream Do you have a clue? Alexander Indenbaum

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[alsa-devel] ~ALSA Audio coordinator/manager
by Harsha priya gupta 18 Mar '08

18 Mar '08

Hi all, I am looking to see if anyone is working on an Audio manager kind of solution as a part of ALSA. 'By Audio manager, I mean a coordinator of all audio input and output streams. A coordinator which is application and devices aware . Example functionality of an audio manager would to direct different steams to different devices like (voice output to headset and music output to speakers). Or to pause an media player when a voice recording starts.' If anyone is doing any work on these lines, please let me know about the work and its integration into ALSA as well. Thanks in Advance, Harsha.

2 1

[alsa-devel] basic questions
by Harsha priya gupta 18 Mar '08

18 Mar '08

Hi, I am ramping up on ALSA architecture and framework. I have a couple of basic questions. I apologize if these questions are very basic to be raised in this mailing list. If anyone can help me find answers, it would be great. Thanks in advance. 1. My assumption is that ALSA handles only PCM streams today. Does it handle encoded streams from application to sound card (Which can do hardware decoding)? 2. What are the hwdep files in ALSA used for? Where do the concerned IOCTLS land to? Thanks and Regards, Harsha.

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[alsa-devel] [PATCH] Add missing audio path between Mono Mixer and Mic PGAs.
by Mark Brown 18 Mar '08

18 Mar '08

From: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9713.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e2949c5..172ca1a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -500,6 +500,8 @@ static const char *audio_map[][3] = { {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"}, + {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"}, {"Mono Mixer", NULL, "Capture Mono Mux"}, /* DAC inv mux 1 */ -- 1.5.4.4

2 1

[alsa-devel] [PATCH] Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267)
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:18:52.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:48:18.000000000 -0300 @@ -7763,6 +7763,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), -- []'s Herton

2 1

[alsa-devel] [PATCH] Fix DAC assignment order in ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Actually clfe and surround DACs are inverted in alc883_dac_nids array (see ALC883 datasheet). I discovered this while testing multi-channel setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped in 6 channel mode (also in 4 channel mode you didn't get rear left/right output). Other models also were affected by this bug, as can be seen by the mixer layouts that "workaround" this (the real bug was not noticed, and some other models simply played with mixer and initial verbs). Thus along with fixing the order of dac nids, also change the models that relied on previous dac ordering properly. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:16:12.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:18:52.000000000 -0300 @@ -6450,7 +6450,7 @@ static int patch_alc882(struct hda_codec static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03, 0x05 + 0x02, 0x03, 0x04, 0x05 }; static hda_nid_t alc883_adc_nids[2] = { @@ -6958,12 +6958,12 @@ static struct snd_kcontrol_new alc883_me static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -6998,12 +6998,12 @@ static struct snd_kcontrol_new alc888_6s static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -7036,12 +7036,12 @@ static struct snd_kcontrol_new alc888_3s static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -7332,26 +7332,14 @@ static struct hda_verb alc883_haier_w66_ { } /* end */ }; -static struct hda_verb alc888_6st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ - { } -}; - static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; static struct hda_verb alc888_6st_dell_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -7985,7 +7973,7 @@ static struct alc_config_preset alc883_p }, [ALC888_6ST_HP] = { .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, + .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, -- []'s Herton

2 1

[alsa-devel] query about snd_pcm_hw_refine
by Pharaoh . 18 Mar '08

18 Mar '08

Hi I get following error when trying to run aplay with my plugin: #aplay -D myplug test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, St ereo aplay: set_params:882: Broken configuration for this PCM: no configurations available. After looking at the code, I figured out that its snd_pcm_hw_refine which is causing problem. It looks like following in pcm_params.c in alsa lib 1.0.13, int snd_pcm_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { int res; #ifdef REFINE_DEBUG snd_output_t *log; snd_output_stdio_attach(&log, stderr, 0); #endif printf("#####%s \n", __FUNCTION__); assert(pcm && params); #ifdef REFINE_DEBUG snd_output_printf(log, "REFINE called:\n"); snd_pcm_hw_params_dump(params, log); #endif printf("#####%s calling hw_refine\n", __FUNCTION__); res = pcm->ops->hw_refine(pcm->op_arg, params); printf("#####% return val of hw_refine = %d\n", res); #ifdef REFINE_DEBUG snd_output_printf(log, "refine done - result = %i\n", res); snd_pcm_hw_params_dump(params, log); snd_output_close(log); #endif return res; } I have modified the alsa.conf for my plugin by adding following in it at the end, pcm.myplug { type myplug card 0 } What could be the problem? -pharaoh.

1 3

[alsa-devel] DbAu1200/1210/1250 AC97 Driver
by Chris Gray 17 Mar '08

17 Mar '08

Hello, After looking for DbAu12xx AC97 ALSA support I ported the existing drivers to support Au1200 AC'97 PSC configuration. Being new to the development mailing list, is there a protocol on how to post patches for this sort of thing? Is this even the right forum? Thanks in advance! -Chris

1 0

[alsa-devel] no front speaker sound with ALC262
by Rene Dohmen 17 Mar '08

17 Mar '08

Dear List, I have a problem with an Intel HDA soundcard lspci output: ----- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) The sound volume is very very low when i use the headphone output. The device has integrated speakers but i don't hear any sound coming out of them (maybe because the volume is so low...). I've put all the mixer channels to the maximum values. When booted in Win XP the soundcard works correctly. Is see one line in my logs that indicate that something is wrong: hda_codec: Unknown model for ALC262, trying auto-probe from BIOS... I didn't have it working with other versions of ALSA. Tested with 1.0.14(provided with ubunty gutsy), then started reading, lots of suggestions stated that the problem would be solved in 1.0.15, so i compiled 1.0.16 from src and replaced the default alsa modules with the new ones. De device i 'm using is supplied by tangent ( http://www.tangent.com/t_all_in_one_LCDPC/vita_8000s.htm) its an all in one solution with touchscreen and speakers integrated in one small cabinet. I runned the alsa-info.sh script output and uploaded it: http://pastebin.ca/913865 I don't know if this is the correct place for getting help but all help is appreciated, Best Regards, Rene Dohmen

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[alsa-devel] Emvy24control s/pidf input issue
by Udo van den Heuvel 17 Mar '08

17 Mar '08

Hello, I just noticed that S/PDIF input on my Audiophile 2496 does not work. I see: [udo@wda ~]$ envy24control using --- input_channels: 0 --- output_channels: 4 --- pcm_output_channels: 8 --- spdif in/out channels: 2 Unable to read S/PDIF input switch: No such file or directory Changing the Digital Input from coax to optical or vice versa gives: Unable to write S/PDIF on/off switch: No such file or directory Unable to write S/PDIF input switch: No such file or directory This is on Fedora 8 x86_64 with alsa-tools 1.0.12-4 from Fedora Core 7, envycontrol 0.6.0. S/PDIF output works without a problem. Did not have S/PDIF input working before, as far as I remember. Using -v I do not see signal in the meters. In the source I see in spdif_input_update: if ((card_eeprom.subvendor != ICE1712_SUBDEVICE_DELTADIO2496) && ! card_is_dmx6fire) return; if ((err = snd_ctl_elem_read(ctl, spdif_input)) < 0) g_print("Unable to read S/PDIF input switch: %s\n", snd_strerror(err)); (etc) Why does snd_ctl_elem_read fail? How could I make S/PDIF input work again? Tips are very welcome! Kind regards, Udo

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[alsa-devel] AD1988B: ALSA doesn't use dmix automatically
by Dâniel Fraga 17 Mar '08

17 Mar '08

ALSA 1.0.15 http://alsa.opensrc.org/index.php/Dmix NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix. Dmix is enabled as default for soundcards which don't support hw mixing. So why if I try to play 2 sounds at the same time, the second one complains the "resource is busy"? Shouldn't it detect that AD1988B can't do hardware mixing and use dmix automatically? Obs.: I don't have any ~/.asoundrc or /etc/asound.conf lying around. Thanks! -- Linux 2.6.24: Arr Matey! A Hairy Bilge Rat! http://u-br.net http://www.abusar.org/FELIZ_2008.html Napalm Death - "Circle Of Hypocrisy" (Harmony Corruption - 1990)

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Re: [alsa-devel] ALSACTL wrong restore on ARM9
by Liam Girdwood 17 Mar '08

17 Mar '08

On Fri, 2008-03-07 at 16:28 +0100, Alexandre BOUIN wrote: > > Can you describe 'bad values' in a little more detail. > > You're right, some details are missing. > > For example, for first control, a playback volume control, value can go from > 0 to 63. > Before "store" command, I'd set it to 20. > I've check that in "/var/lib/alsa/asound/state", I had the same value. Here > is a part of this file : > > state.codec { > control.1 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 63' > iface MIXER > name 'HP Playback Volume' > value.0 20 > value.1 20 > } > > So saved value seems ok. > > After "restore" command, I get this : > > $amixer cget numid=1 > numid=1,iface=MIXER,name='HP Playback Volume' > ; type=INTEGER,access=rw------,values=2,min=0,max=63,step=0 > : values=45,45 > > That's why I say that values are not correct. > I'm currently using debian alsa-utils_1.0.16-1_arm package. > It's possible that your codec drivers register caching is borked. This can be easily confirmed with debug in your driver kcontrol code. If your driver looks ok, then I'd suggest looking into your alsa-lib cross build. There have been some problems with it being incorrectly cross compiled for ARM in the past (pls search the list for details). Fwiw, I use OpenEmbedded to build ALSA for ARM and I've never ran into this issue. Liam

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[alsa-devel] PATCH - some fixes and cleanup for ICE1724 cards
by Pavel Hofman 17 Mar '08

17 Mar '08

Hi Takashi, here is a small patch with minor fixes and cleanup. * removing the hack with NON_AKM ak4xxx type * support for card-specific flags in ak4114_stats * definition of the flags for corresponding cards Signed-by: Pavel Hofman <dustin(a)seznam.cz> Thanks, Pavel

2 1

[alsa-devel] [PATCH] [ML403-AC97CR] Remove duplicate snd_card_set_dev().
by Joachim Foerster 17 Mar '08

17 Mar '08

Hi Takashi, I noticed, that there was a change in ml403-ac97cr.c, which introduces a second call of snd_card_set_dev() in the _probe()-function. The "original" one is in the snd_ml403_ac97cr_create()-function. According the other changes, I guess, we want to have it in the _probe()-function, now (=> following patch)? From: Joachim Foerster <JOFT(a)gmx.de> We want to have snd_card_set_dev() in _probe(), but not a second one in snd_ml403_ac97cr_create(). Signed-off-by: Joachim Foerster <JOFT(a)gmx.de> --- sound/drivers/ml403-ac97cr.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 05a871a..b654007 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1191,8 +1191,6 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, return err; } - snd_card_set_dev(card, &pfdev->dev); - *rml403_ac97cr = ml403_ac97cr; return 0; } -- 1.5.4.3

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[alsa-devel] Grab PCM stream into file
by Alexander Indenbaum 17 Mar '08

17 Mar '08

Hello, I'm playing with embedded system without any hardware sound device. Strategically we'd like to forward PCM stream from ALSA clients on user level to proprietary application. As initial proof of concept stage I'd like to grab sound from ALSA application, for instance Firefox displaying YouTube clip inside flash plugin and to store output PCM stream into file on local disk. Please advice how I could achieve this goal. Alexander Indenbaum

1 0

[alsa-devel] How to stop playback on alsa synth?
by Lukasz Pawelczyk 16 Mar '08

16 Mar '08

Hello. I'd like to ask how to completely stop playback using alsa synth. Basically my program initialized output in the following way: snd_seq_open() snd_seq_set_client_name() snd_seq_client_id() snd_seq_parse_address() snd_seq_create_simple_port() snd_seq_connect_to() snd_seq_alloc_named_queue() Then I create a new thread that processes events. It sets tempo and in infinite loop it does: snd_seq_ev_set_fixed() snd_seq_event_output() This all is based on aplaymidi code. How can I asynchronously immediately stop all the playback? Basicaly it requires breaking snd_seq_event_output() thats locked on my thread. I tried everything I could come up with. I know very little about alsa synth programming and basically thats the only thing i need to do now. Tried: snd_seq_ev_set_queue_stop() snd_seq_stop_queue() snd_seq_remove_events() or even closing the seq completely to reopen it on another file playback with: snd_seq_stop_queue() snd_seq_disconnect_to() snd_seq_delete_simple_port() snd_seq_close() All failed, results were very various. I simply run out of ideas. It doesn't matter if the method would require to close seq or not. I just need to stop the playback so i can gracefully stop the thread and play another file (with eventual reopening ports/seq). Would appreciate any help. -- Regards Havner {jid,mail}:havner(at)pld-linux.org "Quis custodiet ipsos custodes?"

1 1

[PATCH] Correct dapm controls length.
by Robert Jarzmik 16 Mar '08

16 Mar '08

Dapm control names should follow alsa standard, which defines controls name's length to 44 and not 32. Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4b03e3f..a61024c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -281,7 +281,7 @@ static int dapm_new_mixer(struct snd_soc_card *soc_card, struct snd_soc_dapm_widget *w) { int i, ret = 0; - char name[32]; + char name[44]; struct snd_soc_dapm_path *path; /* add kcontrol */ @@ -295,7 +295,7 @@ static int dapm_new_mixer(struct snd_soc_card *soc_card, continue; /* add dapm control with long name */ - snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name); + snprintf(name, 44, "%s %s", w->name, w->kcontrols[i].name); path->long_name = kstrdup (name, GFP_KERNEL); if (path->long_name == NULL) return -ENOMEM; -- 1.5.6.5

1 0

[PATCH] Dirty hack to use GPIO95 in AC97 reset.
by Robert Jarzmik 16 Mar '08

16 Mar '08

Mio Mitac A701 uses GPIO95 instead of GPIO113 as the reset line connected to the WM9713 chip. This is dirty, and should be replaced by an elegant solution where cold and warm reset are implemented in machine driver mioa701_wm9713.c Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- sound/soc/pxa/pxa2xx-ac97.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index cb12544..86fb9c2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -145,10 +145,10 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) #ifdef CONFIG_PXA27x /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + pxa_gpio_mode(95 | GPIO_OUT | GPIO_DFLT_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + pxa_gpio_mode(95 | GPIO_ALT_FN_1_OUT); udelay(500); #elif defined(CONFIG_PXA3xx) /* Can't use interrupts */ @@ -303,8 +303,8 @@ static int pxa2xx_ac97_resume(struct platform_device *dev) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); #endif #ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 95 as AC97 Reset on Bulverde */ + pxa_gpio_mode(95 | GPIO_ALT_FN_1_OUT); #endif clk_enable(ac97_clk); return 0; @@ -457,8 +457,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev) return -ENOMEM; #ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 95 as AC97 Reset on Bulverde */ + pxa_gpio_mode(95 | GPIO_ALT_FN_1_OUT); ac97conf_clk = clk_get(&pdev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { -- 1.5.5.3 --=-=-= Content-Type: text/x-diff Content-Disposition: inline; filename=0002-Dirty-hack-to-workaround-pxa27x-buggy-hardreset.patch

1 0

[alsa-devel] Maya44 driver in the official ALSA tree
by Piotr Zaryk 15 Mar '08

15 Mar '08

Hi, When will be the next alsa-driver version released? Can't wait till maya44 support will be added to the official alsa tree. It will help with the development of the new driver. -- Piotr Zaryk

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Re: [alsa-devel] no front speaker sound with ALC262
by Takashi Iwai 14 Mar '08

14 Mar '08

At Fri, 29 Feb 2008 17:24:03 +0100, Rene Dohmen wrote: > > No I still don't know where the speakers are conencted. > I asked the support engineers at tangent to send me a detailed > blockdiagram of their sound solution and ik think i'll receive that > sooner or later. > > So, for now, I just tried passing the EAPD option to all NIDS in the > codec#0 file. > > I attached the code#0 file while working with very low volume > > When I run > sudo hda-verb/hda-verb /dev/snd/hwC0D0 0x03 SET_EAPD 0x02 > or > sudo hda-verb/hda-verb /dev/snd/hwC0D0 0x02 SET_EAPD 0x02 Doh, I actually misunderstood that your codec were ALC268 but it's ALC262. EAPD command can be issued only on widgets that support. ALC262 has no EAPD pin caps at all. so it doesn't work like that. Instead, ALC262 (and other Realtek codecs) has a special sequence with coefficients to the vendor-defined widget. Yes, it's a black magic. Try the following sequence: 0x20 SET_COEF_INDEX 0x07 0x20 SET_PROC_COEF 0x3060 0x3060 can be different numbers like 0x3070 or 0x3050. If you set a different number, reset the coef index again (i.e. always call as a pair). Takashi

3 3

[alsa-devel] aplay: playback:2021: read error
by Pharaoh . 14 Mar '08

14 Mar '08

Hi, I am getting aplay: playback:2021: read error while running my plugin on an arm based board, it seems that, this has to do something with permissions etc or may be the sound card is being shared etc. I don't have a registered sound card since there is no kernel driver for my sound hw. I use my plugin as, aplay -D my_plug test.wav -pharaoh.

2 2

[alsa-devel] AK4114 - capturing spdif input stops the stream
by Pavel Hofman 14 Mar '08

14 Mar '08

Hi, here is my scenario: ICE1724 card, trying to record from SPDIF input via AK4114, ICE1724 slaved to SPDIF clock from the receiver. The card detects SPDIF input rate correctly (in my case ESI Juli where AK4114 is provided with independent clock signal to enable the rate detection). Let's say it is 192000. Now I want to arecord the input stream, let's say in CD quality, using the plug plugin. The command fails, because of the check in ak4114.c: res = external_rate(rcs1); if (!(flags & AK4114_CHECK_NO_RATE) && runtime && runtime->rate != res) { snd_pcm_stream_lock_irqsave(ak4114->capture_substream, _flags); if (snd_pcm_running(ak4114->capture_substream)) { printk(KERN_DEBUG "rate changed (%i <- %i)\n", runtime->rate, res); snd_pcm_stop(ak4114->capture_substream, SNDRV_PCM_STATE_DRAINING); res = 1; } snd_pcm_stream_unlock_irqrestore(ak4114->capture_substream, _flags); } The stream gets stopped because res = 192000 and runtime->rate = 44100. The problem is that the capture device still offers all the available rates, instead of the single SPDIF input one. Thus, the plug plugin is not forced to convert from 192000 to 44100 and runtime->rate is 44100 instead of 192000. Of course recording at 192000 works fine. What would be the best way to force the SPDIF capture device to offer (advertise) only the current rate? This functionality would have to be applicable only to the few cards correctly detecting incoming rate (unlike most ice1724 cards without independent clock in slave mode). If the card cannot detect input frequency reliably (the driver developer should know), I guess it should not do the check in ak4114.c at all - I can prepare a simple patch for that. The same will most probably apply to cards with AK4117 as the code looks the same. I know it is a minor issue but if a user hits this problem, it will take him a lot of effort to fix his setup appropriately. Thanks, Pavel.

2 7

[alsa-devel] [ALC262 patch_realtek.c] Fujitsu Lifebook port replicator/dock headphone jack sense
by Tony Vroon 14 Mar '08

14 Mar '08

Good afternoon, Could you please consider the following patch for the 2.6.26 merge window. The docking station headphone output had no audio and jack sense was not considered. Jack information from the laptop itself and the dock are combined, as the dock does not obscure the connector. Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin P.S. Codec graph included in case anyone wants to start using mic jack sense for automated capture source selection.

2 3

[alsa-devel] Underrun errors in aplay
by Barclay, Bruce 14 Mar '08

14 Mar '08

I have written a driver for Alsa and I think the sound is correct but I do get underrun errors. I don't understand how aplay would know if I am starved. This doesn't make sense to me. I would only think that the driver should know about underruns but maybe I am confused with this. I am wondering if I am calling snd_pcm_period_elapsed too often. I am calling that when I have free H/W buffers. Thanks, Bruce

2 1

[alsa-devel] reading peak meter values from a RME HDSP madi soundcard
by ListsAccount 14 Mar '08

14 Mar '08

Hi, I'd like to read the peak values of the RME HDSP madi soundcard. To do this, I use the function: snd_hwdep_ioctl() with the ioctl command "SNDRV_HDSPM_IOCTL_GET_PEAK_RMS". I use the following code: // create container hdspm_peak_rms_t madi_peak_rms; hdspm_peak_rms_t* ptr_madi_peak_rms = &madi_peak_rms; //snd_hwdep_open ... // fill the container with values snd_hwdep_ioctl(hw, SNDRV_HDSPM_IOCTL_GET_PEAK_RMS, (void *)&ptr_madi_peak_rms); for (unsigned int chn = 0; chn < numberOfChannels; ++chn) { //put values from madi soundcard into std::vector containers currentInputPeak[chn] = madi_peak_rms.level_offset[chn]; currentPlaybackPeak[chn] = madi_peak_rms.level_offset[chn+64]; currentOutputPeak[chn] = madi_peak_rms.level_offset[chn+128]; } The upper 24 bit of the read out values should be the signal peak. The next 8 bit seem to represent a clipping counter. For the first eight input channels this works pretty well. But for every eighth channel of all further channels (input, playback and output) the peak value is to low and is not in time with the other channels. Do i use the function in the right way? Is there a special function to get the peak meter values? (In a similar way as for ADAT soundcards?) Best regards.

1 0

[alsa-devel] bugfix for RME HDSPM alsa driver
by ListsAccount 14 Mar '08

14 Mar '08

Hi, I have had the following problem with the RME HDSPM alsa driver: settings of card: Preferred Sync Reference = Word Sample Clock Source = AutoSync and no input plugged in (neither madi nor optical) playing a file (important!) Monitoring the external rate with amixer, the value was jumping between "None" and "48000 kHz". The proc output showed that the sample rate on word clock was still right at 48000 kHz while the external rate was jumping here, too. External rate seems to jump between word clock and madi input select. When "Preferred Sync Reference = Word" is selected, just the word clock rate should be displayed! I fixed the problem with following patch in alsa-driver-1.0.16/alsa-kernel/pci/rme9652/hdspm.c in function hdspm_external_sample_rate() line 668: There was a typo: instead of status2, status was used. 668c668 < (status & HDSPM_SelSyncRef0) == 0) { --- > (status2 & HDSPM_SelSyncRef0) == 0) { (same way as in snd_hdspm_proc_read_madi function line 3123) greetings

1 0

[alsa-devel] Scope of snd_pcm_hw_params_alloca() ?
by Alan Horstmann 14 Mar '08

14 Mar '08

I've spent a couple of days pining this down. snd_pcm_hw_params_alloca() is part of the user API -right? But accomplished as a #define in pcm.h. However if it is called inside a function, and the pointer stored for future use, after leaving the function the allocation seems to have gone, and attempting thereafter to use the pointer to access the hwparams space in another API function results in seg-fault. This seems rather counter-intuitive (ie unexpected) to someone who has no knowledge of the inner workings, but simply uses the API; the allocation disappears without having been specifically free'd or released in any way. I know that calling it globally gets round the problem, but certainly I could do without all the debug needed to discover that. Comments? Alan

2 3

Re: [alsa-devel] Scope of snd_pcm_hw_params_alloca() ?
by Takashi Iwai 14 Mar '08

14 Mar '08

At Thu, 13 Mar 2008 15:41:02 -0300, Aldrin Martoq wrote: > > Hello, > > On Thu, Mar 13, 2008 at 1:44 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Mar 2008 16:21:51 +0000, > > Alan Horstmann wrote: > > > I've spent a couple of days pining this down. > > > snd_pcm_hw_params_alloca() is part of the user API -right? But accomplished > > > as a #define in pcm.h. > > Because alloca can be used only with a macro due to its nature. > > > However if it is called inside a function, and the pointer stored for future > > > use, after leaving the function the allocation seems to have gone, and > > > attempting thereafter to use the pointer to access the hwparams space in > > > another API function results in seg-fault. > > You should use snd_pcm_hw_params_malloc() instead. *_alloca() from > asoundlib are based in alloca(). From my system man: > > $ man alloca > NAME > alloca - memory allocator > > SYNOPSIS > #include <alloca.h> > > void *alloca(size_t size); > > DESCRIPTION > The alloca() function allocates size bytes of space in the stack frame > of the caller. This temporary space is automatically freed when the > function that called alloca() returns to its caller. > > [..] > > Yup, this is actually very useful in many many many places! Especially in C, that isn't easy to have implicit destructors. BTW, a pitfall once I fell into is that the alloca frees the data at the function exit, not at the block exit. So, the following code eats stack up and shall evetually dies: { ... for (;;) { char *tmp = alloca(4096); ... } ... } Takashi

1 0

[alsa-devel] [PATCH] Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX
by Herton Ronaldo Krzesinski 14 Mar '08

14 Mar '08

Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up linux-2.6.24/sound/pci/hda/patch_realtek.c.orig linux-2.6.24/sound/pci/hda/patch_realtek.c --- linux-2.6.24/sound/pci/hda/patch_realtek.c.orig 2008-03-11 16:00:01.000000000 -0300 +++ linux-2.6.24/sound/pci/hda/patch_realtek.c 2008-03-11 16:09:43.000000000 -0300 @@ -13399,6 +13399,7 @@ static const char *alc662_models[ALC662_ }; static struct snd_pci_quirk alc662_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), -- []'s Herton

2 1

[alsa-devel] [PATCH 3/5] asihpi: Remove HPI4000
by Eliot Blennerhassett 14 Mar '08

14 Mar '08

Remove HPI4000 Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com --- diff -r 6def4892d3f5 pci/asihpi/Makefile --- a/pci/asihpi/Makefile Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/Makefile Thu Mar 06 14:31:55 2008 +1300 @@ -7,7 +7,7 @@ include $(SND_TOPDIR)/Makefile.conf snd-asihpi-objs := asihpi.o hpimod.o hpimsginit.o\ hpicmn.o hpifunc.o hpidebug.o hpidspcd.o\ - hpios_linux_kernel.o hpi4000.o hpi6000.o hpi6205.o hpimsgx.o + hpios_linux_kernel.o hpi6000.o hpi6205.o hpimsgx.o obj-$(CONFIG_SND_ASIHPI) += snd-asihpi.o diff -r 6def4892d3f5 pci/asihpi/hpimsgx.c --- a/pci/asihpi/hpimsgx.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpimsgx.c Thu Mar 06 12:21:49 2008 +1300 @@ -476,8 +476,6 @@ void HPI_MessageEx( ep_name = "HPI_6000"; else if (ep == HPI_6205) ep_name = "HPI_6205"; - else if (ep == HPI_4000) - ep_name = "HPI_4000"; else ep_name = "unknown"; @@ -598,7 +596,8 @@ static void InStreamClose( if (hOwner == aIStreamUserOpen[phm->wAdapterIndex][phm->u.d.wStreamIndex]. hOwner) { - /* HPI_DEBUG_LOG(INFO,"closing adapter %d instream %d owned by %p\n", + /* HPI_DEBUG_LOG(INFO,"closing adapter %d " + "instream %d owned by %p\n", phm->wAdapterIndex, phm->u.d.wStreamIndex, hOwner); */ aIStreamUserOpen[phm->wAdapterIndex][phm->u.d.wStreamIndex]. hOwner = NULL; @@ -701,7 +700,8 @@ static void OutStreamClose( if (hOwner == aOStreamUserOpen[phm->wAdapterIndex][phm->u.d.wStreamIndex]. hOwner) { - /* HPI_DEBUG_LOG(INFO,"closing adapter %d outstream %d owned by %p\n", + /* HPI_DEBUG_LOG(INFO,"closing adapter %d " + "outstream %d owned by %p\n", phm->wAdapterIndex, phm->u.d.wStreamIndex, hOwner); */ aOStreamUserOpen[phm->wAdapterIndex][phm->u.d.wStreamIndex]. hOwner = NULL; @@ -904,8 +904,9 @@ static u16 HPIMSGX_Init( phr->wError = HPI_ERROR_PROCESSING_MESSAGE; return phr->wError; } - /* if the adapter was created succesfully save the mapping for future use */ if (hr.wError == 0) { + /* the adapter was created succesfully + save the mapping for future use */ hpi_entry_points[hr.u.s.wAdapterIndex] = entry_point_func; /* prepare adapter (pre-open streams etc.) */ HPI_DEBUG_LOG(DEBUG, @@ -936,7 +937,7 @@ static void HPIMSGX_Cleanup( } for (; wAdapter < wAdapterLimit; wAdapter++) { - /* printk(KERN_INFO "Cleanup adapter #%d\n",wAdapter); */ + /* printk(KERN_INFO "Cleanup adapter #%d\n",wAdapter); */ for (i = 0; i < HPI_MAX_STREAMS; i++) { if (hOwner == aOStreamUserOpen[wAdapter][i].hOwner) { struct hpi_message hm; diff -r 6def4892d3f5 pci/asihpi/hpipcida.h --- a/pci/asihpi/hpipcida.h Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpipcida.h Wed Mar 05 21:59:17 2008 +1300 @@ -34,20 +34,4 @@ HPI_PCI_VENDOR_ID_TI, HPI_ADAPTER_PCI204 HPI_PCI_VENDOR_ID_AUDIOSCIENCE, PCI_ANY_ID, 0, 0, (kernel_ulong_t) HPI_6000} , { -HPI_PCI_VENDOR_ID_MOTOROLA, HPI_ADAPTER_DSP56301, - HPI_PCI_VENDOR_ID_AUDIOSCIENCE, PCI_ANY_ID, 0, 0, - (kernel_ulong_t) HPI_4000} -, - /* look for ASI cards that have 0x12cf sub-vendor ID, - like the 4300 and 4601 */ -{ -HPI_PCI_VENDOR_ID_MOTOROLA, HPI_ADAPTER_DSP56301, 0x12CF, PCI_ANY_ID, - 0, 0, (kernel_ulong_t) HPI_4000} -, - /* look for ASI cards that have sub-vendor-ID = 0, - like the 4501, 4113 and 4215 revC and below */ -{ -HPI_PCI_VENDOR_ID_MOTOROLA, HPI_ADAPTER_DSP56301, 0, PCI_ANY_ID, 0, 0, - (kernel_ulong_t) HPI_4000} -, { 0,}

2 4

[alsa-devel] [PATCH] hda: 92HD73xxx distortion fix
by Matthew Ranostay 14 Mar '08

14 Mar '08

Fixed issue on some laptops that if the Master mixer and DAC mixers are turned all the way up that will cause distortion. This is fixed by limiting the max volume with the volume knob nid. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 547e051a88b5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Mar 12 13:12:15 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Mar 12 21:48:13 2008 -0400 @@ -526,6 +526,25 @@ static struct hda_verb stac92hd73xx_6ch_ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb dell_eq_core_init[] = { + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, + /* setup audio connections */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, /* setup import muxs */ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -3460,17 +3479,19 @@ again: switch (spec->board_config) { case STAC_DELL_M6: - spec->init = dell_m6_core_init; + spec->init = dell_eq_core_init; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; break; - case 0x10280254: /* Digital Mics */ + case 0x10280271: /* Digital Mics */ + case 0x10280272: + spec->init = dell_m6_core_init; + /* fall-through */ + case 0x10280254: case 0x10280255: - case 0x10280271: - case 0x10280272: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; break;

2 1

[alsa-devel] alsa conf file
by tomars 14 Mar '08

14 Mar '08

I write a asound.conf file in the /etc/ , the detail is as follows. pcm.hardware{ type hw:0 card 0 device 0 } pcm_slave.test { pcm "hardware" format S16_LE rate 44100 channels 2 period_time 20000 buffer_time 80000 } pcm.mix1{ type dmix ipc_key 1235675 ipc_key_add_uid true slave "test" } pcm.mix2{ type dmix ipc_key 1235675 ipc_key_add_uid true slave { pcm "hw:0" format S16_LE rate 44100 channels 2 period_time 20000 buffer_time 80000 } } pcm.!default { type plug slave.pcm mix2 #slave.pcm mix1 } when I use the slave.pcm mix2, the aplay works well. but if I select the slave.pcm mix1, the aplay gets the info: ALSA lib pcm_direct.c:1462:(_snd_pcm_direct_get_slave_ipc_offset) Invalid type 'dmix' for slave PCM I compare the definition of mix1 with the definition of mix2, I feel they have the identical meaning. so why does the mix2 get the error? thank you!

2 1

[alsa-devel] problems howto send simple 'noteon' event with pyalsa...
by ki.ber.kom.uni.st 14 Mar '08

14 Mar '08

e, i was happy user of PySeq [1] and it was easy to send directly 'noteon' event.. more or less in python interactoin session: import pyseq seq = pyseq.PySeq() seq.createOutPort() event = snd_seq_event() event.setSource(0) event.setSubscribers() event.setNoteOn(ch, note, velocity) event.sendAsIs(seq) it seems that pyalsa will go along alsa development so i tried pyalsa.. but.... i tried this: from pyalsa import alsaseq seq = alsaseq.Sequencer() seq.create_simple_port("myPort", alsaseq.SEQ_PORT_TYPE_APPLICATION, alsaseq.SEQ_PORT_CAP_SUBS_READ | alsaseq.SEQ_PORT_CAP_READ | alsaseq.SEQ_PORT_CAP_WRITE | alsaseq.SEQ_PORT_CAP_SUBS_WRITE) event = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_NOTE) event.set_data({'note.note' : 0x40, 'note.velocity' : 127, 'note.duration' : 1, 'note.off_velocity' : 0}) event.dest = (20, 0) seq.output_event(event) seq.drain_output() after drain it says: "SequencerError: Failed to drain output: Invalid argument" i can connect and disconnect ports around and (20,0) is functional port where i send events with no problems from PySeq... any ideas? do i have to use queues to be able to send 'noteon'.. for my purposes it seems to me that that's overkill.... thanx.. [1] http://www.sci.ccny.cuny.edu/~brinkman/software/pyseq/

3 3

[alsa-devel] [PATCH] Fix ALC662 DAC mixer mutes
by Herton Ronaldo Krzesinski 13 Mar '08

13 Mar '08

Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03, 0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to mute?). The result is that currently mute for "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch" and "LFE Playback Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard with 3stack-6ch model). The solution I found for this is to mute the proper inputs in 0x0c, 0x0d, 0x0e audio mixers. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up linux-2.6.24/sound/pci/hda/patch_realtek.c.orig linux-2.6.24/sound/pci/hda/patch_realtek.c --- linux-2.6.24/sound/pci/hda/patch_realtek.c.orig 2008-03-11 13:45:07.000000000 -0300 +++ linux-2.6.24/sound/pci/hda/patch_realtek.c 2008-03-11 14:01:57.000000000 -0300 @@ -13004,13 +13004,13 @@ static struct hda_channel_mode alc662_5s static struct snd_kcontrol_new alc662_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), /*Input mixer control */ @@ -13027,7 +13027,7 @@ static struct snd_kcontrol_new alc662_ba static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -13044,13 +13044,13 @@ static struct snd_kcontrol_new alc662_3S static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), -- []'s Herton

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[alsa-devel] Hardware and software period size
by Mathieu Monney 13 Mar '08

13 Mar '08

Hi all, I'm currently developping a layer between ALSA and a VoIP library for an embedded device. The device cannot have a different period size than 1024 for playback and 8 for capture (I don't know why :-( ). So my problem is that the VoIP library ask me to have a period size different of the hardware period size. So I decided to read/write to the sound device using the provided software period size rather than the hardware period size.. Is it a good idea? It seems to work but I have some sound quality issues :-( ! Thank you Mathieu

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Re: [alsa-devel] [Alsa-user] Metallic recording in cs46xx on Thinkpad T22
by Rene Herman 13 Mar '08

13 Mar '08

On 11-03-08 08:24, Gadi Oron wrote: > I am trying to record sound on an old Thinkpad T22 that uses the cs46xx > sound driver. > > Each time you start to record you have a 10% chance of having the > recording completely distorted and having a metallic sound. When you > look at the waveform it looks as though there are small segments with > sharp transitions between them, a little like if these segments were > moved a little from their correct place. No insights, but I confirm the bug with a TerraTec DMX XFire 1024 (CS4624). Rene.

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[alsa-devel] Trigger problems while writing Alsa Shim
by bruce＠azbarclay.com 13 Mar '08

13 Mar '08

I am writing a shim from alsa to our own sound card drivers. I was able to get the init driver working and our sound card recognized. Things seem to be set up correctly. When I do aplay to my device (USB device works find), it calls my copy callback twice and then it calls the trigger and fails. The call that fails is in pcm_write (writei_func) with a -5. I know that my callback returns zero but there must be something else under the covers since that gets to be zero. Any help would be appreciated. T

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Re: [alsa-devel] ALSA utility aplay/arecord error
by Saurav SAHU 13 Mar '08

13 Mar '08

Hi all The platform on which I am trying to perform ALSA playback and record with USB sound device is based on a MIPS CPU running Linux kernel 2.6.14. The kernel uses non-coherent DMA in the configuration: CONFIG_DMA_NONCOHERENT=y CONFIG_DMA_NEED_PCI_MAP_STATE=y CONFIG_CPU_BIG_ENDIAN=y CONFIG_SYS_SUPPORTS_BIG_ENDIAN=y CONFIG_BOOT_ELF32=y CONFIG_MIPS_L1_CACHE_SHIFT=5 I have attached the kernel .config file for further information. So far, I have not succeeded in hearing sound with 'aplay', or capturing sound with 'arecord'. If there have been any similar issues reported on ALSA usage with MIPS CPU based on non-coherent DMA in the past, please let me know. Also, if there have been any patches developed for successful resolution of this issue on similar platform/configuration, it would be very useful for me. Thanks & Regards Saurav ----- Original Message ----- From: Saurav Sahu To: alsa-user(a)lists.sourceforge.net Sent: Friday, March 07, 2008 4:33 PM Subject: Re: ALSA utility aplay/arecord error I have been able to resolve the previously reported errors by setting the environment variable for ALSA configuration file on the target filesystem of the embedded platform. However, I do not hear any audio from USB sound device with speaker-test or aplay: Upon using arecord for recording, I get an error: Is this problem related to a low-level driver issue which may not be working well on target platform? Could there be a DMA related issue as the samples don't seem to be transferred to the USB audio device. ----- Original Message ----- From: Saurav Sahu To: alsa-user(a)lists.sourceforge.net Sent: Thursday, March 06, 2008 4:28 PM Subject: ALSA utility aplay/arecord error Hello I am working towards using ALSA for playback and capture on an embedded platform running Linux. The kernel uses ALSA driver version 1.0.10rc1. I had manually built and installed ALSA library (v1.0.10) and ALSA utilities (v1.0.10) for the target platform filesystem. I have a USB audio device (C-Media) connected to the USB port of the board and it seems to be successfully successfully detected by the driver. However, when I run ALSA utilities like aplay, arecord or speaker-test, I observe errors opening the hardware device which may be related to the configuration on the filesystem. Are these errors from ALSA utility/library due to the detection of hardware, ALSA configuration files or something else related to my target device filesystem? Thanks in advance for any help. Do let me know if you need any more details. Best Regards Saurav

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[alsa-devel] [PATCH] asihpi - new checkpatch = more changes
by Eliot Blennerhassett 13 Mar '08

13 Mar '08

Signed-off-by: Eliot Blennerhasssett <linux(a)audioscience.com> * HPI version 3.09.15 * Updating to latest version of checkpatch resulted in some more warnings to clean up. * A few whitespace cleanups * Removed DBG_TXT and DBG_CHAR macros --- patch attached -- Eliot Blennerhassett www.audioscience.com

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[alsa-devel] Hda-intel: AC_VERB_SET_PIN_WIDGET_CONTROL appears to do nothing for HDMI device
by Ben Firshman 13 Mar '08

13 Mar '08

This is on 1.0.16. Forgive me if I appear clueless, I have only been working on this for the past few hours. In an attempt to get HDMI audio working on the AppleTV, I have added this line to the presets in patch_atihdmi.c: { .id = 0x10951390, .name = "SI1390 HDMI", .patch = patch_atihdmi }, The device is detected fine: ben@mythfrontend:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC882 Digital [ALC882 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ben@mythfrontend:~$ and I can send audio to the device, but it doesn't work. I presume this because this is doing nothing: static struct hda_verb atihdmi_basic_init[] = { /* enable digital output on pin widget */ { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {} /* terminator */ }; ben@mythfrontend:~$ cat /proc/asound/card0/codec#1 Codec: Generic 1095 SI1390 HDMI Address: 1 Vendor Id: 0x10951390 Subsystem Id: 0xffffffff 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 0x02 [Audio Output] wcaps 0x6211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Node 0x03 [Pin Complex] wcaps 0x40738d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0894: OUT Detect R/L Pin Default 0x98460010: [Fixed] SPDIF Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x02 "Pin-ctls: 0x00:" should be reading "Pin-ctls: 0x40: OUT" should it not? What is wrong? Thanks Ben

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[alsa-devel] aplay doesnt work - new problem`
by Pharaoh . 13 Mar '08

13 Mar '08

I am trying to run my pcm external io type plugin on an arm based board. I have no kernel alsa driver and the user space plugin is supposed to take care of talking to DSP using shared memorty and send sound data. Since, I dont have kernel space ALSA driver, do I need to enable kernel ALSA support in .config? I am facing following problem, #aplay ALSA lib conf.c:2852:(snd_config_hook_load) cannot access file /home/pharaoh/alsa/cards/aliases.conf ALSA lib pcm.c:2014:(snd_pcm_open_conf) Invalid type for PCM default definition (id: default, value: cards.pcm.default) aplay: main:550: audio open error: Invalid argument I can see that, aliases.conf is being searched at wrong place (its actually looking on my host machine), how can I fix this? -pharaoh.

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[alsa-devel] driver development questions
by gmail zaq79324 12 Mar '08

12 Mar '08

Hi everyone, I'm currently developing an ALSA driver from scratch. It is almost done, but memory buffer management problem remains. I have tried in various way of debugging. But currently I'm almost giving it up. My target hardware has a small data buffer(2K+2K ring buffer). The driver fills the first half buffer(2K) and writes hardware port. After this the interrupt occurs to urge writting the second half buffer and hardware port. The second buffer is filled in the same way. The buffer filling scheme will repeat continuously until the end of the sound. Through my driver, aplay command can generate the sound within few seconds, but it stops before finishing completely. It seems to have the problem of buffer management. So I set xrun_debug option like as follows. echo 1 > /proc/asound/SM502I2S/pcm0p/xrun_debug And I got the following output. Does anyone know what the problem is and how to fix it? # ./bin/aplay --channels=1 --period-size=2048 sample.wav Playing WAVE 'sample.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:1266: write error: Input/output error ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -2048, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -2048, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -2048, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -512, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -2048, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1536, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? ALSA sound/core/pcm_lib.c:240: Unexpected hw_pointer value [1] (stream = 0, delta: -1024, max jitter = 11025): wrong interrupt acknowledge? Regards, Tadashi

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[alsa-devel] It's time to discuss how to get a Gadget Labs driver into the Alsa tree...
by Mike Mazarick 12 Mar '08

12 Mar '08

As discussed and posted to the Alsa Devel and Planet CCRMA lists back in December (when this was 'a dream' ), we currently have a working Gadget Labs Alsa driver. There are a few more 'features' that are being considered for development (multi card support, a control panel, etc) and more extensive testing and verification by novice users is desired. The main idea is that we intend to ask the main Alsa developers for information twice; first when we have a functional driver and second when we have one that is 'user tested' and is ready to go (ie. when there is a high degree of certainty that all the features are finished and it's use won't cause any major problems). We are now at the first crossroads. There is a small team of developers, Dave Marion and Rick Macri, with some assistance from Pete Mander on SVN setup (Dave and Rick finished before Pete got a chance to 'play'). We have set up a sourceforge page to centralize our effort. The developers and I believe the code currently in the SVN trunk (gl824) is suitable for identifying the appropriate steps to get it from the project SVN repository into the Alsa Hg tree. The project main page is here: https://sourceforge.net/projects/wavepro-driver/ The source code is in an SVN repository here: http://wavepro-driver.svn.sourceforge.net/viewvc/wavepro-driver/ There is a very short web page listed for development: http://wavepro-driver.sourceforge.net/ There is a private mailing list of about 16 people that is tracking what is going on. The group email address is: wavepro-driver-developers(a)lists.sourceforge.net (however, you may have to be subscribed to mail something to the list; I'm not sure). If you are subscribed, there is an archive available here: https://sourceforge.net/mailarchive/forum.php?forum_name=wavepro-driver-deve lopers Our goal is to have this driver included in Alsa so that an end user could have this type of card in their system, and everything would automagically work like all other cards. The only way this can happen is if the driver was included as part of Alsa. The key 'next' questions are: 1. When are the next couple of releases of Alsa planned? How much time before the release date should all drivers be 'in the Hg tree'? 2. Does the code that is currently in the Gadget Labs SVN repository meet the coding and syntax standards of Alsa? We've tried to make it meet the standards, and it is fairly easy to change at this stage if you happen to spot something that is inconsistent with what you'd recommend. 3. We have not modified Alsa to make this driver automatically; there is a 'Makefile' included currently. We currently 'modprobe snd_rawmidi' and 'modproble snd_pcm' and then 'insmod gl824.ko' to get it installed. The required changes to Alsa are fairly trivial and would be the same for every other sound card. Please let us know if there is something that should happen besides what is in the GL repository. 4. One of the key reasons that this driver was able to be developed and tested in 2-3 months of part time effort is because there is already complete source available for a Windows driver, complete with new firmware and vhdl source (see 'External' in the repository). The Windows developer is a member of our mailing list and has been encouraging/helping our effort. However, he didn't bother with GPL licensing (it's a 'Mostek/Waldemar' license instead, but the source is readily available). There may be a few header files that he wrote and the PLD firmware is definitely from his development. Will everything (including the firmware) need to be GPLv2 before it can be included in the alsa tree? 5. What is needed for support? What would you like us to be available to do going forward? 6. How can we make your life a little easier by modifying our software? What would you like to see that you don't see? Although we will diligently try to get a broader base of testing to happen, we will tell you realistically that there are probably very few users who are likely to test this out. I don't know of the exact number of Gadget Labs cards that were produced, but the best guess is ~5,000 cards before the manufacturer went bankrupt in 2001. We feel the effort of creating a linux driver has been advertised in most of the appropriate places, but the mailing list that is tracking this is only 16 people, with 4 main participants. In other words, I don't think this will be a 'SoundBlaster' or 'Audigy' with thousands of people giving feedback on how it works. However, I do believe that this will be a very viable and cost effective solution for some people who want an 8 channel in/out, 24 bit, 48 khz, semi-professional soundcard solution with midi. Although the soundcard does not support DMA directly (a limitation of the PLX9052 interface chip), Rick has developed an innovative scheme of using a user buffer and dumping the result into a kernel ring buffer that provides as little as 2ms latency. You may also find the method/architecture for development of interest to you (see web page and SVN listings). Please provide some of the key information you need so that we can work together most effectively. Answering some the questions above would be a good way to start. We would welcome your feedback so that we can incorporate into the GL project codebase. After your feedback has been incorporated in the code and testing is completed, we will be pinging you again (the second time) to let you know we are ready to put the code in the alsa tree. Regards, Mike Mazarick mazarick(a)bellsouth.net

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[alsa-devel] ioctl(arg) - uninitialised byte(s) in alsa lib
by Mathieu Monney 12 Mar '08

12 Mar '08

Hi all, I'm currently working on a small audio stack on top of alsa and I'm getting some "Syscall param ioctl(arg) contains uninitialized byte(s)" errors when I run my program using valgrind. I don't know if it is a bug in my program or in alsa lib. What I'm sure is that my program is working even though these errors appear. The dump: ==14279== Syscall param ioctl(arg) contains uninitialised byte(s) ==14279== at 0x40007F2: (within /lib/ld-2.6.1.so) ==14279== by 0x40DE911: snd_pcm_prepare (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40F2552: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40DE911: snd_pcm_prepare (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4F0C: snd_pcm_hw_params (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x80A6599: AlsaConfigureDevice (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== by 0x80A6DE3: pjmedia_snd_open (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== by 0x809EC85: start_sound_device (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== by 0x805EE1E: pjsua_set_snd_dev (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== by 0x80615AA: pjsua_media_subsys_start (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== by 0x805AAC5: pjsua_start (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== by 0x805001B: app_main (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) ==14279== ==14279== Syscall param ioctl(arg) contains uninitialised byte(s) ==14279== at 0x40007F2: (within /lib/ld-2.6.1.so) ==14279== by 0x40DE911: snd_pcm_prepare (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4F0C: snd_pcm_hw_params (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x411C213: snd_pcm_direct_initialize_slave (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x4115DFB: snd_pcm_dmix_open (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x4116286: _snd_pcm_dmix_open (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4384: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4A82: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4AF2: snd_pcm_open_slave (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x4100CEF: _snd_pcm_plug_open (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4384: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4A82: (within /usr/lib/libasound.so.2.0.0) ==14279== ==14279== Syscall param ioctl(arg) contains uninitialised byte(s) ==14279== at 0x40007F2: (within /lib/ld-2.6.1.so) ==14279== by 0x40DE7D1: snd_pcm_start (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x411C5DF: snd_pcm_direct_initialize_slave (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x4115DFB: snd_pcm_dmix_open (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x4116286: _snd_pcm_dmix_open (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4384: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4A82: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4AF2: snd_pcm_open_slave (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x4100CEF: _snd_pcm_plug_open (in /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4384: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x40E4A82: (within /usr/lib/libasound.so.2.0.0) ==14279== by 0x80A6E1B: pjmedia_snd_open (in /home/mathieu/BProject/WiFi-VoIP/pjsip/pjsip_host_x86/pjsip-apps/bin/pjsua-i686-pc-linux-gnu) Thank you, Mathieu Monney

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[alsa-devel] I changed cross-compiler to eabi. But alsa applications don't working.
by Jin-Young Park 12 Mar '08

12 Mar '08

Hello. Yesterday, I was changed cross-compiler to eabi. All components(Kernel, libraries and applications) was re-compiled, no problem. Almost all components are well working, but alsa application do not working. Any clue? Thanks. ------ error messages ------ # madplay-alsa --no-tty-control I.O.U.mp3 MPEG Audio Decoder 0.15.2 (beta) - Copyright (C) 2000-2004 Robert Leslie et al. Title: I.O.U Artist: Carry&Ron ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable Access type not available for playback: Invalid argument output: Invalid argument ------ error messages ------ # aplay -M -D hw:0,0 Torn.wav Playing WAVE 'Torn.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:896: Broken configuration for this PCM: no configurations available

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[alsa-devel] [patch 0/2] PNP: disable PNP motherboard resources that overlap PCI BARs
by Bjorn Helgaas 12 Mar '08

12 Mar '08

The following two patches fix several problems caused by BIOSes that report PNP motherboard device resources that overlap with PCI BARs. I think these should go in before 2.6.25 because some of these problems showed up in 2.6.24 when we increased PNP_MAX_MEM from 4 to 12, and there's no good way to work around them. Bjorn --

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[alsa-devel] aplay: can't resolve symbol 'snd_dlsym_start
by Pharaoh . 12 Mar '08

12 Mar '08

Hi, I have written an user space plugin and the plugin shared library is copied on board appropriately, libasound.so is also copied. When I try: #aplay -D omx test.wav //omx is my plugin I get following error: aplay: can't resolve symbol 'snd_dlsym_start Any idea what is the problem? -pharaoh

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Re: [alsa-devel] ICE1724 - File descriptor in bad state when capturing
by Clemens Ladisch 11 Mar '08

11 Mar '08

Pavel Hofman wrote: > Clemens Ladisch wrote: > > Pavel Hofman wrote: > > > Whereas for the broken analog input, the sequence is > > > > > > 1. snd_pcm_lib_read1 with SNDRV_PCM_STATE_PREPARED - returns 0 > > > 2. snd_pcm_lib_read1 with SNDRV_PCM_STATE_SETUP - returns -EBADFD > > > > I'd guess that your trigger callback fails somehow, bit this is > > hard to determine without the source code. > > Thanks, I will do more testing in the evening. Just in case, I am > including the relevant files. I moved all rate-relevant stuff to > card-specific routines, redefined in juli.c. The trigger code was not > changed at all. > > Perhaps those new snd_pcm_hardware structs in juli.c (I do not know the > exact meaning of the _bytes_ ones)? These values are in bytes (and not in frames). It's possible that the AK4114 code stops the stream because it thinks that the input sample rate has changed. Try uncommenting the printk in i2c/other/ak4114.c. Regards, Clemens

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[alsa-devel] ALSA Version change
by Chris Healy 11 Mar '08

11 Mar '08

Currently, I am using a 2.6.19.2 kernel with ALSA version 1.0.13. This is on an i.MX31 ARM11 platform. (This is what it came with.) We want to use a codec that currently has and ASoC supported driver so it seems best that we move forward to a newer version of ALSA. (1.0.14 or later has ASoC.) If we are going to the trouble of moving from 1.0.13 to a later version, does it make sense to port to 1.0.14 or 1.0.16? It makes sense that moving to 1.0.14 would take the least effort, but obviously, 1.0.16 has bug fixes and features. Do you think there is a large delta porting wise between moving to 1.0.16 instead of 1.0.14. (The assumption is that we have to move over to the later alsa-lib and alsa-utils, correct?) The other question we have is in reference to the max number of cards we are working with. This is an embedded system and we want to control 16 audio codecs, (cards). We won't be using the PCM interfaces of these cards, but we do need to control the codecs through the mixer interface. I have seen limitations previously alluding to a maximum of 8 cards but this was a few years ago. Is this true with the most recent versions, and if so, any ideas on what is necessary to change this? Thanks, Chris

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[alsa-devel] ICE1724 - File descriptor in bad state when capturing
by Pavel Hofman 11 Mar '08

11 Mar '08

Hi, I have most of the ESI Juli driver working, but I have hit a problem with the capture stream. I had to modify some ice1724.c code to allow redefining clock-related functions in the actual card file (Juli has a different clocking scheme), but all the DMA code is left intact, as well as the methods defining the capture stream. The ice1724 vu-meters detect input ADC data, I can hear correct sound when setting the internal mixer to H/W-In. Yet, when trying to record, I get arecord -v -D hw:0,0,0 -r 44100 -f S32_LE -c2 .... RIFF$WAVEfmt ... data arecord: pcm_read:1346: read error: File descriptor in bad state At the same time, my second Envy24-based card Prodigy192 (uses the original methods in ice1724.c) works OK, arecord gives no errors. Swapping the cards using the model parameter (to swap the /dev/snd/pcmCXDXc device file) makes no difference. Thank you for any hints of where I should focus my attention to. I am not attaching the patches, as they are rather complicated. If you find that important, I will send the preliminary version. Pavel Hofman.

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[alsa-devel] edirol m100fx
by J Brumbagh 11 Mar '08

11 Mar '08

I saw a post on the soundcard list to contact you if I have a Roland/Edirol M-100fx 10 channel mixer. I do have one and I would like to help out in anyway I can. The post seems to have been from Clemens Ladisch but I'm not sure if that matters. Please let me know how I can be of assistance. On another note I also have an internal sound card lspci lists as an ALi Corporation M5451 PCI AC-Link Controller Audio Device (rev 02)" This card has a headphone jack that seconds as a MiniTOSlink fiber optic jack. The sound card works fine however the fiber optic doesn't work. I think I do have windows drivers for this device on the OEM XP installation cd I have. I would greatly like to have this working properly and am also willing to help with the development. Thanks, Justin ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[alsa-devel] What does fifo_size mean in struct snd_pcm_hardware?
by Jin-Young Park 11 Mar '08

11 Mar '08

Hello. Struct snd_pcm_hardware have fifo_size field. The fifo place in audio interface(AC97, I2S) or in ALSA architecture? What does it meaning? I could not find it that related implementation in ALSA source code. Thanks.

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Re: [alsa-devel] X-Fi support in ALSA.
by Spike 11 Mar '08

11 Mar '08

> Hi, > > I have now had a chance to read some of the specs of the X-Fi cards. > aka. emu20k1. > It turns out that some of the X-Fi cards have a compatibility mode. > By default, they boot up in compatibility mode. > The interesting point is that this "compatibility mode" is in fact > the intel-HD standard so could use the intel-hda driver. > For one X-Fi card with IDs: > Device 1102:0009 Subsystem: 1102:0010 > works a bit with the ALSA hda-intel driver. > So far, the off the shelf hda-intel driver, with simple PCIs > adjustment, works for the above card, but only sound capture works so > far, so probably some quirk would be needed. > This compatibility mode works with the Windows Vista UAA drivers. > > Not all X-Fi cards support with compatibility mode. For example, any > card with (after a cold boot into Linux) > Device 1102:0005 Subsystem: xxxx:xxxx > will be X-Fi only. > > With regard to actual full X-Fi support, I am still in the progress of > creating a GPL emu20k1.h file, that is the first step towards an > snd-emu20k1 native ALSA driver. > > James Hello, That first ID is an X-Fi Xtreme Audio PCIe (CA0110-IBG) is it not? Not a real EMU20K1 based X-Fi, fully software driven, and from what I read its just a PCIe variation of the old CA0106. This thing will probably work with snd-ca0106 if creative kept to their old tactics regarding their low end X-Fi products. As for supporting the real X-Fi's, this is great news, its been a very longtime coming. One just has to question Creative's motives behind finally releasing the datasheets out of the blue like that after more than 2 years of trying to get them. Are these sheets under an NDA? Full support of an X-Fi will probably also need a firmware loader. If the PCIe CA0110-IBG doesn't share any common traits with a full blown EMU20K1, then I think the real X-Fi's should be focused on first as many more people are going to have these real X-Fi's that are useless to them at the moment, (about 3 years remember). I'll be around to help test code and can try anything you will require to help out supporting a real X-Fi card. 02:08.0 Multimedia audio controller: Creative Labs SB X-Fi [1102:0005] Subsystem: Unknown device 415a:0034 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at a400 [size=32] Memory at efc00000 (64-bit, non-prefetchable) [size=2M] Memory at e8000000 (64-bit, non-prefetchable) [size=64M] Bren

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[alsa-devel] Problems with nonblocking capture every 20ms
by Johannes Dohmen 10 Mar '08

10 Mar '08

Hi there, With a sampling rate of 8000 I want to capture every 20ms a chunk (160 frames) of data from the soundcard. That works ok, but every ~5 seconds I get a EGAIN from snd_pcm_readi(). I understand that this is not an error but a notification that the sound card is not ready and should be asked again for data later on. Probably the easiest way to get rid of the EGAINs would be to use blocking IO, which I would prefer, but I have timing restrictions and snd_pcm_readi() has no sort of timeout (which I could see at least). Additional Information: The reason for me the read from the sound card every 20ms is, that I need to do so for my voip-app (rtp&sip) which needs to send 160 frames every 20ms. The calculation is: PacketsPerSecond=50 SampleRate=8000 FramesPerPacket=8000/50=160 Wait=1000ms/50=20ms I know that EAGAINs might occur and I'm confident that I handle them ok on the RTP-Level (which is to notify the other side which then plays empty samples). Still I would guess, that I could receive much less EGAINs if I would setup the sound card better. If played around with the period time and size and buffer time and size, but honestly I did not really get the point of them and try&error has not helped me either. Thanks in advance & greetings from Germany Johannes

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Re: [alsa-devel] [regression] 2.6.25-rc4 snd-es18xx broken on Alpha
by Rene Herman 10 Mar '08

10 Mar '08

On 09-03-08 16:34, Ivan Kokshaysky wrote: > On Fri, Mar 07, 2008 at 09:58:28PM -0600, Bob Tracy wrote: >> (1) Using the sox-based "play" command to play a short ".wav" file >> results in the file being played approx. 1.5 times, i.e., it plays >> all the way through, then restarts from the beginning and quits >> somewhere in the middle of the second playback. At least the >> sound doesn't seem distorted... >> >> (2) Using "mpg123" results in silence at first, then after about 30 >> seconds, a tight loop of random repeating sound (doesn't sound like >> part of the mp3 file, but it's probably derived from it :-)) until >> the process is forcibly killed. > > Sounds like a DMA screwup. However, I'm pretty sure that it's not a > generic ALSA problem with ISA DMA, as my good old GUS MAX still works > perfectly on lx164 (machine that is very similar to your miata) as > I checked yesterday with patched 2.6.25-rc4. :-) > >> I have to hit <ctrl>C twice to interrupt playback. >> >> /proc/asound/cards looks normal: >> >> 0 [ES1888 ]: ES1888 - ESS AudioDrive ES1888 >> ESS AudioDrive ES1888 at 0x220, irq 5, dma1 1, dma2 5 > > I wonder if it still works on x86 - perhaps you ought to ask ALSA folks. > I'll search my machine room next week - I recall that there were a dozen > of old ISA soundcards, maybe there is some es18xx stuff, but chances > are slim... Missed the original mssage, adding alsa-devel. I have a few dozen es18xx's cards but no Alpha's. Am compiling a 2.6.25-rc4 onto an x86 test box now. Rene.

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[alsa-devel] [patch] dmix skipping first set of samples
by Mike Gorse 10 Mar '08

10 Mar '08

There was a change in alsa-lib 1.0.16 which looks like it was designed to make dmix skip samples in the case of underruns, but it causes the first sample to be skipped since dmix->slave_hw_ptr == dmix->slave_appl_ptr. The following patch fixes this and fixes a small typo in the comment: --- src/pcm/pcm_dmix.c.orig 2008-01-18 20:00:10.000000000 -0500 +++ src/pcm/pcm_dmix.c 2008-03-09 18:01:13.000000000 -0400 @@ -311,9 +311,9 @@ if (size >= pcm->boundary / 2) size = pcm->boundary - size; - /* the slave_app_ptr can be far behing the slave_hw_ptr */ + /* the slave_app_ptr can be far behind the slave_hw_ptr */ /* reduce mixing and errors here - just skip not catched writes */ - if (dmix->slave_hw_ptr < dmix->slave_appl_ptr) + if (dmix->slave_hw_ptr <= dmix->slave_appl_ptr) slave_size = dmix->slave_appl_ptr - dmix->slave_hw_ptr; else slave_size = dmix->slave_appl_ptr + (dmix->slave_boundary - dmix->slave_hw_ptr); Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

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[alsa-devel] [PATCH] hda: Reorganized DAC outputs.
by Matthew Ranostay 10 Mar '08

10 Mar '08

Changed so that internal speakers point to the Front mixer instead of Surround. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 3f01993fbc3d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Sat Mar 08 11:08:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Sat Mar 08 20:37:27 2008 -0500 @@ -538,8 +538,8 @@ static struct hda_verb dell_m6_core_init /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* setup audio connections */ - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},

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[alsa-devel] snd-hda-intel: Fix volume-wheel and microphone on Fujitsu-Siemens Amilo M3438G
by M G 10 Mar '08

10 Mar '08

Currently the default model that gets selected on this hardware is "F1734". Using this model neither the volume-wheel nor microphone (internal or external) work. The first part of my patch changes the default model to "fujitsu", this enables the volume wheel on the left side of the laptop. The second part changes the capture settings and thus makes it possible to record from microphones. Now when capturing from the external microphone I can hear myself talking but its quite quiet. I can improve this a little by raising the capture volume, but if I raise it above 40 I can no longer record anything. Questions: Is this patch any good or is there a better way to fix my problems? Why is the recording so quiet and why can't I record with capture value above 40? Patch: diff -r c4c5e6ced12f pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Wed Feb 06 15:48:06 2008 +0100 +++ b/pci/hda/patch_realtek.c Wed Feb 06 15:59:28 2008 +0100 @@ -2917,7 +2917,7 @@ static struct snd_pci_quirk alc880_cfg_t SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), @@ -3161,7 +3161,7 @@ static struct alc_config_preset alc880_p .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, + .input_mux = &alc880_6stack_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, .init_hook = alc880_uniwill_p53_hp_automute, },

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[alsa-devel] [PATCH] 2/2 alsa-python: API coverage documentation
by Aldrin Martoq 08 Mar '08

08 Mar '08

HG Log: ------- Added python coverage doc Description: ------------ This is the output of the alsa-python-coverage tool for the current HG repository. It is intented to be used by the alsa-python without the need to call the tool. Sample usage: ------------- $ grep ^STAT doc/COVERAGE $ grep ^N/A doc/COVERAGE Cheers!, -- Aldrin Martoq <amartoq(a)dcc.uchile.cl>

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Re: [alsa-devel] [Alsa-user] Newbie question - cannot get sound to work in Ubuntu
by Nigel Henry 07 Mar '08

07 Mar '08

On Wednesday 05 March 2008 23:48, Peter Toye wrote: > Nigel, > > Thanks. My comments are below. > > Best regards, > > Peter > mailto:alsa@ptoye.com > www.ptoye.com > > ------------------------- > > Tuesday, March 4, 2008, 6:43:43 PM, you wrote: > > Hi Peter. I've spent most of the afternoon (as I have nothing better to > > do) googling your problem, and there are a few things you can try. First > > though, do you have a live cd of Gutsy Gibbon, or Knoppix that you can > > bootup with, and see if you have the sounds working? > > That's very kind of you to take so much time. > > I booted up from a live Gutsy Gibbon CD (the one I used to create my > system) and it's quite interesting. Typing "alsamixer" in a terminal > session brings up the mixer OK! But with a seriously confusing set of > controls, only some of which seem to have much connection with the sound > card. And the "line in" column in the "capture" view doesn't have a fader > associated, which seems a bit odd. Maybe there's something I don't > understand here. > > And I can use the GNOME utilities to record and play back. Didn't have time > to try the command-line versions. Won't have time until Friday at the > earliest. > > > Now moving on to my googling stuff. > lspci -v output: > 00:0b.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1 (rev > 07) Subsystem: Creative Labs CT4830 SBLive! Value > Flags: bus master, medium devsel, latency 64, IRQ 10 > I/O ports at d400 [size=32] > Capabilities: [dc] Power Management version 1 > > 00:0b.1 Input device controller: Creative Labs SB Live! Game Port (rev 07) > Subsystem: Creative Labs Gameport Joystick > Flags: bus master, medium devsel, latency 64 > I/O ports at dc00 [size=8] > Capabilities: [dc] Power Management version 1 > > Alsamixer when opened shows the default card (card0), but have a look at > > the manpages for alsamixer. I now open these in a webbrowser as > > man:alsamixer. To start alsamixer for other cards you have the -c option. > > For example, alsamixer -c0 brings up the default cards mixer settings. > > alsamixer -c1 brings up the mixer settings for card 1, and so on. > > If I type alsamixer -c0 the mixer comes up with the same options as the CD > boot (see above). I thought that -c0 was the default, but it doesn't seem > to be. Hi Peter. It's getting a bit surreal now, as typing alsamixer, is ( under normal circumstances) the same as typing alsamixer -c0. For the sake of those on the alsa-devel list I've put the error you get when just typing alsamixer as user below. Typing "alsamixer" in a terminal window gives: "alsamixer: function snd_ctl_open failed for default: No such device" Whereas typing alsamixer -c0 opens alsamixer. At least you have access to the mixer settings now, although you say that some of the controls appear to be a bit bizarre. Would you download this script from: http://hg.alsa-project.org/alsa/raw-file/tip/alsa-info.sh There are a bunch of options now available for the script, so after making it executable, run it as user as below. ./alsa-info.sh --with-amixer --with-alsactl --with-configs The script will trawl out a bunch of stuff about your sounds, and upload it to pastebin. Just post back the link to the pastebin site when you reply. Not sure about an easy Alsa guide. All the best. Nigel.

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[alsa-devel] HD-audio dynamic configuration
by Takashi Iwai 07 Mar '08

07 Mar '08

Hi, time passed since the last discussion about where the HD-audio driver development to go to. I checked and tried to rework a bit the user-space configuration tool, and found out that it's really hard to keep it up to date. One of the big obstacles is that the HD-audio codecs often behave unexpectedly, in most cases, due to hardware bugs. For example, some codecs need the override of AMP caps and some needs rewrite of pin caps, etc. This is hard to catch with a sole generic code. It means, we need more or less codec-specific workarounds. Consequently, I believe it's easier to keep a little amount of codec-specific parser codes in the hda driver. This is of course hard-coded in the kernel. Then the goal is to reduce the amount of "patches" we have now for each specific device type. For this purpose, I've implemented a new feature -- allow hd-audio driver to reconfigure dynamically with additional hints. The hd-audio driver would have sysfs entries that can read/write the information dynamically and trigger the re-configuration of the parser. Right now, it's created as sysfs entries of each hwdep device, so that it can be bound to each codec chip. For example, suppose the card0 has codec#2. It corresponds to hwC0D2 device. /sys/class/sound/hwC0D2/ has the following new entries: 1. vendor_id, subsystem_id, revision_id, afg, mfg, 2. name, modelname 3. init-verbs, hints 4. reconfig, clear The files in 1 are the codec information that can be obtained from the codec chip. These are digits, and can be read *and* written. The write means the specific value will be overridden, and the driver handles as if the codec returns so. For example, if you have a new, unsupported codec chip that is compatible with others, you can change vendor_id value on the fly. The files in 2 are the strings that appear in the mixer, etc. The modelname is the model module option. This can be changed dynamically as well. The init-verbs and hints are write-only. You can pass the additional init verbs (e.g. for reprogramming the pincfg) and pass hints to the codec parser code. The hints are simply strings. Then you can write (any value) to "reconfig" to trigger the reconfiguration of the codec parser. Then all existing PCMs and mixer elements of this codec will be once removed and re-generated again with new init-verbs and hints. The "clear" is to clear the existing PCMs, mixer elements and init-verbs/hints entries. Now, the big missing piece is more clever codec parser code. Especially, the generic parser could be greatly improved with hints. A lot of works ahead, though :) For people who are interested, I put preliminary patches on my git tree here: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git For building the external modules, you'd need this build stub tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build-unsta… and run utils/setup-alsa-kernel script once (see INSTALL). The alsa-driver tarball snapshots are available at: http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ These experimental patches are in alsa-driver-unstable-snapshot tarball. Any comments appreciated. thanks, Takashi

1 0

Re: [alsa-devel] What recommended performance for ALSA(ASoC)?
by Jin-Young Park 07 Mar '08

07 Mar '08

Thanks for your reply. Your advises are very important information for me. Because explained your target system is very similar my target system. > I would like to say ALSA will perfectly runs on your target SOC clock. > I am running it on 192MHz ARM926 core along with H.264 video decoder!!!. > BogoMIPS = 95.84. > Even I had the same glitch problem before. Following debugging tips are very helpful. I have questions about debugging tips. > Few debug steps here: > 1. Check the DMA Configuration. (Transfer size, buffer size and number of Transfers) > 2. Does your period,period bytes ,buffer size of Ringbuffer matcheswith the DMA buffer? What is your ALSA driver setting that buffer bytes, period bytes, periods? Current my driver setting values are buffer bytes(4KBytes), period bytes(2KBytes), periods(2). >Check the correct re-initilization of "period" once the buffer is full. What does meannig? I don't understand. And what is proper function for checking buffer and period? prepare,trigger,irq handler, pointer... > 3. Check the current DMA postion at the point of "pointer" callback.This should be equal to number of samples send to the PCM middle layer. > 4. As Jaroslav says disc read speed should be > (44100*2*16) for 44.1K 16bit stereo. Or else read from NFSBOOT if your SOC support Ethernet. > 5. Always test with a "sinewave" for glitchs, not with your favoritesong J Tell me recommended music for audio test. Thanks.

1 0

[alsa-devel] OSS4 X-Fi driver port to ALSA
by William Pitcock 07 Mar '08

07 Mar '08

Hi, On Feburary 1, 4Front released their XFi driver to the public. Is anyone looking into porting this? If not, I will give it a try soon enough. It seems to not require any firmware at all too. William

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[alsa-devel] [PATCH] Fix for alsa-lib cross-compilation problems with ALSA_CONFIG_DIR and ALSA_PLUGIN_DIR
by Pawel MOLL 07 Mar '08

07 Mar '08

Fix for alsa-lib cross-compilation problems with ALSA_CONFIG_DIR and ALSA_PLUGIN_DIR "./configure" options for selecting ALSA configuration (default /usr/share/alsa) and plugin (/usr/lib/alsa-lib) directories introduced by alsa-hg/alsa-lib changeset 2284 cause problems with cross-compilation and packaging - there is no way to redefine them in runtime, during installation phase. This patch adds a level of indirection between constants and their usage - alsaconfigdir for ALSA_CONFIG_DIR and alsaplugindir for ALSA_PLUGIN_DIR - which can be redefined during "make install" stage. Signed-off-by: Pawel MOLL <pawel.moll(a)st.com> --- Index: alsa-lib-1.0.16/modules/mixer/simple/Makefile.am =================================================================== --- alsa-lib-1.0.16.orig/modules/mixer/simple/Makefile.am +++ alsa-lib-1.0.16/modules/mixer/simple/Makefile.am @@ -1,4 +1,5 @@ -pkglibdir = @ALSA_PLUGIN_DIR@/smixer +alsaplugindir = @ALSA_PLUGIN_DIR@ +pkglibdir = $(alsaplugindir)/smixer pythonlibs = @PYTHON_LIBS@ pythonincludes = @PYTHON_INCLUDES@ Index: alsa-lib-1.0.16/src/conf/Makefile.am =================================================================== --- alsa-lib-1.0.16.orig/src/conf/Makefile.am +++ alsa-lib-1.0.16/src/conf/Makefile.am @@ -10,5 +10,6 @@ endif EXTRA_DIST = $(cfg_files) -alsadir = @ALSA_CONFIG_DIR@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +alsadir = $(alsaconfigdir) alsa_DATA = $(cfg_files) Index: alsa-lib-1.0.16/src/conf/cards/Makefile.am =================================================================== --- alsa-lib-1.0.16.orig/src/conf/cards/Makefile.am +++ alsa-lib-1.0.16/src/conf/cards/Makefile.am @@ -1,4 +1,5 @@ -alsadir = @ALSA_CONFIG_DIR@/cards +alsaconfigdir = @ALSA_CONFIG_DIR@ +alsadir = $(alsaconfigdir)/cards cfg_files = aliases.conf \ AACI.conf \ ATIIXP.conf \ @@ -58,7 +59,7 @@ endif alsa_DATA = $(cfg_files) if BUILD_ALISP -SI7018dir = @ALSA_CONFIG_DIR@/cards/SI7018 +SI7018dir = $(alsaconfigdir)/cards/SI7018 SI7018_files = \ SI7018/sndoc-mixer.alisp \ SI7018/sndop-mixer.alisp Index: alsa-lib-1.0.16/src/conf/pcm/Makefile.am =================================================================== --- alsa-lib-1.0.16.orig/src/conf/pcm/Makefile.am +++ alsa-lib-1.0.16/src/conf/pcm/Makefile.am @@ -7,5 +7,6 @@ cfg_files = default.conf front.conf rear EXTRA_DIST = $(cfg_files) -alsadir = @ALSA_CONFIG_DIR@/pcm +alsaconfigdir = @ALSA_CONFIG_DIR@ +alsadir = $(alsaconfigdir)/pcm alsa_DATA = $(cfg_files)

2 1

Re: [alsa-devel] ALSACTL wrong restore on ARM9
by Liam Girdwood 07 Mar '08

07 Mar '08

On Fri, 2008-03-07 at 16:11 +0100, Alexandre BOUIN wrote: > Hi list, > > I'm using an intern AT91SAM92620-EK board with a WM8900 codec. > ALSACTL restores bad values to controls after doing this : > > $ rm /var/lib/alsa/asound.state > $ alsactl store > $ alsactl restore > > Anyone has seen this issue before ? > Any ideas to solve it ? (I don't know much these utils neither alsa API) > Could it come from our ASoC codec driver ? > Can you describe 'bad values' in a little more detail. Liam

2 1

[alsa-devel] ALSACTL wrong restore on ARM9
by Alexandre BOUIN 07 Mar '08

07 Mar '08

Hi list, I'm using an intern AT91SAM92620-EK board with a WM8900 codec. ALSACTL restores bad values to controls after doing this : $ rm /var/lib/alsa/asound.state $ alsactl store $ alsactl restore Anyone has seen this issue before ? Any ideas to solve it ? (I don't know much these utils neither alsa API) Could it come from our ASoC codec driver ? Thanks in advance. Regards, Alex.

1 0

[alsa-devel] M-Audio Delta Audiophile 192 again
by Computer AG im FSK 07 Mar '08

07 Mar '08

Hi, we didn't recieve an reply on our last email, so once again we are asking for support. Of course we know that everybody's time is small... We would like to know if someone will work on the bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3313 We repeat our offer of lending one card to someone who would like to fix it, like the reporter of the bug did. The GPL Version of the Open Sound System get this card working, so I hope it will not be that much work compared to other bugs ? If nobody will find time to fix it, a definetive answer would be nice so we can sell our 4 cards. But the probable loss of selling 4 card and buying 4 new ones could better be a donation for the alsa project ;) So please please help us out, we would be more than glad to give the money to the project than to loose it buying 4 new cards. Greetings, Toast

1 0

[alsa-devel] HDA Codecs: add support for Toshiba Equium L30
by Tobin Davis 07 Mar '08

07 Mar '08

This patch adds support for the Toshiba Equium L30 laptop and renames the mixer controls to match Laptop usages. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] What recommended performance for ALSA(ASoC)?
by Jinyoung Park 07 Mar '08

07 Mar '08

Hello, I tred developing new ALSA(ASoC) driver for our company's embedded processor. Our processor supported I2S and DMA. So, my ALSA drvier implementation using DMA. I was tested my ALSA driver. When I play wav file(sample rate 44100) using aplay( aplay -M -D hw:0,0 test.wav), sometimes occured breaks. Also, play mp3 file(sample rate 44100Hz) using madplay and mpg123, sometimes occured noises. So, I was tested wav file and mp3 file that down sampling 44100 -> 22050. They was not occurring break and decreasing noises. I tried improving my ALSA driver, optimize souce code and change variety buffer size and period size. But It doesn't improve much. In my guess, my target system is not enough performance because ALSA is heavy. Target system' processor(ARM9) clock is 200MHz and BogoMIPS is 99.6(cat /proc/cpuinfo). Linux kernel version is 2.6.24.1. What recommended performance for ALSA(ASoC)? If my target system is enough performance, shall I do that solve problem? Thanks.

3 2

[alsa-devel] [RFC] add a "private" field to the ac97 struct
by Sebastian Siewior 06 Mar '08

06 Mar '08

I have a PowerPC board with an AC97 controller and that one has an UCB1400 attached. Currently the UCB1400 driver discovers the interrupt via probing. This doesn't really work here. The resources are read from the device tree. I haven't found any other driver besides ucb1400_ts that uses the ac97_bus_type so I can't check where others get their HW information from. My proposal is to introduce a new field in struct snd_ac97 where the AC97 driver can leave the required HW info regarding the attached (in my case the interrupt for the ucb1400) before calling snd_card_register(). I attached just a single int field since I am not aware if multiple devices / resource are realistic / possible. In that case maybe a void * or struct platform_device would be a better choice. Signed-off-by: Sebastian Siewior <sebastian(a)breakpoint.cc> --- drivers/input/touchscreen/ucb1400_ts.c | 13 ++++++++----- include/sound/ac97_codec.h | 2 ++ 2 files changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/input/touchscreen/ucb1400_ts.c b/drivers/input/touchscreen/ucb1400_ts.c index 607f993..95d1fea 100644 --- a/drivers/input/touchscreen/ucb1400_ts.c +++ b/drivers/input/touchscreen/ucb1400_ts.c @@ -492,11 +492,14 @@ static int ucb1400_ts_probe(struct device *dev) goto err_free_devs; } - error = ucb1400_detect_irq(ucb); - if (error) { - printk(KERN_ERR "UCB1400: IRQ probe failed\n"); - goto err_free_devs; - } + if (!ucb->ac97->device_irq) { + error = ucb1400_detect_irq(ucb); + if (error) { + printk(KERN_ERR "UCB1400: IRQ probe failed\n"); + goto err_free_devs; + } + } else + ucb->irq = ucb->ac97->device_irq; error = request_irq(ucb->irq, ucb1400_hard_irq, IRQF_TRIGGER_RISING, "UCB1400", ucb); diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 0148058..bf920a2 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -474,6 +474,8 @@ struct snd_ac97 { struct snd_ac97_build_ops * build_ops; void *private_data; void (*private_free) (struct snd_ac97 *ac97); + /* device attached to the AC97 bus */ + int device_irq; /* --- */ struct snd_ac97_bus *bus; struct pci_dev *pci; /* assigned PCI device - used for quirks */ -- 1.5.3.5

3 3

[alsa-devel] snd_pcm_open device names
by Reiner 06 Mar '08

06 Mar '08

with snd_pcm_open (handle, "default", SND_PCM_STREAM_PLAYBACK, 0) I can open a device twice with snd_pcm_open (handle, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0) the 2. call fails with error -16. The docu says: The default device is equal to plug plugin with hw plugin as slave. ???? Reiner

3 3

[alsa-devel] [PATCH] Support for playing WAV files with "extensible format" header using aplay.
by Pawel MOLL 06 Mar '08

06 Mar '08

Support for playing WAV files with "extensible format" header using aplay. WAV files with more than 2 channels or with more than 16 bits per samples can be saved with "extensible format" chunk (see http://msdn2.microsoft.com/en-us/library/ms713496(VS.85).aspx). For instance, sox, when converting data to 24- or 32-bits per sample format uses this format, and aplay was unable to play such file. Now the problem is solved :-) Signed-off-by: Pawel MOLL <pawel.moll(a)st.com> --- diff -r 9c822ff3e9b9 aplay/aplay.c --- a/aplay/aplay.c Thu Feb 21 09:12:53 2008 +0100 +++ b/aplay/aplay.c Thu Mar 06 14:25:52 2008 +0000 @@ -745,16 +745,29 @@ static ssize_t test_wavefile(int fd, u_c check_wavefile_space(buffer, len, blimit); test_wavefile_read(fd, buffer, &size, len, __LINE__); f = (WaveFmtBody*) buffer; + if (LE_SHORT(f->format) == WAV_FMT_EXTENSIBLE) { + WaveFmtExtensibleBody *fe = (WaveFmtExtensibleBody*)buffer; + if (len < sizeof(WaveFmtExtensibleBody)) { + error(_("unknown length of extensible 'fmt ' chunk (read %u, should be %u at least)"), + len, (u_int)sizeof(WaveFmtExtensibleBody)); + exit(EXIT_FAILURE); + } + if (memcmp(fe->guid_tag, WAV_GUID_TAG, 14) != 0) { + error(_("wrong format tag in extensible 'fmt ' chunk")); + exit(EXIT_FAILURE); + } + f->format = fe->guid_format; + } if (LE_SHORT(f->format) != WAV_FMT_PCM && LE_SHORT(f->format) != WAV_FMT_IEEE_FLOAT) { error(_("can't play WAVE-file format 0x%04x which is not PCM or FLOAT encoded"), LE_SHORT(f->format)); exit(EXIT_FAILURE); } - if (LE_SHORT(f->modus) < 1) { - error(_("can't play WAVE-files with %d tracks"), LE_SHORT(f->modus)); - exit(EXIT_FAILURE); - } - hwparams.channels = LE_SHORT(f->modus); + if (LE_SHORT(f->channels) < 1) { + error(_("can't play WAVE-files with %d tracks"), LE_SHORT(f->channels)); + exit(EXIT_FAILURE); + } + hwparams.channels = LE_SHORT(f->channels); switch (LE_SHORT(f->bit_p_spl)) { case 8: if (hwparams.format != DEFAULT_FORMAT && @@ -1805,7 +1818,7 @@ static void begin_wave(int fd, size_t cn f.format = LE_SHORT(WAV_FMT_IEEE_FLOAT); else f.format = LE_SHORT(WAV_FMT_PCM); - f.modus = LE_SHORT(hwparams.channels); + f.channels = LE_SHORT(hwparams.channels); f.sample_fq = LE_INT(hwparams.rate); #if 0 tmp2 = (samplesize == 8) ? 1 : 2; diff -r 9c822ff3e9b9 aplay/formats.h --- a/aplay/formats.h Thu Feb 21 09:12:53 2008 +0100 +++ b/aplay/formats.h Thu Mar 06 14:25:52 2008 +0000 @@ -69,6 +69,10 @@ typedef struct voc_ext_block { #define WAV_FMT_PCM 0x0001 #define WAV_FMT_IEEE_FLOAT 0x0003 #define WAV_FMT_DOLBY_AC3_SPDIF 0x0092 +#define WAV_FMT_EXTENSIBLE 0xfffe + +/* Used with WAV_FMT_EXTENSIBLE format */ +#define WAV_GUID_TAG "\x00\x00\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71" /* it's in chunks like .voc and AMIGA iff, but my source say there are in only in this combination, so I combined them in one header; @@ -81,13 +85,22 @@ typedef struct { } WaveHeader; typedef struct { - u_short format; /* should be 1 for PCM-code */ - u_short modus; /* 1 Mono, 2 Stereo */ + u_short format; /* see WAV_FMT_* */ + u_short channels; u_int sample_fq; /* frequence of sample */ u_int byte_p_sec; u_short byte_p_spl; /* samplesize; 1 or 2 bytes */ u_short bit_p_spl; /* 8, 12 or 16 bit */ } WaveFmtBody; + +typedef struct { + WaveFmtBody format; + u_short ext_size; + u_short bit_p_spl; + u_int channel_mask; + u_short guid_format; /* WAV_FMT_* */ + u_char guid_tag[14]; /* WAV_GUID_TAG */ +} WaveFmtExtensibleBody; typedef struct { u_int type; /* 'data' */

2 1

[alsa-devel] Problems with ALSA in embedded device
by Imanol Trapote 06 Mar '08

06 Mar '08

Hi all, I am developing a product in an embedded system that uses ALSA library. This system can't use the dynamic linker, I am using flat binaries. I use this configuration options: --without-libdl \ --with-pthread \ --with-softload \ --disable-alisp \ --disable-hwdep \ --disable-seq \ --disable-resmgr \ --disable-rawmidi \ --disable-old-symbols \ --disable-python \ --disable-mixer I try to use the same configuration options in a PC for debugging, and I get the same problem, when I try to list devices with aplay, I get this: aplay -L ALSA lib conf.c:2699:(snd_config_hooks_call) Cannot open shared library (null) ALSA lib conf.c:3076:(snd_config_update_r) hooks failed, removing configuration It's possible use ALSA without dynamic libraries, with this configuration Thanks in advance. Imanol Trapote ----------------------------------------- PLEASE NOTE ------------------------------------------- This message, along with any attachments, may be confidential or legally privileged. It is intended only for the named person(s), who is/are the only authorized recipients. If this message has reached you in error, kindly destroy it without review and notify the sender immediately. Thank you for your help. ZIV uses virus scanning software but excludes any liability for viruses contained in any attachment. ------------------------------------ ROGAMOS LEA ESTE TEXTO ------------------------------- Este mensaje y sus anexos pueden contener información confidencial y/o con derecho legal. Está dirigido únicamente a la/s persona/s o entidad/es reseñadas como único destinatario autorizado. Si este mensaje le hubiera llegado por error, por favor elimínelo sin revisarlo ni reenviarlo y notifíquelo inmediatamente al remitente. Gracias por su colaboración. ZIV utiliza software antivirus, pero no se hace responsable de los virus contenidos en los ficheros anexos.

2 3

[alsa-devel] [PATCH] hda_intel: Add the DIDs of nvidia MCP79 HD audio controller to hda_intel.c
by peerchen 06 Mar '08

06 Mar '08

Add the Device IDs of nvidia MCP79 HD audio controller to hda_intel.c Signed-off-by: Peer Chen <peerchen(a)gmail.com> --- --- linux-2.6.25-rc4/sound/pci/hda/hda_intel.c.orig 2008-03-04 21:51:07.000000000 -0500 +++ linux-2.6.25-rc4/sound/pci/hda/hda_intel.c 2008-03-04 21:53:43.000000000 -0500 @@ -2036,6 +2036,10 @@ static struct pci_device_id azx_ids[] = { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0bd4, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP7B */ + { 0x10de, 0x0bd5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP7B */ + { 0x10de, 0x0bd6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP7B */ + { 0x10de, 0x0bd7, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP7B */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -

2 1

[alsa-devel] How to do pause/resume in ALSA application
by Aggarwal, Anuj 06 Mar '08

06 Mar '08

Hi, I am using the snd_pcm_pause API in my application but it is returning error: Invalid argument. I am passing the PCM handle of the playback stream which I opened earlier and using to play a WAV file. Although I am able to hear the WAV file correctly, I am not able to pause. I checked whether the pause functionality is supported, using snd_pcm_hw_params_can_pause() and it returned me success. Any help is welcome. Thanks & Regards, Anuj Aggarwal

2 3

[alsa-devel] aplay error - cant resolve symbol alsa_lisp_init
by Pharaoh . 06 Mar '08

06 Mar '08

Hi, I am trying to run aplay on my board, I have got alsa-lib and alsa-utils on board and when I try to run aplay, I get following. # aplay -h Dec 31 16:04:42 kernel: Error -5 while decompressing! Dec 31 16:04:42 kernel: c0234d4a(1911)->c35e3000(4096) aplay: can't resolve symbol 'alsa_lisp_init_objects' Dec 31 16:04:42 kernel: aplay used greatest stack depth: 5676 bytes left # aplay --help aplay: can't resolve symbol 'alsa_lisp_init_objects' Any clue? -pharaoh.

2 1

[alsa-devel] [PATCH 5/5] asihpi: add hwdep (experimental)
by Eliot Blennerhassett 06 Mar '08

06 Mar '08

Add hpi ioctl via hwdep (experimental, disabled by default). Adjust sampleclock control to use new hpi apis. Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com --- diff -r 6def4892d3f5 pci/Kconfig --- a/pci/Kconfig Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/Kconfig Sat Feb 23 20:24:59 2008 +1300 @@ -17,6 +17,7 @@ config SND_ASIHPI depends on SND && X86 select FW_LOADER select SND_PCM + select SND_HWDEP help Say 'Y' or 'M' to include support for AudioScience ASIxxxx soundcards. diff -r 6def4892d3f5 pci/asihpi/asihpi.c --- a/pci/asihpi/asihpi.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/asihpi.c Wed Mar 05 21:59:31 2008 +1300 @@ -23,6 +23,9 @@ */ /* >0: print Hw params, timer vars. >1: print stream write/copy sizes */ #define REALLY_VERBOSE_LOGGING 0 + +/* Experimental HPI hwdep interface */ +#define EXPERIMENTAL_HWDEP 0 #if REALLY_VERBOSE_LOGGING #define VPRINTK1 printk @@ -66,19 +69,28 @@ #include <sound/info.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/hwdep.h> #include "hpi.h" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; - -module_param_array(index, int, NULL, 0444); +static int enable_hpi_hwdep; + +module_param_array(index, int, NULL, S_IRUGO); MODULE_PARM_DESC(index, "ALSA Index value for AudioScience soundcard."); -module_param_array(id, charp, NULL, 0444); + +module_param_array(id, charp, NULL, S_IRUGO); MODULE_PARM_DESC(id, "ALSA ID string for AudioScience soundcard."); -module_param_array(enable, bool, NULL, 0444); + +module_param_array(enable, bool, NULL, S_IRUGO); MODULE_PARM_DESC(enable, "ALSA Enable AudioScience soundcard."); + +module_param(enable_hpi_hwdep, bool, S_IRUGO|S_IWUSR); +MODULE_PARM_DESC(enable_hpi_hwdep, + "ALSA Enable HPI hwdep for AudioScience soundcard " + "(experimental)"); /* used by hpimod.c, stop sparse from complaining */ int snd_asihpi_bind(struct hpi_adapter *hpi_card); @@ -114,6 +126,7 @@ struct clk_source { struct clk_cache { int count; + int has_local; struct clk_source s[MAX_CLOCKSOURCES]; }; @@ -1602,7 +1615,6 @@ static int __devinit snd_asihpi_aesebu_t snd_control->info = snd_asihpi_aesebu_format_info; snd_control->get = snd_asihpi_aesebu_tx_format_get; snd_control->put = snd_asihpi_aesebu_tx_format_put; - snd_control->index = asihpi_control->wDstNodeIndex + ixb; asihpi_ctl_name(snd_control, asihpi_control, "Format"); @@ -1909,9 +1921,9 @@ static int log2lin[] = { 6790940, 2147484, /* -60dB */ 679094, - 214748, + 214748, /* -80 */ 67909, - 21475, + 21475, /* -100 */ 6791, 2147, 679, @@ -2153,8 +2165,9 @@ static void __devinit snd_asihpi_cmode_n ------------------------------------------------------------*/ static char *sampleclock_sources[MAX_CLOCKSOURCES] = - { "N/A", "Adapter", "AES/EBU Sync", "Word External", "Word Header", - "SMPTE", "AES/EBU In1", "Auto", "Cobranet", + { "N/A", "Local PLL", "AES/EBU Sync", "Word External", "Word Header", + "SMPTE", "AES/EBU In1", "Auto", "Network", "Invalid", + "Prev Module", "AES/EBU In2", "AES/EBU In3", "AES/EBU In4", "AES/EBU In5", "AES/EBU In6", "AES/EBU In7", "AES/EBU In8"}; @@ -2239,7 +2252,9 @@ static void __devinit snd_asihpi_clksrc_ snd_control->info = snd_asihpi_clksrc_info; snd_control->get = snd_asihpi_clksrc_get; snd_control->put = snd_asihpi_clksrc_put; - asihpi_ctl_name(snd_control, asihpi_control, "ClockSource"); + asihpi_ctl_name(snd_control, asihpi_control, "Source"); + + clkcache->has_local = 0; for (i = 0; i <= HPI_SAMPLECLOCK_SOURCE_LAST; i++) { if (HPI_ControlQuery(phSubSys, hSC, @@ -2250,6 +2265,8 @@ static void __devinit snd_asihpi_clksrc_ clkcache->s[i].name = sampleclock_sources[wSource]; if (wSource == HPI_SAMPLECLOCK_SOURCE_AESEBU_INPUT) hasAesIn = 1; + if (wSource == HPI_SAMPLECLOCK_SOURCE_LOCAL) + clkcache->has_local = 1; } if (hasAesIn) /* already will have picked up index 0 above */ @@ -2271,7 +2288,7 @@ static void __devinit snd_asihpi_clksrc_ /*------------------------------------------------------------ Clkrate controls ------------------------------------------------------------*/ -static int snd_asihpi_clkrate_info(struct snd_kcontrol *kcontrol, +static int snd_asihpi_clklocal_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -2283,18 +2300,18 @@ static int snd_asihpi_clkrate_info(struc return 0; } -static int snd_asihpi_clkrate_get(struct snd_kcontrol *kcontrol, +static int snd_asihpi_clklocal_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { u32 hControl = kcontrol->private_value; u32 dwRate; - HPI_SampleClock_GetSampleRate(phSubSys, hControl, &dwRate); + HPI_SampleClock_GetLocalRate(phSubSys, hControl, &dwRate); ucontrol->value.integer.value[0] = dwRate; return 0; } -static int snd_asihpi_clkrate_put(struct snd_kcontrol *kcontrol, +static int snd_asihpi_clklocal_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int change; @@ -2304,21 +2321,55 @@ static int snd_asihpi_clkrate_put(struct asihpi->mixer_clkrate[addr][1] != right; */ change = 1; - HPI_SampleClock_SetSampleRate(phSubSys, hControl, + HPI_SampleClock_SetLocalRate(phSubSys, hControl, ucontrol->value.integer.value[0]); return change; } +static void __devinit snd_asihpi_clklocal_new( + struct hpi_control *asihpi_control, + struct snd_kcontrol_new *snd_control) +{ + snd_control->info = snd_asihpi_clklocal_info; + snd_control->get = snd_asihpi_clklocal_get; + snd_control->put = snd_asihpi_clklocal_put; + + asihpi_ctl_name(snd_control, asihpi_control, "LocalRate"); +} + +static int snd_asihpi_clkrate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 8000; + uinfo->value.integer.max = 192000; + uinfo->value.integer.step = 100; + + return 0; +} + +static int snd_asihpi_clkrate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 hControl = kcontrol->private_value; + u32 dwRate; + + HPI_SampleClock_GetSampleRate(phSubSys, hControl, &dwRate); + ucontrol->value.integer.value[0] = dwRate; + return 0; +} + static void __devinit snd_asihpi_clkrate_new(struct hpi_control *asihpi_control, struct snd_kcontrol_new *snd_control) { snd_control->info = snd_asihpi_clkrate_info; snd_control->get = snd_asihpi_clkrate_get; - snd_control->put = snd_asihpi_clkrate_put; + snd_control->access = + SNDRV_CTL_ELEM_ACCESS_VOLATILE | SNDRV_CTL_ELEM_ACCESS_READ; asihpi_ctl_name(snd_control, asihpi_control, "Rate"); } - /*------------------------------------------------------------ Mixer ------------------------------------------------------------*/ @@ -2404,6 +2455,14 @@ static int __devinit snd_card_asihpi_mix return err; snd_asihpi_clkrate_new(&asihpi_control, &snd_control); + err = snd_ctl_add(card, + snd_ctl_new1(&snd_control, asihpi)); + if (err < 0) + return err; + if (asihpi->cc.has_local) + snd_asihpi_clklocal_new(&asihpi_control, + &snd_control); + break; case HPI_CONTROL_CONNECTION: /* ignore these */ continue; @@ -2510,6 +2569,55 @@ static void __devinit snd_asihpi_proc_in } /*------------------------------------------------------------ + HWDEP + ------------------------------------------------------------*/ + +static int snd_asihpi_hpi_open(struct snd_hwdep * hw, struct file *file) +{ + return 0; +} + +static int snd_asihpi_hpi_release(struct snd_hwdep * hw, struct file *file) +{ + return 0; +} + +long asihpi_hpi_ioctl(struct file *file, + unsigned int cmd, unsigned long arg); + +static int snd_asihpi_hpi_ioctl(struct snd_hwdep * hw, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return asihpi_hpi_ioctl(file, cmd, arg); +} + + +/* results in /dev/snd/hwC#D0 file for each card with index # + also /proc/asound/hwdep will contain '#-00: asihpi (HPI) for each card' +*/ +int __devinit snd_asihpi_hpi_new(struct snd_card_asihpi *asihpi, int device, + struct snd_hwdep ** rhwdep) +{ + struct snd_hwdep *hw; + int err; + + if (rhwdep) + *rhwdep = NULL; + err = snd_hwdep_new(asihpi->card, "HPI", device, &hw); + if (err < 0) + return err; + strcpy(hw->name, "asihpi (HPI)"); + hw->iface = SNDRV_HWDEP_IFACE_LAST; + hw->ops.open = snd_asihpi_hpi_open; + hw->ops.ioctl = snd_asihpi_hpi_ioctl; + hw->ops.release = snd_asihpi_hpi_release; + hw->private_data = asihpi; + if (rhwdep) + *rhwdep = hw; + return 0; +} + +/*------------------------------------------------------------ CARD ------------------------------------------------------------*/ int __devinit snd_asihpi_bind(struct hpi_adapter *hpi_card) @@ -2602,10 +2710,13 @@ int __devinit snd_asihpi_bind(struct hpi HPI_CONTROL_SAMPLECLOCK, &hControl); if (!err) - err = HPI_SampleClock_SetSampleRate(phSubSys, + err = HPI_SampleClock_SetLocalRate(phSubSys, hControl, adapter_fs); snd_asihpi_proc_init(asihpi); + + if (enable_hpi_hwdep) + snd_asihpi_hpi_new(asihpi, 0, NULL); if (asihpi->support_mmap) strcpy(card->driver, "ASIHPI-MMAP"); diff -r 6def4892d3f5 pci/asihpi/hpimod.c --- a/pci/asihpi/hpimod.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpimod.c Wed Mar 05 21:59:31 2008 +1300 @@ -24,14 +24,10 @@ Linux HPI driver module #include "hpi.h" #include "hpidebug.h" #include "hpimsgx.h" + #include <linux/fs.h> #include <linux/slab.h> #include <linux/moduleparam.h> -#include <linux/pci.h> -#include <linux/device.h> -#include <linux/interrupt.h> -#include <linux/mutex.h> -#include <linux/version.h> #include <asm/uaccess.h> #include <linux/stringify.h> @@ -167,13 +163,13 @@ static int hpi_release( } #if LINUX_VERSION_CODE > KERNEL_VERSION(2, 6, 11) -static long hpi_ioctl( +long asihpi_hpi_ioctl( struct file *file, unsigned int cmd, unsigned long arg ) #else -static int hpi_ioctl( +static int asihpi_hpi_ioctl( struct inode *inode, struct file *file, unsigned int cmd, @@ -225,8 +221,18 @@ static int hpi_ioctl( /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; - int nAdapter = hm.wAdapterIndex; + u32 nAdapter = hm.wAdapterIndex; switch (hm.wFunction) { + case HPI_SUBSYS_CREATE_ADAPTER: + case HPI_SUBSYS_DELETE_ADAPTER: + /* Application must not use these functions! */ + hr.wSize = sizeof(struct hpi_response_header); + hr.wError = HPI_ERROR_INVALID_OPERATION; + hr.wFunction = hm.wFunction; + uncopied_bytes = copy_to_user(phr, &hr, hr.wSize); + if (uncopied_bytes) + return -EFAULT; + return 0; case HPI_OSTREAM_WRITE: case HPI_ISTREAM_READ: /* Yes, sparse, this is correct. */ @@ -252,60 +258,49 @@ static int hpi_ioctl( break; } - if ((nAdapter >= HPI_MAX_ADAPTERS || nAdapter < 0) && - (hm.wObject != HPI_OBJ_SUBSYSTEM)) - hr.wError = HPI_ERROR_INVALID_OBJ_INDEX; - else { - if (mutex_lock_interruptible(&adapters[nAdapter]. - mutex)) - return -EINTR; - - if (wrflag == 0) { - if (size > bufsize) { - mutex_unlock(&adapters[nAdapter]. - mutex); - HPI_DEBUG_LOG(ERROR, - "Requested transfer of %d " - "bytes, max buffer size " - "is %d bytes.\n", size, - bufsize); - return -EINVAL; - } - - uncopied_bytes = - copy_from_user(pa->pBuffer, ptr, - size); - if (uncopied_bytes) - HPI_DEBUG_LOG(WARNING, - "Missed %d of %d " - "bytes from user\n", - uncopied_bytes, size); + if (mutex_lock_interruptible(&adapters[nAdapter].mutex)) + return -EINTR; + + if (wrflag == 0) { + if (size > bufsize) { + mutex_unlock(&adapters[nAdapter].mutex); + HPI_DEBUG_LOG(ERROR, + "Requested transfer of %d " + "bytes, max buffer size " + "is %d bytes.\n", size, bufsize); + return -EINVAL; } - HPI_MessageF(&hm, &hr, file); - - if (wrflag == 1) { - if (size > bufsize) { - mutex_unlock(&adapters[nAdapter]. - mutex); - HPI_DEBUG_LOG(ERROR, - "Requested transfer of %d " - "bytes, max buffer size is " - "%d bytes.\n", size, bufsize); - return -EINVAL; - } - - uncopied_bytes = - copy_to_user(ptr, pa->pBuffer, size); - if (uncopied_bytes) - HPI_DEBUG_LOG(WARNING, - "Missed %d of %d " - "bytes to user\n", - uncopied_bytes, size); + uncopied_bytes = + copy_from_user(pa->pBuffer, ptr, size); + if (uncopied_bytes) + HPI_DEBUG_LOG(WARNING, + "Missed %d of %d " + "bytes from user\n", + uncopied_bytes, size); + } + + HPI_MessageF(&hm, &hr, file); + + if (wrflag == 1) { + if (size > bufsize) { + mutex_unlock(&adapters[nAdapter].mutex); + HPI_DEBUG_LOG(ERROR, + "Requested transfer of %d " + "bytes, max buffer size is " + "%d bytes.\n", size, bufsize); + return -EINVAL; } - mutex_unlock(&adapters[nAdapter].mutex); + uncopied_bytes = copy_to_user(ptr, pa->pBuffer, size); + if (uncopied_bytes) + HPI_DEBUG_LOG(WARNING, + "Missed %d of %d " + "bytes to user\n", + uncopied_bytes, size); } + + mutex_unlock(&adapters[nAdapter].mutex); } /* on return response size must be set */ @@ -322,9 +317,9 @@ static struct file_operations hpi_fops = static struct file_operations hpi_fops = { .owner = THIS_MODULE, #if LINUX_VERSION_CODE > KERNEL_VERSION(2, 6, 11) - .unlocked_ioctl = hpi_ioctl, + .unlocked_ioctl = asihpi_hpi_ioctl, #else - .ioctl = hpi_ioctl, + .ioctl = asihpi_hpi_ioctl, #endif .open = hpi_open, .release = hpi_release @@ -399,6 +394,9 @@ static int __devinit adapter_probe( /* call CreateAdapterObject on the relevant hpi module */ HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); + if (hr.wError) + goto err; + adapter.pBuffer = vmalloc(bufsize); if (adapter.pBuffer == NULL) { HPI_DEBUG_LOG(ERROR, @@ -407,39 +405,37 @@ static int __devinit adapter_probe( goto err; } - if (hr.wError == 0) { - adapter.index = hr.u.s.wAdapterIndex; - adapter.type = hr.u.s.awAdapterList[adapter.index]; + adapter.index = hr.u.s.wAdapterIndex; + adapter.type = hr.u.s.awAdapterList[adapter.index]; + hm.wAdapterIndex = adapter.index; + + err = HPI_AdapterOpen(NULL, adapter.index); + if (err) + goto err; + + adapter.snd_card_asihpi = NULL; + /* WARNING can't init sem in 'adapter' + * and then copy it to adapters[] ?!?! + */ + adapters[hr.u.s.wAdapterIndex] = adapter; + mutex_init(&adapters[adapter.index].mutex); +#ifdef ALSA_BUILD + if (snd_asihpi_bind(&adapters[adapter.index])) { + HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, + HPI_SUBSYS_DELETE_ADAPTER); hm.wAdapterIndex = adapter.index; - - err = HPI_AdapterOpen(NULL, adapter.index); - if (err) - goto err; - - adapter.snd_card_asihpi = NULL; - /* WARNING can't init sem in 'adapter' - * and then copy it to adapters[] ?!?! - */ - adapters[hr.u.s.wAdapterIndex] = adapter; - mutex_init(&adapters[adapter.index].mutex); -#ifdef ALSA_BUILD - if (snd_asihpi_bind(&adapters[adapter.index])) { - HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_DELETE_ADAPTER); - hm.wAdapterIndex = adapter.index; - HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); - goto err; - } -#endif - - pci_set_drvdata(pci_dev, &adapters[adapter.index]); - adapter_count++; - - printk(KERN_INFO - "Probe found adapter ASI%04X HPI index #%d.\n", - adapter.type, adapter.index); - return 0; - } + HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); + goto err; + } +#endif + + pci_set_drvdata(pci_dev, &adapters[adapter.index]); + adapter_count++; + + printk(KERN_INFO + "Probe found adapter ASI%04X HPI index #%d.\n", + adapter.type, adapter.index); + return 0; err: for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) {

4 6

[alsa-devel] [PATCH] model for cx20549 to support laptop HP530
by Jiang zhe 06 Mar '08

06 Mar '08

Hi, Please refer to [0003821]. Currently the model laptop-hpsense use the 0x12 as ExtMic, and use 0x14 as Internal IntMic. But the hp530 only have one ExtMic, the Pin widget is 0x14. In this patch, I changed the mixer item for them. I still reserved the IntMic item, it will be helpful if other machine may use this model. Best regards! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20080306/alsa-kernel/pci/hda/patch_conexant.c 2008-03-01 09:00:07.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_conexant.c 2008-03-06 16:12:23.000000000 +0800 @@ -519,6 +519,14 @@ static struct hda_input_mux cxt5045_capt } }; +static struct hda_input_mux cxt5045_capture_source_hp530 = { + .num_items = 2, + .items = { + { "ExtMic", 0x1 }, + { "IntMic", 0x2 }, + } +}; + /* turn on/off EAPD (+ mute HP) as a master switch */ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -647,6 +655,37 @@ static struct snd_kcontrol_new cxt5045_b {} }; +static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x1, HDA_INPUT), + HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5045_hp_master_sw_put, + .private_value = 0x10, + }, + + {} +}; + static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -841,6 +880,7 @@ enum { CXT5045_LAPTOP_MICSENSE, CXT5045_LAPTOP_HPMICSENSE, CXT5045_BENQ, + CXT5045_LAPTOP_HP530, #ifdef CONFIG_SND_DEBUG CXT5045_TEST, #endif @@ -852,6 +892,7 @@ static const char *cxt5045_models[CXT504 [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", [CXT5045_BENQ] = "benq", + [CXT5045_LAPTOP_HP530] = "laptop-hp530", #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif @@ -865,7 +906,7 @@ static struct snd_pci_quirk cxt5045_cfg_ SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -949,6 +990,14 @@ static int patch_cxt5045(struct hda_code spec->num_mixers = 2; codec->patch_ops.init = cxt5045_init; break; + case CXT5045_LAPTOP_HP530: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; + spec->input_mux = &cxt5045_capture_source_hp530; + spec->num_init_verbs = 2; + spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; + spec->mixers[0] = cxt5045_mixers_hp530; + codec->patch_ops.init = cxt5045_init; + break; #ifdef CONFIG_SND_DEBUG case CXT5045_TEST: spec->input_mux = &cxt5045_test_capture_source;

2 1

[alsa-devel] [PATCH] model for alc883 to support FUJITSU Pi2515
by Jiang zhe 06 Mar '08

06 Mar '08

Hi, Please refer to [0003800] on the alsa mantis. There is no suitable model for Pi2515. This model is to support it. Thanks! Best regards! Jiang,zhe Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20080306/alsa-kernel/pci/hda/patch_realtek.c 2008-03-05 09:00:08.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-03-06 15:20:46.000000000 +0800 @@ -202,6 +202,7 @@ enum { ALC888_6ST_DELL, ALC883_MITAC, ALC883_CLEVO_M720R, + ALC883_FUJITSU_PI2515, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6484,6 +6485,14 @@ static struct hda_input_mux alc883_lenov }, }; +static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Int Mic", 0x1 }, + }, +}; + #define alc883_mux_enum_info alc_mux_enum_info #define alc883_mux_enum_get alc_mux_enum_get /* ALC883 has the ALC882-type input selection */ @@ -6673,6 +6682,33 @@ static struct snd_kcontrol_new alc883_cl { } /* end */ }; +static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7220,6 +7256,20 @@ static struct hda_verb alc883_clevo_m720 { } /* end */ }; +static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { + /* HP */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7423,6 +7473,25 @@ static void alc883_clevo_m720r_unsol_eve alc883_clevo_m720r_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_2ch_fujitsu_pi2515_automute(codec); +} + static void alc883_haier_w66_automute(struct hda_codec *codec) { unsigned int present; @@ -7661,6 +7730,7 @@ static const char *alc883_models[ALC883_ [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M720R] = "clevo-m720r", + [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC883_AUTO] = "auto", }; @@ -7706,6 +7776,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), @@ -7950,6 +8021,18 @@ static struct alc_config_preset alc883_p .unsol_event = alc883_mitac_unsol_event, .init_hook = alc883_mitac_automute, }, + [ALC883_FUJITSU_PI2515] = { + .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, + .init_verbs = { alc883_init_verbs, alc883_2ch_fujitsu_pi2515_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_fujitsu_pi2515_capture_source, + .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, + .init_hook = alc883_2ch_fujitsu_pi2515_automute, + }, };

2 1

[alsa-devel] [PATCH 4/5] asihpi: replace volatile with barriers
by Eliot Blennerhassett 06 Mar '08

06 Mar '08

Replace volatile with barriers where required. Other minor cleanups. Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com --- diff -r 6def4892d3f5 pci/asihpi/hpi6205.c --- a/pci/asihpi/hpi6205.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpi6205.c Thu Mar 06 12:21:54 2008 +1300 @@ -27,6 +27,7 @@ (C) Copyright AudioScience Inc. 1998-2003 *******************************************************************************/ #define SOURCEFILE_NAME "hpi6205.c" + #include "hpi.h" #include "hpidebug.h" #include "hpi6205.h" @@ -35,42 +36,42 @@ /*****************************************************************************/ /* HPI6205 specific error codes */ -#define HPI6205_ERROR_BASE 1000 -#define HPI6205_ERROR_MEM_ALLOC 1001 -#define HPI6205_ERROR_6205_NO_IRQ 1002 -#define HPI6205_ERROR_6205_INIT_FAILED 1003 +#define HPI6205_ERROR_BASE 1000 +/*#define HPI6205_ERROR_MEM_ALLOC 1001 */ +#define HPI6205_ERROR_6205_NO_IRQ 1002 +#define HPI6205_ERROR_6205_INIT_FAILED 1003 /*#define HPI6205_ERROR_MISSING_DSPCODE 1004 */ -#define HPI6205_ERROR_UNKNOWN_PCI_DEVICE 1005 -#define HPI6205_ERROR_6205_REG 1006 -#define HPI6205_ERROR_6205_DSPPAGE 1007 -#define HPI6205_ERROR_BAD_DSPINDEX 1008 -#define HPI6205_ERROR_C6713_HPIC 1009 -#define HPI6205_ERROR_C6713_HPIA 1010 -#define HPI6205_ERROR_C6713_PLL 1011 -#define HPI6205_ERROR_DSP_INTMEM 1012 -#define HPI6205_ERROR_DSP_EXTMEM 1013 -#define HPI6205_ERROR_DSP_PLD 1014 -#define HPI6205_ERROR_MSG_RESP_IDLE_TIMEOUT 1015 -#define HPI6205_ERROR_MSG_RESP_TIMEOUT 1016 -#define HPI6205_ERROR_6205_EEPROM 1017 -#define HPI6205_ERROR_DSP_EMIF 1018 +#define HPI6205_ERROR_UNKNOWN_PCI_DEVICE 1005 +#define HPI6205_ERROR_6205_REG 1006 +#define HPI6205_ERROR_6205_DSPPAGE 1007 +#define HPI6205_ERROR_BAD_DSPINDEX 1008 +#define HPI6205_ERROR_C6713_HPIC 1009 +#define HPI6205_ERROR_C6713_HPIA 1010 +#define HPI6205_ERROR_C6713_PLL 1011 +#define HPI6205_ERROR_DSP_INTMEM 1012 +#define HPI6205_ERROR_DSP_EXTMEM 1013 +#define HPI6205_ERROR_DSP_PLD 1014 +#define HPI6205_ERROR_MSG_RESP_IDLE_TIMEOUT 1015 +#define HPI6205_ERROR_MSG_RESP_TIMEOUT 1016 +#define HPI6205_ERROR_6205_EEPROM 1017 +#define HPI6205_ERROR_DSP_EMIF 1018 #define Hpi6205_Error(nDspIndex, err) (err) /*****************************************************************************/ /* for C6205 PCI i/f */ /* Host Status Register (HSR) bitfields */ -#define C6205_HSR_INTSRC 0x01 -#define C6205_HSR_INTAVAL 0x02 -#define C6205_HSR_INTAM 0x04 -#define C6205_HSR_CFGERR 0x08 -#define C6205_HSR_EEREAD 0x10 +#define C6205_HSR_INTSRC 0x01 +#define C6205_HSR_INTAVAL 0x02 +#define C6205_HSR_INTAM 0x04 +#define C6205_HSR_CFGERR 0x08 +#define C6205_HSR_EEREAD 0x10 /* Host-to-DSP Control Register (HDCR) bitfields */ -#define C6205_HDCR_WARMRESET 0x01 -#define C6205_HDCR_DSPINT 0x02 -#define C6205_HDCR_PCIBOOT 0x04 +#define C6205_HDCR_WARMRESET 0x01 +#define C6205_HDCR_DSPINT 0x02 +#define C6205_HDCR_PCIBOOT 0x04 /* DSP Page Register (DSPP) bitfields, */ /* defines 4 Mbyte page that BAR0 points to */ -#define C6205_DSPP_MAP1 0x400 +#define C6205_DSPP_MAP1 0x400 /* BAR0 maps to prefetchable 4 Mbyte memory block set by DSPP. * BAR1 maps to non-prefetchable 8 Mbyte memory block @@ -79,7 +80,7 @@ * needs to be added to the BAR1 base address set in the PCI config reg */ #define C6205_BAR1_PCI_IO_OFFSET (0x027FFF0L) -#define C6205_BAR1_HSR (C6205_BAR1_PCI_IO_OFFSET) +#define C6205_BAR1_HSR (C6205_BAR1_PCI_IO_OFFSET) #define C6205_BAR1_HDCR (C6205_BAR1_PCI_IO_OFFSET+4) #define C6205_BAR1_DSPP (C6205_BAR1_PCI_IO_OFFSET+8) @@ -87,23 +88,23 @@ #define C6205_BAR0_TIMER1_CTL (0x01980000L) /* For first 6713 in CE1 space, using DA17,16,2 */ -#define HPICL_ADDR 0x01400000L -#define HPICH_ADDR 0x01400004L -#define HPIAL_ADDR 0x01410000L -#define HPIAH_ADDR 0x01410004L -#define HPIDIL_ADDR 0x01420000L -#define HPIDIH_ADDR 0x01420004L -#define HPIDL_ADDR 0x01430000L -#define HPIDH_ADDR 0x01430004L - -#define C6713_EMIF_GCTL 0x01800000 -#define C6713_EMIF_CE1 0x01800004 -#define C6713_EMIF_CE0 0x01800008 -#define C6713_EMIF_CE2 0x01800010 -#define C6713_EMIF_CE3 0x01800014 -#define C6713_EMIF_SDRAMCTL 0x01800018 -#define C6713_EMIF_SDRAMTIMING 0x0180001C -#define C6713_EMIF_SDRAMEXT 0x01800020 +#define HPICL_ADDR 0x01400000L +#define HPICH_ADDR 0x01400004L +#define HPIAL_ADDR 0x01410000L +#define HPIAH_ADDR 0x01410004L +#define HPIDIL_ADDR 0x01420000L +#define HPIDIH_ADDR 0x01420004L +#define HPIDL_ADDR 0x01430000L +#define HPIDH_ADDR 0x01430004L + +#define C6713_EMIF_GCTL 0x01800000 +#define C6713_EMIF_CE1 0x01800004 +#define C6713_EMIF_CE0 0x01800008 +#define C6713_EMIF_CE2 0x01800010 +#define C6713_EMIF_CE3 0x01800014 +#define C6713_EMIF_SDRAMCTL 0x01800018 +#define C6713_EMIF_SDRAMTIMING 0x0180001C +#define C6713_EMIF_SDRAMEXT 0x01800020 struct hpi_hw_obj { /* PCI registers */ @@ -114,7 +115,7 @@ struct hpi_hw_obj { u32 dwDspPage; struct consistent_dma_area hLockedMem; - volatile struct bus_master_interface *pInterfaceBuffer; + struct bus_master_interface *pInterfaceBuffer; u16 flagOStreamJustReset[HPI_MAX_STREAMS]; /* a non-NULL handle means there is an HPI allocated buffer */ @@ -134,6 +135,17 @@ struct hpi_hw_obj { /* local prototypes */ #define CheckBeforeBBMCopy(status, pBBMData, lFirstWrite, lSecondWrite) + +static int WaitDspAck( + struct hpi_hw_obj *pHw6205, + int state, + int timeout_us +); + +static void SendDspCommand( + struct hpi_hw_obj *pHw6205, + int cmd +); static u16 Hpi6205_AdapterBootLoadDsp( struct hpi_adapter_obj *pao, @@ -312,11 +324,13 @@ static void ControlMessage( switch (phm->wFunction) { case HPI_CONTROL_GET_STATE: - if (pao->wHasControlCache) + if (pao->wHasControlCache) { + rmb(); /* make sure we see updates DMAed from DSP */ if (HpiCheckControlCache (&pHw6205->pControlCache[phm->u.c. wControlIndex], phm, phr)) break; + } HW_Message(pao, phm, phr); break; case HPI_CONTROL_GET_INFO: @@ -597,7 +611,7 @@ static u16 CreateAdapterObj( ) { struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface; + struct bus_master_interface *interface; u32 dwPhysAddr; u32 dwTimeOut = HPI6205_TIMEOUT; u32 dwTemp1; @@ -631,6 +645,7 @@ static u16 CreateAdapterObj( HPI_DEBUG_LOG(DEBUG, "Interface buffer address %p\n", pHw6205->pInterfaceBuffer); + if (pHw6205->pInterfaceBuffer) { memset((void *)pHw6205->pInterfaceBuffer, 0, sizeof(struct bus_master_interface)); @@ -647,7 +662,7 @@ static u16 CreateAdapterObj( /* allow boot load even if mem alloc wont work */ if (!pHw6205->pInterfaceBuffer) - return (Hpi6205_Error(0, HPI6205_ERROR_MEM_ALLOC)); + return (Hpi6205_Error(0, HPI_ERROR_MEMORY_ALLOC)); interface = pHw6205->pInterfaceBuffer; @@ -671,14 +686,8 @@ static u16 CreateAdapterObj( iowrite32(C6205_HSR_INTSRC, pHw6205->prHSR); /* make sure the DSP has started ok */ - dwTimeOut = 100; - while ((interface->dwDspAck != H620_HIF_RESET) && dwTimeOut) { - dwTimeOut--; - HpiOs_DelayMicroSeconds(10000); - } - - if (dwTimeOut == 0) { - HPI_DEBUG_LOG(ERROR, "Timed out waiting ack \n"); + if (!WaitDspAck(pHw6205, H620_HIF_RESET, HPI6205_TIMEOUT)) { + HPI_DEBUG_LOG(ERROR, "Timed out waiting reset state \n"); return (Hpi6205_Error(0, HPI6205_ERROR_6205_INIT_FAILED)); } /* Note that *pao, *pHw6205 are zeroed after allocation, @@ -746,14 +755,7 @@ static u16 CreateAdapterObj( } } } - /* set interface to idle */ - interface->dwHostCmd = H620_HIF_IDLE; - /* interrupt the DSP again */ - dwTemp1 = ioread32(pHw6205->prHDCR); - dwTemp1 |= (u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); - dwTemp1 &= ~(u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); + SendDspCommand(pHw6205, H620_HIF_IDLE); { struct hpi_message hM; @@ -789,7 +791,7 @@ static u16 CreateAdapterObj( } HPI_DEBUG_LOG(VERBOSE, "Get adapter info OK\n"); - pao->wOpen = 0; /* upon creation the adapter is closed */ + pao->wOpen = 0; /* upon creation the adapter is closed */ HPI_DEBUG_LOG(INFO, "Bootload DSP OK\n"); return (0); @@ -872,8 +874,7 @@ static void OutStreamHostBufferAllocate( u32 dwCommand = phm->u.d.u.Buffer.dwCommand; /*u16 wStreamIndex = phm->u.d.wStreamIndex; */ struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; HPI_InitResponse(phr, phm->wObject, phm->wFunction, 0); @@ -941,7 +942,7 @@ static void OutStreamHostBufferAllocate( /* GRANT phase. Set up the BBM status, tell the DSP about the buffer so it can start using BBM. */ - volatile struct hostbuffer_status_6205 *status; + struct hostbuffer_status_6205 *status; if (phm->u.d.u.Buffer. dwBufferSize & (phm->u.d.u.Buffer.dwBufferSize - 1)) { @@ -1005,7 +1006,7 @@ static void OutStreamHostBufferFree( } static long OutStreamGetSpaceAvailable( - volatile struct hostbuffer_status_6205 *status + struct hostbuffer_status_6205 *status ) { return status->dwSizeInBytes - ((long)(status->dwHostIndex) - @@ -1019,9 +1020,8 @@ static void OutStreamWrite( ) { struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; - volatile struct hostbuffer_status_6205 *status; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; + struct hostbuffer_status_6205 *status; long dwSpaceAvailable; if (!pHw6205->OutStreamHostBufferSize[phm->u.d.wStreamIndex]) { /* there is no BBM buffer */ @@ -1126,9 +1126,8 @@ static void OutStreamGetInfo( ) { struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; - volatile struct hostbuffer_status_6205 *status; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; + struct hostbuffer_status_6205 *status; if (!pHw6205->OutStreamHostBufferSize[phm->u.d.wStreamIndex]) { HW_Message(pao, phm, phr); @@ -1188,13 +1187,13 @@ static void InStreamHostBufferAllocate( u16 err = 0; u32 dwCommand = phm->u.d.u.Buffer.dwCommand; struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; HPI_InitResponse(phr, phm->wObject, phm->wFunction, 0); if (dwCommand == HPI_BUFFER_CMD_EXTERNAL || dwCommand == HPI_BUFFER_CMD_INTERNAL_ALLOC) { + phm->u.d.u.Buffer.dwBufferSize = roundup_pow_of_two(phm->u.d.u.Buffer.dwBufferSize); phr->u.d.u.stream_info.dwDataAvailable = @@ -1247,7 +1246,7 @@ static void InStreamHostBufferAllocate( if (dwCommand == HPI_BUFFER_CMD_EXTERNAL || dwCommand == HPI_BUFFER_CMD_INTERNAL_GRANTADAPTER) { - volatile struct hostbuffer_status_6205 *status; + struct hostbuffer_status_6205 *status; if (phm->u.d.u.Buffer. dwBufferSize & (phm->u.d.u.Buffer.dwBufferSize - 1)) { @@ -1267,6 +1266,7 @@ static void InStreamHostBufferAllocate( status->dwDSPIndex = 0; status->dwHostIndex = status->dwDSPIndex; status->dwSizeInBytes = phm->u.d.u.Buffer.dwBufferSize; + HW_Message(pao, phm, phr); if (phr->wError && HpiOs_LockedMem_Valid(&pHw6205-> @@ -1297,10 +1297,10 @@ static void InStreamHostBufferFree( } if (dwCommand == HPI_BUFFER_CMD_EXTERNAL || - dwCommand == HPI_BUFFER_CMD_INTERNAL_FREE) { + dwCommand == HPI_BUFFER_CMD_INTERNAL_FREE) HpiOs_LockedMem_Free(&pHw6205-> InStreamHostBuffers[phm->u.d.wStreamIndex]); - } + } else { /* Should HPI_ERROR_INVALID_OPERATION be returned if no host buffer is allocated? */ @@ -1321,7 +1321,7 @@ static void InStreamStart( } static long InStreamGetBytesAvailable( - volatile struct hostbuffer_status_6205 *status + struct hostbuffer_status_6205 *status ) { return (long)(status->dwDSPIndex) - (long)(status->dwHostIndex); @@ -1334,9 +1334,8 @@ static void InStreamRead( ) { struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; - volatile struct hostbuffer_status_6205 *status; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; + struct hostbuffer_status_6205 *status; long dwDataAvailable; u8 *pBBMData; long lFirstRead; @@ -1387,9 +1386,8 @@ static void InStreamGetInfo( ) { struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; - volatile struct hostbuffer_status_6205 *status; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; + struct hostbuffer_status_6205 *status; if (!pHw6205->InStreamHostBufferSize[phm->u.d.wStreamIndex]) { HW_Message(pao, phm, phr); return; @@ -1451,7 +1449,7 @@ static u16 Hpi6205_AdapterBootLoadDsp( dwTemp = C6205_HDCR_WARMRESET; iowrite32(dwTemp, pHw6205->prHDCR); HpiOs_DelayMicroSeconds(1000); -/* for(i=0;i<1000; i++) dwTemp = ioread32(pao,pHw6205->prHDCR); */ +/* for(i=0;i<1000; i++) dwTemp = ioread32(pao,pHw6205->prHDCR); */ /* check that PCI i/f was configured by EEPROM */ dwTemp = ioread32(pHw6205->prHSR); @@ -1620,7 +1618,7 @@ static u16 Hpi6205_AdapterBootLoadDsp( if (pHw6205->pInterfaceBuffer) { /* we need to tell the card the physical PCI address */ u32 dwPhysicalPCIaddress; - volatile struct bus_master_interface *interface = + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; u32 dwHostMailboxAddressOnDsp; u32 dwPhysicalPCIaddressVerify = 0; @@ -1628,6 +1626,7 @@ static u16 Hpi6205_AdapterBootLoadDsp( /* set ack so we know when DSP is ready to go */ /* (dwDspAck will be changed to HIF_RESET) */ interface->dwDspAck = H620_HIF_UNKNOWN; + wmb(); /* ensure ack is written before dsp writes back */ err = HpiOs_LockedMem_GetPhysAddr(&pHw6205->hLockedMem, &dwPhysicalPCIaddress); @@ -1715,7 +1714,7 @@ static u16 BootLoader_WriteMem32( struct hpi_hw_obj *pHw6205 = pao->priv; u16 err = 0; __iomem u32 *pData; - /* u32 dwVerifyData=0; */ + /* u32 dwVerifyData=0; */ if (nDSPIndex == 0) { /* DSP 0 is always C6205 */ @@ -1771,8 +1770,8 @@ static u16 BootLoader_ConfigEMIF( /* Set the EMIF */ /* memory map of C6205 */ - /* 00000000-0000FFFF 16Kx32 internal program */ - /* 00400000-00BFFFFF CE0 2Mx32 SDRAM running @ 100MHz */ + /* 00000000-0000FFFF 16Kx32 internal program */ + /* 00400000-00BFFFFF CE0 2Mx32 SDRAM running @ 100MHz */ /* EMIF config */ /*------------ */ @@ -1899,7 +1898,7 @@ static u16 BootLoader_ConfigEMIF( BootLoader_WriteMem32(pao, nDSPIndex, 0x01B7C120, 0x8002); /* peri = 189/2 */ BootLoader_WriteMem32(pao, nDSPIndex, 0x01B7C11C, 0x8001); - /* cpu = 189/1 */ + /* cpu = 189/1 */ BootLoader_WriteMem32(pao, nDSPIndex, 0x01B7C118, 0x8000); HpiOs_DelayMicroSeconds(1000); /* ** SGT test to take GPO3 high when we start the PLL */ @@ -1980,15 +1979,15 @@ static u16 BootLoader_TestMemory( dwTestAddr); if (dwData != dwTestData) { HPI_DEBUG_LOG(VERBOSE, - "Memtest error details " + "Memtest error details " "%08x %08x %08x %i\n", dwTestAddr, dwTestData, dwData, nDSPIndex); return (1); /* error */ } dwTestData = dwTestData << 1; - } /* for(j) */ - } /* for(i) */ + } /* for(j) */ + } /* for(i) */ /* for the next 100 locations test each location, leaving it as zero */ /* write a zero to the next word in memory before we read */ @@ -2001,7 +2000,7 @@ static u16 BootLoader_TestMemory( dwData = BootLoader_ReadMem32(pao, nDSPIndex, dwTestAddr); if (dwData != dwTestData) { HPI_DEBUG_LOG(VERBOSE, - "Memtest error details " + "Memtest error details " "%08x %08x %08x %i\n", dwTestAddr, dwTestData, dwData, nDSPIndex); return (1); /* error */ @@ -2015,7 +2014,7 @@ static u16 BootLoader_TestMemory( dwTestAddr = dwStartAddress + (u32)i *4; BootLoader_WriteMem32(pao, nDSPIndex, dwTestAddr, 0x0); } - return (0); /*success! */ + return (0); /*success! */ } static u16 BootLoader_TestInternalMemory( @@ -2134,22 +2133,14 @@ static short Hpi6205_TransferData( /*u8 *pData =(u8 *)phm->u.d.u.Data.dwpbData; */ /*u16 wTimeOut=8; */ u16 err = 0; - u32 dwTimeOut, dwTemp1, dwTemp2; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; + u32 dwTimeOut, dwTemp2; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; dwDataSize &= ~3L; /* round dwDataSize down to nearest 4 bytes */ /* make sure state is IDLE */ - dwTimeOut = HPI6205_TIMEOUT; - dwTemp2 = 0; - while ((interface->dwDspAck != H620_HIF_IDLE) && dwTimeOut--) - HpiOs_DelayMicroSeconds(1); - - if (interface->dwDspAck != H620_HIF_IDLE) + if (!WaitDspAck(pHw6205, H620_HIF_IDLE, HPI6205_TIMEOUT)) return HPI_ERROR_DSP_HARDWARE; - - interface->dwHostCmd = nOperation; while (dwDataTransferred < dwDataSize) { u32 nThisCopy = dwDataSize - dwDataTransferred; @@ -2162,13 +2153,7 @@ static short Hpi6205_TransferData( &pData[dwDataTransferred], nThisCopy); interface->dwTransferSizeInBytes = nThisCopy; - - /* interrupt the DSP */ - dwTemp1 = ioread32(pHw6205->prHDCR); - dwTemp1 |= (u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); - dwTemp1 &= ~(u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); + SendDspCommand(pHw6205, nOperation); /* spin waiting on the result */ dwTimeOut = HPI6205_TIMEOUT; @@ -2212,20 +2197,55 @@ static short Hpi6205_TransferData( if (interface->dwDspAck != nOperation) HPI_DEBUG_LOG(DEBUG, "interface->dwDspAck=%d, expected %d\n", interface->dwDspAck, nOperation); - /* err=HPI_ERROR_DSP_HARDWARE; */ - - /* set interface back to idle */ - interface->dwHostCmd = H620_HIF_IDLE; - /* interrupt the DSP again */ - dwTemp1 = ioread32(pHw6205->prHDCR); - dwTemp1 |= (u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); - dwTemp1 &= ~(u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); + /* err=HPI_ERROR_DSP_HARDWARE; */ + + SendDspCommand(pHw6205, H620_HIF_IDLE); return err; } +/* wait for up to timeout_us microseconds for the DSP + to signal state by DMA into dwDspAck +*/ +static int WaitDspAck( + struct hpi_hw_obj *pHw6205, + int state, + int timeout_us +) +{ + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; + int t = timeout_us / 4; + + rmb(); /* ensure interface->dwDspAck is up to date */ + while ((interface->dwDspAck != state) && t--) { + HpiOs_DelayMicroSeconds(4); + rmb(); /* DSP changes dwDspAck by DMA */ + } + + /*HPI_DEBUG_LOG(VERBOSE, "Spun %d for %d\n", timeout_us/4-t, state); */ + return t; +} + +/* set the busmaster interface to cmd, then interrupt the DSP */ +static void SendDspCommand( + struct hpi_hw_obj *pHw6205, + int cmd +) +{ + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; + + u32 r; + + interface->dwHostCmd = cmd; + wmb(); /* DSP gets state by DMA, make sure it is written to memory */ + /* before we interrupt the DSP */ + r = ioread32(pHw6205->prHDCR); + r |= (u32)C6205_HDCR_DSPINT; + iowrite32(r, pHw6205->prHDCR); + r &= ~(u32)C6205_HDCR_DSPINT; + iowrite32(r, pHw6205->prHDCR); +} + static unsigned int messageCount; static u16 Hpi6205_MessageResponseSequence( @@ -2234,46 +2254,26 @@ static u16 Hpi6205_MessageResponseSequen struct hpi_response *phr ) { - u32 dwTemp1, dwTemp2, dwTimeOut, dwTimeOut2; + u32 dwTemp2, dwTimeOut, dwTimeOut2; struct hpi_hw_obj *pHw6205 = pao->priv; - volatile struct bus_master_interface *interface = - pHw6205->pInterfaceBuffer; + struct bus_master_interface *interface = pHw6205->pInterfaceBuffer; u16 err = 0; messageCount++; /* Assume buffer of type struct bus_master_interface is allocated "noncacheable" */ - /* make sure state is IDLE */ - dwTimeOut = HPI6205_TIMEOUT; - dwTemp2 = 0; - while ((interface->dwDspAck != H620_HIF_IDLE) && --dwTimeOut) - HpiOs_DelayMicroSeconds(1); - - if (dwTimeOut == 0) { + if (!WaitDspAck(pHw6205, H620_HIF_IDLE, HPI6205_TIMEOUT)) { HPI_DEBUG_LOG(DEBUG, "Timeout waiting for idle\n"); return (Hpi6205_Error (0, HPI6205_ERROR_MSG_RESP_IDLE_TIMEOUT)); } interface->u.MessageBuffer = *phm; /* signal we want a response */ - interface->dwHostCmd = H620_HIF_GET_RESP; - - /* interrupt the DSP */ - dwTemp1 = ioread32(pHw6205->prHDCR); - dwTemp1 |= (u32)C6205_HDCR_DSPINT; - HpiOs_DelayMicroSeconds(1); - iowrite32(dwTemp1, pHw6205->prHDCR); - dwTemp1 &= ~(u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); - - /* spin waiting on state change (start of msg process) */ - dwTimeOut2 = HPI6205_TIMEOUT; - dwTemp2 = 0; - while ((interface->dwDspAck != H620_HIF_GET_RESP) && --dwTimeOut2) { - dwTemp2 = ioread32(pHw6205->prHSR); - dwTemp2 &= C6205_HSR_INTSRC; - } + SendDspCommand(pHw6205, H620_HIF_GET_RESP); + + dwTimeOut2 = WaitDspAck(pHw6205, H620_HIF_GET_RESP, HPI6205_TIMEOUT); + if (dwTimeOut2 == 0) { HPI_DEBUG_LOG(DEBUG, "(%u)Timed out waiting for " @@ -2284,16 +2284,16 @@ static u16 Hpi6205_MessageResponseSequen "(%u)Transition to GET_RESP after %u\n", messageCount, HPI6205_TIMEOUT - dwTimeOut2); } - /* spin waiting on HIF interrupt flag (end of msg process) */ dwTimeOut = HPI6205_TIMEOUT; dwTemp2 = 0; while ((dwTemp2 == 0) && --dwTimeOut) { dwTemp2 = ioread32(pHw6205->prHSR); dwTemp2 &= C6205_HSR_INTSRC; - /* HpiOs_DelayMicroSeconds(5); */ + HpiOs_DelayMicroSeconds(1); } if (dwTemp2 == C6205_HSR_INTSRC) { + rmb(); /* ensure we see latest value for dwDspAck */ if ((interface->dwDspAck != H620_HIF_GET_RESP)) { HPI_DEBUG_LOG(DEBUG, "(%u)interface->dwDspAck(0x%x) != " @@ -2306,12 +2306,12 @@ static u16 Hpi6205_MessageResponseSequen messageCount, HPI6205_TIMEOUT - dwTimeOut); } - } -/* need to handle this differently... */ - else + } else { + /* can we do anything else in response to the error ? */ HPI_DEBUG_LOG(ERROR, "Interrupt from HIF module BAD (wFunction %x)\n", phm->wFunction); + } /* reset the interrupt from the DSP */ iowrite32(C6205_HSR_INTSRC, pHw6205->prHSR); @@ -2321,15 +2321,8 @@ static u16 Hpi6205_MessageResponseSequen *phr = interface->u.ResponseBuffer; /* set interface back to idle */ - interface->dwHostCmd = H620_HIF_IDLE; - /* interrupt the DSP again */ - dwTemp1 = ioread32(pHw6205->prHDCR); - dwTemp1 |= (u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); - dwTemp1 &= ~(u32)C6205_HDCR_DSPINT; - iowrite32(dwTemp1, pHw6205->prHDCR); - -/* EWB move timeoutcheck to after IDLE command, maybe recover? */ + SendDspCommand(pHw6205, H620_HIF_IDLE); + if ((dwTimeOut == 0) || (dwTimeOut2 == 0)) { HPI_DEBUG_LOG(DEBUG, "Something timed out!\n"); return Hpi6205_Error(0, HPI6205_ERROR_MSG_RESP_TIMEOUT); @@ -2337,13 +2330,7 @@ static u16 Hpi6205_MessageResponseSequen /* special case for adapter close - */ /* wait for the DSP to indicate it is idle */ if (phm->wFunction == HPI_ADAPTER_CLOSE) { - /* make sure state is IDLE */ - dwTimeOut = HPI6205_TIMEOUT; - dwTemp2 = 0; - while ((interface->dwDspAck != H620_HIF_IDLE) && --dwTimeOut) - HpiOs_DelayMicroSeconds(1); - - if (dwTimeOut == 0) { + if (!WaitDspAck(pHw6205, H620_HIF_IDLE, HPI6205_TIMEOUT)) { HPI_DEBUG_LOG(DEBUG, "Timeout waiting for idle " "(on AdapterClose)\n"); diff -r 6def4892d3f5 pci/asihpi/hpicmn.c --- a/pci/asihpi/hpicmn.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpicmn.c Thu Mar 06 12:21:31 2008 +1300 @@ -148,7 +148,7 @@ static void SubSysGetAdapters( /* input: NONE */ /* output: wNumAdapters */ - /* awAdapter[] */ + /* awAdapter[] */ /* */ short i; @@ -178,7 +178,7 @@ static void SubSysGetAdapters( * It returns nonzero if a cache hit occurred, zero otherwise. */ short HpiCheckControlCache( - volatile struct hpi_control_cache_single *pC, + struct hpi_control_cache_single *pC, struct hpi_message *phm, struct hpi_response *phr ) @@ -301,7 +301,7 @@ Multiplexer does so use sent values Multiplexer does so use sent values */ void HpiSyncControlCache( - volatile struct hpi_control_cache_single *pC, + struct hpi_control_cache_single *pC, struct hpi_message *phm, struct hpi_response *phr ) diff -r 6def4892d3f5 pci/asihpi/hpicmn.h --- a/pci/asihpi/hpicmn.h Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpicmn.h Thu Mar 06 12:21:30 2008 +1300 @@ -44,12 +44,12 @@ void HpiDeleteAdapter( ); short HpiCheckControlCache( - volatile struct hpi_control_cache_single *pC, + struct hpi_control_cache_single *pC, struct hpi_message *phm, struct hpi_response *phr ); void HpiSyncControlCache( - volatile struct hpi_control_cache_single *pC, + struct hpi_control_cache_single *pC, struct hpi_message *phm, struct hpi_response *phr );

1 0

[alsa-devel] [PATCH 1/5] asihpi: minor checkpatch cleanups
by Eliot Blennerhassett 06 Mar '08

06 Mar '08

A few minor checkpatch cleanups. Make hpi ioctl cmd unique. Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> --- diff -r 6def4892d3f5 pci/asihpi/hpi6000.c --- a/pci/asihpi/hpi6000.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpi6000.c Wed Mar 05 21:59:25 2008 +1300 @@ -923,15 +923,15 @@ static short Hpi6000_AdapterBootLoadDsp( 0 - 1 - 2 - - 3 CLK2EN=1 CLKOUT2 enabled - 4 CLK1EN=0 CLKOUT1 disabled - 5 EKEN=1 <----------------!! C6713 specific, enables ECLKOUT + 3 CLK2EN = 1 CLKOUT2 enabled + 4 CLK1EN = 0 CLKOUT1 disabled + 5 EKEN = 1 <--!! C6713 specific, enables ECLKOUT 6 - - 7 NOHOLD=1 external HOLD disabled - 8 HOLDA=0 HOLDA output is low - 9 HOLD=0 HOLD input is low - 10 ARDY=1 ARDY input is high - 11 BUSREQ=0 BUSREQ output is low + 7 NOHOLD = 1 external HOLD disabled + 8 HOLDA = 0 HOLDA output is low + 9 HOLD = 0 HOLD input is low + 10 ARDY = 1 ARDY input is high + 11 BUSREQ = 0 BUSREQ output is low 12,13 Reserved = 1 */ HpiWriteWord(pdo, 0x01800000, 0x34A8); @@ -952,12 +952,12 @@ static short Hpi6000_AdapterBootLoadDsp( /* EMIF SDRAM Extension 31-21 0 20 WR2RD = 0 - 19-18 WR2DEAC=1 - 17 WR2WR=0 - 16-15 R2WDQM=2 - 14-12 RD2WR=4 - 11-10 RD2DEAC=1 - 9 RD2RD= 1 + 19-18 WR2DEAC = 1 + 17 WR2WR = 0 + 16-15 R2WDQM = 2 + 14-12 RD2WR = 4 + 11-10 RD2DEAC = 1 + 9 RD2RD = 1 8-7 THZP = 10b 6-5 TWR = 2-1 = 01b (tWR = 10ns) 4 TRRD = 0b = 2 ECLK (tRRD = 14ns) diff -r 6def4892d3f5 pci/asihpi/hpi6205.h --- a/pci/asihpi/hpi6205.h Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpi6205.h Wed Mar 05 22:06:08 2008 +1300 @@ -30,14 +30,14 @@ Copyright AudioScience, Inc., 2003 /*********************************************************** Defines used for basic messaging ************************************************************/ -#define H620_HIF_RESET 0 -#define H620_HIF_IDLE 1 -#define H620_HIF_GET_RESP 2 -#define H620_HIF_DATA_DONE 3 -#define H620_HIF_DATA_MASK 0x10 -#define H620_HIF_SEND_DATA 0x14 -#define H620_HIF_GET_DATA 0x15 -#define H620_HIF_UNKNOWN 0xffff +#define H620_HIF_RESET 0 +#define H620_HIF_IDLE 1 +#define H620_HIF_GET_RESP 2 +#define H620_HIF_DATA_DONE 3 +#define H620_HIF_DATA_MASK 0x10 +#define H620_HIF_SEND_DATA 0x14 +#define H620_HIF_GET_DATA 0x15 +#define H620_HIF_UNKNOWN 0xffff /*********************************************************** Types used for mixer control caching @@ -48,7 +48,7 @@ Copyright AudioScience, Inc., 2003 #define HPI_NMIXER_CONTROLS 2048 /********************************************************************* - This is used for background buffer bus mastering stream buffers. +This is used for background buffer bus mastering stream buffers. **********************************************************************/ struct hostbuffer_status_6205 { u32 dwSamplesProcessed; @@ -62,7 +62,7 @@ struct hostbuffer_status_6205 { }; /********************************************************************* - This is used for dynamic control cache allocation +This is used for dynamic control cache allocation **********************************************************************/ struct controlcache_6205 { u32 dwNumberOfControls; @@ -71,7 +71,7 @@ struct controlcache_6205 { }; /********************************************************************* - This is used for dynamic allocation of async event array +This is used for dynamic allocation of async event array **********************************************************************/ struct async_event_buffer_6205 { u32 dwPhysicalPCI32address; @@ -80,8 +80,8 @@ struct async_event_buffer_6205 { }; /*********************************************************** - The Host located memory buffer that the 6205 will bus master - in and out of. +The Host located memory buffer that the 6205 will bus master +in and out of. ************************************************************/ #define HPI6205_SIZEOF_DATA (16*1024) struct bus_master_interface { diff -r 6def4892d3f5 pci/asihpi/hpimsginit.c --- a/pci/asihpi/hpimsginit.c Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpimsginit.c Thu Mar 06 12:21:47 2008 +1300 @@ -29,7 +29,7 @@ static u16 aMsgSize[HPI_OBJ_MAXINDEX + 1 /* The actual response size for each object type */ static u16 aResSize[HPI_OBJ_MAXINDEX + 1] = HPI_RESPONSE_SIZE_BY_OBJECT; /* Flag to enable alternate message type for SSX2 bypass. */ -static u16 gwSSX2Bypass = 0; +static u16 gwSSX2Bypass; /** \internal * Used by ASIO driver to disable SSX2 for a single process diff -r 6def4892d3f5 pci/asihpi/hpios.h --- a/pci/asihpi/hpios.h Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpios.h Wed Mar 05 21:58:51 2008 +1300 @@ -27,13 +27,6 @@ HPI Operating System function declaratio #ifndef STR_SIZE #define STR_SIZE(a) (a) -#endif - -#ifndef __user -#define __user -#endif -#ifndef __iomem -#define __iomem #endif /* physical memory allocation */ diff -r 6def4892d3f5 pci/asihpi/hpios_linux_kernel.h --- a/pci/asihpi/hpios_linux_kernel.h Mon Mar 03 11:05:48 2008 +0100 +++ b/pci/asihpi/hpios_linux_kernel.h Wed Mar 05 21:59:18 2008 +1300 @@ -28,6 +28,7 @@ HPI Operating System Specific macros for #endif #include <asm/io.h> +#include <asm/system.h> #include <linux/ioctl.h> #include <linux/kernel.h> #include <linux/string.h> @@ -35,10 +36,7 @@ HPI Operating System Specific macros for #include <linux/firmware.h> #include <linux/interrupt.h> #include <linux/pci.h> - -#if (LINUX_VERSION_CODE >= KERNEL_VERSION(2 , 5 , 0)) -# include <linux/device.h> -#endif +#include <linux/device.h> #define INLINE inline @@ -104,9 +102,8 @@ struct hpi_ioctl_linux { /* Conflict?: H is already used by a number of drivers hid, bluetooth hci, and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is ununsed command +*/ #define HPI_IOCTL_LINUX _IOWR('H', 0xFC, struct hpi_ioctl_linux) -*/ -#define HPI_IOCTL_LINUX _IOWR('H', 1, struct hpi_ioctl_linux) #define HPI_DEBUG_FLAG_ERROR KERN_ERR #define HPI_DEBUG_FLAG_WARNING KERN_WARNING

1 0

[alsa-devel] sound/pci/au88x0/au88x0_core.c: incorrect for loop in vortex_mixer_init()
by Roel Kluin 06 Mar '08

06 Mar '08

in sound/pci/au88x0/au88x0_core.c:430 (linus' tree) it reads in commented code: for (x = 0; x > 0x80; x++) { Not sure whether this is important but it appears to be incorrect. should it be 'x < 0x80' ?

1 0

[alsa-devel] volume control
by Reiner 05 Mar '08

05 Mar '08

Hello, is there a function available to control the volume of playback? Thanks Reiner

1 0

[alsa-devel] Alsa MMAP emulation
by Mathieu Monney 05 Mar '08

05 Mar '08

Hi, I'm trying to get working Alsa with the PortAudio library on an embedded device (Nokia N810). The problem is that the sound driver is not an hardware driver for Alsa but a plugin which deals with the internal DSP of the device. This plugin doesn't support mmap'ing at all. So I was wondering if it is possible to configure Alsa a way such PortAudio can use mmap'ing on my DSP plugin ? I saw there is a mmap_emulation flag but the problem is that it seems to be only available on HW driver and not on alsa plugins. Any clues ? Thank you in advance Mathieu

3 3

[alsa-devel] HDMI v. ICH8 on Dell XPS M1330
by Armando Di Cianno 05 Mar '08

05 Mar '08

I figured this was more appropriate for devel, and not user. Sorry, if that's wrong. First, I'm using the 1.0.16 release tarballs. They work better than the in-kernel drivers, as they survive both suspend and hibernate, whereas the kernel drivers do not. I'm eager to see audio over HDMI working on my Dell XPS M1330. I'm pretty sure that this is /not supported/ at the moment -- if this isn't the case, please stop reading, and let me know the particulars. HDMI seems to be supported noted, but mainly for the ATI chips. I'm pretty sure the m1330 has a STAC9228 setup. None of the suggested model= module parameters succeeded in getting audio over HDMI, beyond what is autodetected without any parameter specified. model=5stack seems to show a digital SPDIF device added. If there is anyway I can help the core developers get HDMI audio working, I would be glad to help. I will post all the relevant information I can think of following; please let me know if other information would be helpful. I will gladly turn on any excessive debugging options needed. I am an ALSA n00b, but I will happily do any required work that I can do from this end. Thanks! -- Armando Di Cianno http://dicianno.org/blog armando(a)dicianno.org armando(a)goodship.net w/o model= parameter # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 w/ model=5stack # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 # lspci -nvvvv -d 8086:284b 00:1b.0 0403: 8086:284b (rev 02) Subsystem: 1028:0209 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 21 Region 0: Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot +,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 unlimited ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

6 20

[alsa-devel] [PATCH] model for alc883 to support M720R
by Jiang zhe 04 Mar '08

04 Mar '08

Hi, Please refer to [0003781] on the alsa mantis. There is no suitable model for M720R. This patch is to support HP jack-sensing and mixer. The reporter complained that there was a constant noise when he recorded from Mic, It existed even after he muted the input. I checked the codec info carefully, the input widget was muted indeed. It's strange, Is there anything special for this codec? Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20080304/alsa-kernel/pci/hda/patch_realtek.c 2008-03-01 09:00:07.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-03-04 16:49:09.000000000 +0800 @@ -201,6 +201,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, + ALC883_CLEVO_M720R, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6645,6 +6646,33 @@ static struct snd_kcontrol_new alc883_mi { } /* end */ }; +static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7178,6 +7206,20 @@ static struct hda_verb alc883_mitac_verb { } /* end */ }; +static struct hda_verb alc883_clevo_m720r_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Int speaker */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7361,6 +7403,25 @@ static void alc883_tagra_unsol_event(str alc883_tagra_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_clevo_m720r_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_clevo_m720r_automute(codec); +} + static void alc883_haier_w66_automute(struct hda_codec *codec) { unsigned int present; @@ -7598,6 +7659,7 @@ static const char *alc883_models[ALC883_ [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", + [ALC883_CLEVO_M720R] = "clevo-m720r", [ALC883_AUTO] = "auto", }; @@ -7638,6 +7700,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), @@ -7780,6 +7843,18 @@ static struct alc_config_preset alc883_p .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_CLEVO_M720R] = { + .mixers = { alc883_clevo_m720r_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720r_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_clevo_m720r_unsol_event, + .init_hook = alc883_clevo_m720r_automute, + }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},

3 4

[alsa-devel] Driver recovers from underrun in plain ALSA mode, but not in OSS emulation
by Timur Tabi 04 Mar '08

04 Mar '08

I'm stress-testing my sound driver using mplayer to see how it handles underrun during jumps in CPU loads. That is, I start playing a movie, and then I cause the CPU to jump to 100%, and then I drop the CPU back to normal. During the times when the CPU is at 100%, I get underruns and other problems. That's normal. When I tell mplayer to use the ALSA API, my driver recovers nicely from an underrun situation when the CPU load drops back to normal. However, if I tell mplayer to use the OSS API (i.e. ALSA's OSS emulation), it does not recover from an underrun situation. The audio and video playback remains distorted. I need to tell mplayer to pause and resume to get it to recover. Is this normal for ALSA drivers? Before I try to debug this problem the hard way, I want to know if this could be the result of a common programming mistake in my driver or something like similar. (FYI, the source code for my driver is in sound/soc/fsl/) -- Timur Tabi Linux kernel developer at Freescale

2 3

[alsa-devel] mm functions vs. normal IO
by Reiner 04 Mar '08

04 Mar '08

Hello, I'am new for using ALSA API. I was astonished that a simple wav playback example does not exist. But after reading the pcm examples I have managed to make one (if I will finish it I can post it so it can be added to the examples) Now my question: What should be used: snd_pcm_mmap_commit <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___direct.html#g…> (I think snd_pcm_mmap_writei <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___direct.html#g…> is the same with an additional copy behind ?) or snd_pcm_writei <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gf13067c0e…> I think mmap will have better performance, but will it work with all soundcards? Or should it be in a config what to use? I'am also looking for a function to get the number of played frames. I have played around with "snd_pcm_status" but it seems not to have a counter for this. By the way is there a detailed tutorial (introduction) for ALSA available? The online help is a little bit too short for newbies. ;-) Reiner

4 3

[alsa-devel] [PATCH] ALC288 - Add NEC S970 to the quirk table
by Pascal Terjan 04 Mar '08

04 Mar '08

Hi, NEC S970 has no sound in the internal speakers when autodetection is used. With targa-dig model, there is sound in the speakers and it gets correctly muted when pluging headphones.

2 1

[alsa-devel] Question about snd_pcm_update_hw_ptr_post()
by Timur Tabi 04 Mar '08

04 Mar '08

Can someone explain to me this snippet from snd_pcm_update_hw_ptr_post(): if (avail >= runtime->stop_threshold) { if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) snd_pcm_drain_done(substream); else xrun(substream); return -EPIPE; } During playback, avail = snd_pcm_playback_avail(). Does this mean that avail is number of frames available in the playback DMA buffer for new data? If so, why is avail >= runtime->stop_threshold considered an underrun condition? Does ALSA try to keep the amount of available data to a minimum? If so, how can I adjust runtime->stop_threshold? I'd like to see what happens if I make stop_threshold equal to the size of my DMA buffer. -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] [PATCH 4/4] Convert non-SoC PXA2xx AC97 driver to clock API
by Mark Brown 04 Mar '08

04 Mar '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-ac97.c | 48 +++++++++++++++++++++++++++++++++++++++------- 1 files changed, 40 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..fd7bc86 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/wait.h> +#include <linux/clk.h> #include <linux/delay.h> #include <sound/core.h> @@ -35,6 +36,10 @@ static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif /* * Beware PXA27x bugs: @@ -112,9 +117,9 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) gsr_bits = 0; #ifdef CONFIG_PXA27x /* PXA27x Developers Manual section 13.5.2.2.1 */ - pxa_set_cken(CKEN_AC97CONF, 1); + clk_enable(ac97conf_clk); udelay(5); - pxa_set_cken(CKEN_AC97CONF, 0); + clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); #else @@ -259,7 +264,7 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); GCR |= GCR_ACLINK_OFF; - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); return 0; } @@ -268,7 +273,7 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; - pxa_set_cken(CKEN_AC97, 1); + clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -335,8 +340,21 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) #ifdef CONFIG_PXA27x /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err; + } #endif - pxa_set_cken(CKEN_AC97, 1); + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err; + } + clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) @@ -361,11 +379,19 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) err: if (card) snd_card_free(card); - if (CKEN & (1 << CKEN_AC97)) { + if (ac97_clk) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; + } +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; } +#endif return ret; } @@ -378,7 +404,13 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) platform_set_drvdata(dev, NULL); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif } return 0; -- 1.5.4.3

2 1

[alsa-devel] [PATCH] Add Invert Switch for ROUT2
by Mark Brown 03 Mar '08

03 Mar '08

From: Graeme Gregory <graeme(a)openmoko.org> GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2 needs to be inverted. This patch adds a mixer control for this. Signed-off-by: Graeme Gregory <graeme(a)openmoko.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0288275..76a5c7b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC", "Channel Swap"}; +static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"}; static const struct soc_enum wm8753_enum[] = { SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), @@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel), +SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase), }; @@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), SOC_ENUM("ADC Data Select", wm8753_enum[27]), +SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; /* add non dapm controls */ -- 1.5.4.3

2 1

[alsa-devel] ALSA C++ API updated for 1.0.16
by Lasse Kärkkäinen 01 Mar '08

01 Mar '08

Here's an updated version of the C++ wrapper that I posted earlier in September 2007. I would like this to be included in the ALSA distribution after some peer review. This wrapper has been used in UltraStar NG (a game shipped by Gentoo and others) since September, with no known issues other than that it is incomplete (very PCM-centric). As a reminder, the API closely follows the C API, and also allows mixing direct C API calls with itself transparently.

4 6

[alsa-devel] [PATCH 4/4] Convert non-SoC PXA2xx AC97 driver to clock API
by Mark Brown 29 Feb '08

29 Feb '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-ac97.c | 48 +++++++++++++++++++++++++++++++++++++++------- 1 files changed, 40 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..fd7bc86 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/wait.h> +#include <linux/clk.h> #include <linux/delay.h> #include <sound/core.h> @@ -35,6 +36,10 @@ static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif /* * Beware PXA27x bugs: @@ -112,9 +117,9 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) gsr_bits = 0; #ifdef CONFIG_PXA27x /* PXA27x Developers Manual section 13.5.2.2.1 */ - pxa_set_cken(CKEN_AC97CONF, 1); + clk_enable(ac97conf_clk); udelay(5); - pxa_set_cken(CKEN_AC97CONF, 0); + clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); #else @@ -259,7 +264,7 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); GCR |= GCR_ACLINK_OFF; - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); return 0; } @@ -268,7 +273,7 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; - pxa_set_cken(CKEN_AC97, 1); + clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -335,8 +340,21 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) #ifdef CONFIG_PXA27x /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err; + } #endif - pxa_set_cken(CKEN_AC97, 1); + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err; + } + clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) @@ -361,11 +379,19 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) err: if (card) snd_card_free(card); - if (CKEN & (1 << CKEN_AC97)) { + if (ac97_clk) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; + } +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; } +#endif return ret; } @@ -378,7 +404,13 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) platform_set_drvdata(dev, NULL); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif } return 0; -- 1.5.4.3

1 0

[alsa-devel] [PATCH] hda: incorrect DAC mixer on 92hd71bxxx
by Matthew Ranostay 29 Feb '08

29 Feb '08

If there was two line_outs[] it would overrun into the dmic nid struct below and create a phantom "Surround" mixer. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 54a51b975975 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 28 12:46:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Feb 28 16:52:43 2008 -0500 @@ -246,8 +246,8 @@ static hda_nid_t stac92hd71bxx_dmux_nids 0x1c, }; -static hda_nid_t stac92hd71bxx_dac_nids[1] = { - 0x10, /*0x11, */ +static hda_nid_t stac92hd71bxx_dac_nids[2] = { + 0x10, 0x11 }; #define STAC92HD71BXX_NUM_DMICS 2 @@ -2345,10 +2345,10 @@ static int stac92xx_auto_create_multi_ou for (i = 0; i < cfg->line_outs; i++) { - if (!spec->multiout.dac_nids[i]) + + nid = spec->multiout.dac_nids[i]; + if (!nid || nid == spec->multiout.hp_nid) continue; - - nid = spec->multiout.dac_nids[i]; if (i == 2) { /* Center/LFE */ @@ -3550,7 +3550,7 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 1; + spec->multiout.num_dacs = ARRAY_SIZE(stac92hd71bxx_dac_nids); spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids;

2 3

[alsa-devel] [PATCH] hda: 92HD
by Matthew Ranostay 29 Feb '08

29 Feb '08

Added core_init[] for several 92hd73xxx laptops. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 54a51b975975 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 28 12:46:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Feb 28 21:14:39 2008 -0500 @@ -525,6 +525,24 @@ static struct hda_verb stac92hd73xx_6ch_ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb dell_m6_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, /* setup import muxs */ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -3437,6 +3455,7 @@ again: switch (spec->board_config) { case STAC_DELL_M6: + spec->init = dell_m6_core_init; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f:

2 2

[alsa-devel] [PATCH] IEC958 definitions for consumer status channel, byte 4
by Pawel MOLL 29 Feb '08

29 Feb '08

Hi, Attached are patches for alsa-kernel and alsa-lib, adding mentioned definitions. Regards Paweł

2 1

[alsa-devel] [PATCH] hda: disable power management on fixed ports.
by Matthew Ranostay 29 Feb '08

29 Feb '08

Power management can't be enabled on fixed ports, since the presence will always return false and prevent output. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 54a51b975975 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 28 12:46:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Feb 28 20:39:28 2008 -0500 @@ -3015,11 +3015,15 @@ static int stac92xx_init(struct hda_code ? STAC_HP_EVENT : STAC_PWR_EVENT; int pinctl = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) + continue; + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26);

2 1

[alsa-devel] [PATCH] hda/patch_sigmatel.c Panasonic CF-74 speaker output
by Kyler Laird 29 Feb '08

29 Feb '08

This patch causes ALSA to send sound to the speakers of a Panasonic CF-74 (Toughbook) instead of just its headphone jack. --kyler diff -Naur linux-2.6.24.1/sound/pci/hda/patch_sigmatel.c linux-2.6.24.1-ALSA_CF74/sound/pci/hda/patch_sigmatel.c --- linux-2.6.24.1/sound/pci/hda/patch_sigmatel.c 2008-02-08 14:55:30.000000000 -0500 +++ linux-2.6.24.1-ALSA_CF74/sound/pci/hda/patch_sigmatel.c 2008-02-11 12:05:11.824064617 -0500 @@ -2467,6 +2467,11 @@ spec->init = stac9200_core_init; spec->mixer = stac9200_mixer; + /* Kyler - fix for CF-74 speaker */ + spec->gpio_mask = 0x00000009; + spec->gpio_data = 0x00000000; + stac92xx_enable_gpio_mask(codec); + err = stac9200_parse_auto_config(codec); if (err < 0) { stac92xx_free(codec);

2 1

Re: [alsa-devel] How to set idle S/PDIF sample rate?
by Demian Martin 29 Feb '08

29 Feb '08

I noticed another big problem in my application relating to locking the system at 48 KHz in the background. If I set SPDIF to 44.1KHz on some playback software (Xine for example) between tracks the system briefly switches back to 48 KHz. This causes an audible pop as the external DAC tries to keep up with the rate change. I have not found a way to lock the system at a different rate (44100 for example) so this doesn't happen between tracks. This behavior is common to both the ICE1724 and the VT1708 drivers. -Demian Date: Thu, 28 Feb 2008 16:12:10 -0500 From: Matt Whitlock <matt(a)whitlock.name> Subject: Re: [alsa-devel] How to set idle S/PDIF sample rate? To: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Message-ID: <200802281612.11099.matt(a)whitlock.name> Content-Type: text/plain; charset="iso-8859-1" On Thursday, 28 February 2008 6:43:25 am Takashi Iwai wrote: > The driver code resets the format verb of the relevant widgets at > closing streams. This verb contains the sample rate, too. Maybe > keeping this would prevent the reset. > > The below is a test patch. Give it a try. Wow, fantastic. I will give this patch a try. I'll have to rebuild my kernel to disable the in-kernel ALSA code and build the ALSA module separately from the latest sources, but if this patch fixes things, that would be worth the effort. I'll get back to you. Thanks. -Matt

2 1

[alsa-devel] How to set idle S/PDIF sample rate?
by Matt Whitlock 28 Feb '08

28 Feb '08

When no programs have the sound hardware open, ALSA drives my Intel HDA's S/PDIF output with silence at 48000 Hz. How would I change this default so that the idle sample rate is 44100 Hz? The reason I ask is that my stereo receiver misses the first second or so of each song that I play as it switches from 48000 Hz to 44100 Hz. I understand that by switching the idle sample rate to 44100 Hz, I would have the same problem whenever playing a DVD, but I can much more happily accept the switching delay when starting DVDs than I can when starting music playback. I've tried everything, including replacing /usr/share/alsa/alsa.conf with a single PCM definition that outputs straight to hardware with "rate 44100", but still, whenever nothing is accessing the sound hardware, ALSA reverts to 48000-Hz silence. There must be a way to influence that, right? Alternatively, if ALSA would simply not output anything over S/PDIF when idle, my receiver could just stay at 44100 Hz, albeit unlocked since there would be no data stream at all. So this is also an acceptable solution to me, but I cannot find any way to make ALSA totally shut up when idle. It always wants to output silence. Interestingly, if I mute the IEC958 playback channel and then unmute it, my receiver reports no signal at all until I play a sound, and then after the sound finishes, my receiver reports that it is locked onto PCM at 48000 Hz. Seems kind of like a bug, doesn't it? Thanks to anyone who can lend some assistance.

3 10

[alsa-devel] [PATCH 0/2] Fixes to couple DAPM widget functions
by Jarkko Nikula 28 Feb '08

28 Feb '08

API change done by the commit 9af6d9562414568ecadf96aaef5b88e7e8b19821, "[ALSA] soc - Add support for passing kcontrols with events", doesn't change affected pxa machine and wm9712 drivers. These two patches are generated from master branch of linux-2.6-asoc tree but they apply to alsa git and alsa-kernel hg trees as well. I separated wm9712 to own patch since it is ok in asoc-v2-dev branch. There are other drivers as well with this error in asoc-v2-dev but I don't know are they already in progress? I can post a another patch to them if needed. -- Jarkko

3 6

[alsa-devel] Using ARRAY_SIZE macro when setting up ASoC audio map?
by Jarkko Nikula 28 Feb '08

28 Feb '08

Does an example patch to spitz below make any sense? Yep, noticed when forgot to add array terminator to own code :-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d56709e..1cffcd3 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -268,8 +268,6 @@ static const char *audio_map[][3] = { /* line is connected to input 1 - no bias */ {"LINPUT1", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -317,7 +315,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) } /* Set up spitz specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; i < ARRAY_SIZE(audio_map); i++) { snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); }

3 5

Re: [alsa-devel] a7839e96 (PNP: increase max resources) breaks my ALSA intel8x0 sound card
by Robert Hancock 27 Feb '08

27 Feb '08

Avuton Olrich wrote: > With v2.6.24 my second ALSA sound device stopped working. > > After bisection it says this was the offending commit. > > a7839e960675b549f06209d18283d5cee2ce9261 is first bad commit > commit a7839e960675b549f06209d18283d5cee2ce9261 > Author: Zhao Yakui <yakui.zhao(a)intel.com> > Date: Wed Nov 28 16:21:21 2007 -0800 > > PNP: increase the maximum number of resources > > On some systems the number of resources(IO,MEM) returnedy by PNP device is > greater than the PNP constant, for example motherboard devices. It brings > that some resources can't be reserved and resource confilicts. This will > cause PCI resources are assigned wrongly in some systems, and cause hang. > This is a regression since we deleted ACPI motherboard driver and use PNP > system driver. > > [akpm(a)linux-foundation.org: fix text and coding-style a bit] > Signed-off-by: Li Shaohua <shaohua.li(a)intel.com> > Signed-off-by: Zhao Yakui <yakui.zhao(a)intel.com> > Cc: Bjorn Helgaas <bjorn.helgaas(a)hp.com> > Cc: Thomas Renninger <trenn(a)suse.de> > Cc: <stable(a)kernel.org> > Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> > Signed-off-by: Linus Torvalds <torvalds(a)linux-foundation.org> > > The audio device is 00:1b.0 (see my lspci -vvv output), the other > audio device works fine. > > http://avuton.googlepages.com/v2.6-before (dmesg revision before) > http://avuton.googlepages.com/v2.6-after (dmesg broken revision) > http://avuton.googlepages.com/lspci-vvv > http://avuton.googlepages.com/config (from the broken revision) > http://avuton.googlepages.com/iomem > http://avuton.googlepages.com/ioports Here's why the driver fails to load: [ 31.133060] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 [ 31.133141] PCI: Unable to reserve mem region #1:4000@febf8000 for device 0000:00:1b.0 [ 31.133197] ACPI: PCI interrupt for device 0000:00:1b.0 disabled [ 31.133244] HDA Intel: probe of 0000:00:1b.0 failed with error -16 The iomem location of the HDA controller conflicts with this reservation by the BIOS: [ 22.906654] system 00:08: iomem range 0xfebfa000-0xfebfac00 has been reserved There was a patch floating around to ignore PnPACPI reservations which conflict with PCI BARs, which appears to be what's happening in this case. That patch originally worked for any board, but was later made specific to a certain Supermicro motherboard which had the sata_nv controller MMIO regions marked as reserved, preventing the driver from loading. We may need a more general solution. See: https://bugzilla.redhat.com/show_bug.cgi?id=313491

5 24

[alsa-devel] HDMI Multichannel LPCM/RAW Support/Status
by Dustin Kerstein 27 Feb '08

27 Feb '08

I was just curious if there is any planned support for multichannel LPCM/Raw audio for newer HDMI based chipsets. There are several matx boards with HDMI 1.1 and above compliance, which means they should be able to pass 8-channel LPCM audio. Also, in HDMI 1.3 and above, Bit Stream passing is supported, which actually should be easier, bandwidth wise, than LPCM. Multichannel PCM (above 2 channels) was never an issue because SPDIF lacked the bandwidth. After talking with some people in irc it seems as though the # of channels setting in Alsa is only one bit. Thanks. Dustin

2 2

Re: [alsa-devel] PATCH for alsa-info.sh - using mktemp for instead of hardcoded temp dir.
by Takashi Iwai 27 Feb '08

27 Feb '08

At Thu, 28 Feb 2008 02:42:20 +1000, Travis Place wrote: > > On Thu, 28 Feb 2008 2:39:16 am you wrote: > > At Thu, 28 Feb 2008 00:36:52 +1000, > > > > Travis Place wrote: > > > Email subject should explain well enough. > > > > I see some problems with this change: > > - it doesn't work if /tmp/alsainfo doesn't exist > > - temp directory isn't removed if --no-upload is given > > > > I think you should avoid --tempdir=xxx with mktemp. > > > > The output file name is another question. > > IMO, it'd be better to pass the output file name via a command line > > option, and put it on the home directory (e.g. ~/alsa-info.out) as > > default rather than /tmp. > > well, /tmp/alsainfo/ will exist, as we create it in the script.. its > $ALSAINFODIR .. the temp dir is then created inside that. It's created too late. And, if we create /tmp/alsainfo, there is little merit of mktemp as /tmp/alsainfo could be a symlink. Simply TEMPDIR should be what mktemp creates, and clean it up after quitting. > As for not being cleaned up, if --no-upload is specified, ill check that now.. > > Also, is outputting the resulting file to ~/ wise ? It's at least safer than the fixed filename on /tmp. I don't insist on this option, though. Anyway, --output-file=xxx would be a nice addition. > should this occur even if the result is uploaded, or only on > --no-upload ? I don't think it's needed when uploaded (except for any debugging reason). Just my $0.02. thanks, Takashi

2 1

[alsa-devel] PATCH for alsa-info.sh - using mktemp for instead of hardcoded temp dir.
by Travis Place 27 Feb '08

27 Feb '08

Email subject should explain well enough. -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 2

[alsa-devel] new driver: channel alignment (was: 30mS interval problem)
by William Juul 27 Feb '08

27 Feb '08

Hello I am new to ALSA and trying to write a new driver for a ADC connected with PCM to an AVR32 on a NGW100 reference card provided by Atmel. I have the problem that different channels from the PCM stream comes in different positions in the WAV file. For example when I disable all but the first PCM channel on the ADC, I can see with an oscilloscope that there is only data in the first timeslot on the PCM data stream. But the data from this channel can occur in any of the 4 channels in the WAV file. This is consistent throughout the one capture/file, but in the next capture the data will occur in another channel in the WAV-file. How can I make sure that the data from one particular channel always occur in a fixed position in the WAV file? Best regards William Juul

1 0

[alsa-devel] [ALC262 patch_realtek.c] Add Fujitsu Lifebook E8410 to quirk table
by Tony Vroon 27 Feb '08

27 Feb '08

Good afternoon, Could you please consider the following patch for 2.6.25-rc4. The BIOS autoprobe doesn't do a very good job setting this card up. Card details: 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 10cf:142d (and verbose codec information attached) Regards, Tony V. LINX sysadmin

3 2

Re: [alsa-devel] Ends up with write error (Ning Zhang)
by Ning Zhang 27 Feb '08

27 Feb '08

Hi all, There is no response so far, maybe I am not clear. So in sum, here are the questions: 1. In meddle-level driver, how comes the runtime->control->avail_min? I mean which function set the value, and according what to set the value? Does it have some relationship to the hardware configuration in the device driver( i.e snd_pcm_hardware )? 2. Should I use the snd_pcm_period_elapsed function even I don't use the interrupt mechanism? If yes, In which callback function should I call it(snd_pcm_period_elapsed)? 3,Can anybody do me a favor to explain the mechanism of the middle-level driver or give me some hints? Thanks in advance. BRs, David > > ------------------------------ > > Message: 7 > Date: Wed, 20 Feb 2008 16:30:54 +0800 > From: "Ning Zhang" <ning.zhang06(a)gmail.com> > Subject: [alsa-devel] Ends up with write error > To: alsa-devel <alsa-devel(a)alsa-project.org> > Message-ID: > <48a1cba90802200030v65794b96j7c956406fdc0df9c(a)mail.gmail.com> > Content-Type: text/plain; charset=GB2312 > > Hi all, > I am writing a ALSA driver for my ARM 926 platform. The ARM communicates > with the DSP by accessing a 128-byte shared memory. I don't use IRQ > mechanism.(In fact, I am not sure the platform has this mechanism) > Here is some code slices: > > static struct snd_pcm_hardware snd_arch_playback_hw = { > > .info = (SNDRV_PCM_INFO_MMAP | > > SNDRV_PCM_INFO_INTERLEAVED | > > SNDRV_PCM_INFO_BLOCK_TRANSFER), > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ > > SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ > > SNDRV_PCM_RATE_48000), > > .rate_min = 8000, > > .rate_max = 48000, > > .channels_min = 2, > > .channels_max = 2, > > .buffer_bytes_max = 128, > > .period_bytes_min = 2, > > .period_bytes_max = 128, > > .periods_min = 1, > > .periods_max = 64, > > }; > > > > static snd_pcm_uframes_t snd_arch_pcm_pointer(struct snd_pcm_substream > > *substream) > > { ? > > /* > > * get the current hardware pointer > > */ > > current_ptr = arch_fw_read_dsp_reg(chip, AFE_RWADDR) & AFE_RD_ADDR; > > > return bytes_to_frames(runtime, current_ptr); > > } > > static int snd_arch_pcm_playback_copy(struct snd_pcm_substream *substream, > > int channel, > > snd_pcm_uframes_t pos, > > void __user *src, > > snd_pcm_uframes_t count) > > { ?? > > > > count = frames_to_bytes(runtime, count); > > pos = frames_to_bytes(runtime, pos); > > chip->bufpos = pos; > > ?. > > spin_lock(chip->reg_lock); > > if ( count > 0 && count <= MAX_BUF_SIZE ) { > > if ( copy_from_user(chip->bufptr, src, count) ) { > > snd_printk(KERN_ERR "can not copy user data!\n"); > > return -EFAULT; > > } > > if ( arch_fill_rx_buf(chip, > > (signed short*)(chip->bufptr), > count) < 0 ) { > > snd_printk(KERN_ERR "fill rx buffer failed!\n"); > > return -1; > > } > > ????? > > } > spin_unlock(chip->reg_lock); > > return 0; > > } > > The problem is my driver can make sounds, but ends up a write > error. After > much debugging, I think there might be some problem in the > interworking between > middle-level driver and the device driver. > > I found the runtime->control->avail_min=32, and the > runtime->buffer_size and runtime->period_size are both 32. So at first > middle-level driver call snd_pcm_lib_write1 transfers 32 frames(128 bytes) > to the device driver's copy callback function and trigger function. Then > when middle-level driver wants to transfer another 32 frames, it calls the > pointer callback function and gets the position, because at this time the > whole bytes has not been processed completely, so the pointer function's > return value must less than 32, and the avail must less than > runtime->control_->avail_min, and goes to this flow and ends up a write > error. > > > _______________________________________________________________________________ > > if (((avail < runtime->control->avail_min && size > avail) || > > (size >= runtime->xfer_align && avail < > runtime->xfer_align))) { > > ?????... > > long tout; > > init_waitqueue_entry(&wait, current); > > add_wait_queue(&runtime->sleep, &wait); > > while (1) { > > if (signal_pending(current)) { > > state = SIGNALED; > > break; > > } > > set_current_state(TASK_INTERRUPTIBLE); > > snd_pcm_stream_unlock_irq(substream); > > tout = schedule_timeout(10 * HZ); > > snd_pcm_stream_lock_irq(substream); > > if (tout == 0) { > > if (runtime->status->state != > SNDRV_PCM_STATE_PREPARED && > > runtime->status->state != > SNDRV_PCM_STATE_PAUSED) { > > state = > runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; > > break; > > } > > } > > switch (runtime->status->state) { > > > ????.. > > : > > state = ERROR; > > goto _end_loop; > > > > default: > > break; > > } > > avail = > snd_pcm_playback_avail(runtime); > > if (avail >= > runtime->control->avail_min) > { > > state = READY; > > break; > > } > > } > > _end_loop: > > remove_wait_queue(&runtime->sleep, &wait); > > > > switch (state) { > > ??. > > case EXPIRED: > > snd_printd("playback write error (DMA > or > IRQ trouble?)\n"); > > err = -EIO; > > goto _end_unlock; > > ??? > > } > > } > > This code is a part of snd_pcm_lib_write1() in sound/core/pcm_lib.c > > I think that I misunderstood the mechanism of middle-level driver. Could > you please correct me? > > What should the pointer callback function return? What's it for? > > How comes the runtime->control->avail_min? > > Because I don't use the IRQ, so should I use the snd_pcm_period_elapsed > function? And where? > > ALSA Version is 1.0.14rc1 and I use alsa-lib-1.0.15. > > Can someone comment on this and give me some hints to solve this > problem? Thanks > in advance. > > BRs, > > David > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 12, Issue 69 > ****************************************** >

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Re: [alsa-devel] Via VT82xx VT1708 ICE1724HT and higher
by Demian Martin 27 Feb '08

27 Feb '08

Pavel: Thanks for the insight. I thought I was doing something wrong since no one else has noticed. I tried your suggestion to use plughw:0,1 and the verbose output looks fine, the data rate shifts fine (I'm watching the L/R clock on the input to the DAC on an external DAC) but no audible sound comes out the converter unless it's a 16 bit 44.1 file. The 24 bit stuff is not getting converted into something useful to the DAC. Should I check to see if the data clock is 48X the LR clock? What info do you need from me to look deeper? I can supply a sample clip of a 176.4 KHz recording from Reference Recordings if it would be helpful. I'm also testing with the samples from Linn and Gimmel. I looked at the Plug plugin but have not been able to figure out how to configure it for my needs. There is way too much marginal information on the web and little that directs me to an optimum setting. -Demian From: Pavel Hofman <pavel.hofman(a)insite.cz> Subject: Re: [alsa-devel] Via VT82xx VT1708 ICE1724HT and higher sample rate/bit depth files through SPDIF To: Demian Martin <demianm_1(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Message-ID: <47C423E4.4060900(a)insite.cz> Content-Type: text/plain; charset=windows-1252; format=flowed Hi Martin, > > First- sample rate > > Even though the documents claim that it will play at 176.4 KHz and 192KHz > whenever I play a file at these rates it down converts to ? the rate (88200 > and 96000). I tested this with the ICE1724HT (Juli@) which has almost the > same core, I never could get 176.4KHz at the SPDIF output. I confirmed that > the ICE1724 will play the file at 176.4KHz in Windows so its not a hardware > limitation. I tried to do the same with the VT1708 (on a Via CX700 chipset) > and the Windows driver wouldn?t let me at anything other than 48000. I spent a few days on this issue and confirm that ICE1724HT cannot output 176.4kHz from its internal SPDIF transmitter, WHEN clocked by its internal clock circuits. A scope hooked to corresponding pins/outputs revealed the following: All I2S lines output correct 176.4kHz signal, i.e. D/A codecs receive proper signal. However, I did not find a way to force ICE1724 to output 176.4kHz, it always drops to 88.2kHz, no matter what MT01/MT02 combination is. The same happens with windows drivers of my Prodigy 192. The sequence is: 44.1, 48, 88.2, 96, 88.2, 192 I have not seen any manufacturer of Envy-based cards revealing this deficiency. The "preliminary" datasheet we all use mentions only that the chip forces the 128x mode when switched to 176.4kHz. I believe the output of 88.2kHz is an undocumented feature or perhaps even a bug of the chip. Juli is a card whose manufacturer may be aware of the problem. It is not clocked by ICE1724 internal clocks fed by the two crystals. Instead, the crystals also provide signal to external circuits generating clock signals of required frequency, feeding the chip in slave mode. The circuitry is controlled by ice1724 GPIOs. This way the ICE1724 is not aware of its clocking frequency and cannot thus mis-configure its SPDIF transmitter. The windows driver does output proper 176.4kHz. I am working on Juli driver and have been analyzing this beast for a few nights. The special clocking is already working at home :) > Second, the only way I can get the 24 bit files out in a form that the card > can play is to pass them through dmix. But I need to reconfigure dmix for > each sample rate. And I?m not convinced that it isn?t resampling twice, both > at the input and the output. Or how to configure the buffers etc. to prevent > underruns even when the only conversion is to pass a 24 bit file to the > card. ICE cards use 32bits format. Try aplay -v -D plughw:0 24bit.wav and you will see the conversion to 32 bits done by the plug plugin, no dmix involved. > > I?m trying to build a high quality music server that will pass the files to > either the internal or an external DAC at the bit depth and sample rate they > were recorded at. I have tried to understand the various plugs and settings > but it seems they are all focused on mixing various streams. I want to send > the files unmixed to the output. I can?t figure out what set of plugs and > settings in asound.conf would enable this. Check the plug plugin. Regards, Pavel Hofman. -----------------

3 2

[alsa-devel] ATI HD 2600 Sound?
by Isaac Manuel 26 Feb '08

26 Feb '08

HP Pavilion HDX9200 Laptop PC Ubuntu 7.10 (Gutsy Gibbon) Linux Kernel 2.6.22-14-generic Sound only play through headphones. *aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 0: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 *sudo lspci -v *Audio device: ATI Technologies Inc Unknown device aa08 Subsystem: Hewlett-Packard Company Unknown device 30d4 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at e8310000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Capabilities: [58] Express Legacy Endpoint IRQ 0 Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- * **sudo lspci -nv* 0403: 1002:aa08 Subsystem: 103c:30d4 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at e8310000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Capabilities: [58] Express Legacy Endpoint IRQ 0 Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable-

1 0

[alsa-devel] hdsp firmware
by Karl Grill 26 Feb '08

26 Feb '08

Hi all, as a first step towards implementing support for the hdsp rpm, I've uploaded a patch to the alsa-firmware branch as bug report #3810. Besides the patch, I've put up the firmware binaries in .tar.gz format. This contains replacements for the rev.11 firmware for the digiface/multiface and also for the rpm. The latter is of little use right now, but I'd be very happy if anybody could try those for the digiface/multiface and report if it works, so I know if my assumptions on the drivers are correct. best regards kg PS. sorry I didn't file this report in the firmware section, but the web form there doesn't give me any usable choice for Category yet insists that I supply one.

1 0

[alsa-devel] Zoom H4 hand held studio.
by Joseph klitzke 26 Feb '08

26 Feb '08

Greetings I am inquiring into the development of a driver for the Zoom H4 Hand held recorder. Is there any work being done on the device? If not is it being considered. Is there anything I can do to get it started? In short the H4 can 4 track recorder and a usb audio device with effects, a great piece of recording gear. Thanks Joseph Klitzke

1 0

[alsa-devel] snd-usb-audio syncronisation problem
by Adrian Stancu 26 Feb '08

26 Feb '08

Hi, I'm using an E-MU 0404 USB interface with the snd-usb-audio module from alsa 1.0.16, operating in isochronous asynchronous mode. This interface was producing very annoying click/pop sounds at regular intervals during playback. Eventually I discovered that the syncronisation mechanism telling the driver to send audio data faster or slower to match the interface's consumption speed does not work as expected, for some reason. I did a very dirty hack within the source of this driver, simulating the interface asking for data to be sent a little bit quicker, and this almost fixed the problem *for me*. I really wish to debug this issue further, to find what actually does not work as it should in this feed-back mechanism. However this was the first time I dealt with alsa source code, so I do not have much experience. Therefore, I really need help with debugging this problem and maybe finding an universal fix. Thanks in advance, Adi

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[alsa-devel] HDA-Intel - Patch to support RV7xx HDMI Audio
by Yang, Libin 26 Feb '08

26 Feb '08

Hi Takashi, This patch is to add R7xx HDMI audio support. Would you please accept it? Thanks a lot! Signed-off-by: Libin Yang <Libin.yang(a)amd.com> --- alsa-driver-1.0.16/alsa-kernel/pci/hda/hda_intel.c.orig 2008-02-13 11:57:11.000000000 -0500 +++ alsa-driver-1.0.16/alsa-kernel/pci/hda/hda_intel.c 2008-02-13 12:02:39.000000000 -0500 @@ -2015,6 +2015,9 @@ { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */ { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */ { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */ + { 0x1002, 0xaa38, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV730 HDMI */ + { 0x1002, 0xaa40, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV710 HDMI */ + { 0x1002, 0xaa48, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV740 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ Thanks & Regards, Libin

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[alsa-devel] Via VT82xx VT1708 ICE1724HT and higher sample rate/bit depth files through SPDIF
by Demian Martin 26 Feb '08

26 Feb '08

I have several problems trying to play files higher than 48000 KHz and with bit depth of 24 bits. I submitted this to the users and got a resounding silence. First- sample rate Even though the documents claim that it will play at 176.4 KHz and 192KHz whenever I play a file at these rates it down converts to ½ the rate (88200 and 96000). I tested this with the ICE1724HT (Juli@) which has almost the same core, I never could get 176.4KHz at the SPDIF output. I confirmed that the ICE1724 will play the file at 176.4KHz in Windows so its not a hardware limitation. I tried to do the same with the VT1708 (on a Via CX700 chipset) and the Windows driver wouldnt let me at anything other than 48000. Second, the only way I can get the 24 bit files out in a form that the card can play is to pass them through dmix. But I need to reconfigure dmix for each sample rate. And Im not convinced that it isnt resampling twice, both at the input and the output. Or how to configure the buffers etc. to prevent underruns even when the only conversion is to pass a 24 bit file to the card. Im trying to build a high quality music server that will pass the files to either the internal or an external DAC at the bit depth and sample rate they were recorded at. I have tried to understand the various plugs and settings but it seems they are all focused on mixing various streams. I want to send the files unmixed to the output. I cant figure out what set of plugs and settings in asound.conf would enable this.

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[alsa-devel] [PATCH 1/3] ASoC: Fix TLV320AIC3X PLL divider table for 64 kHz rate
by Mark Brown 26 Feb '08

26 Feb '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 710e028..569ecac 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -681,8 +681,8 @@ static const struct aic3x_rate_divs aic3x_divs[] = { {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ - {22579200, 96000, 96000, 0x1, 8, 7075}, - {33868800, 96000, 96000, 0x1, 5, 8049}, + {22579200, 64000, 96000, 0x1, 8, 7075}, + {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, -- 1.5.4.3

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[alsa-devel] new driver: 30mS interval problem
by William Juul 26 Feb '08

26 Feb '08

Hello I am new to ALSA and trying to write a new driver for a DAC connected with PCM to an AVR32 on a NGW100 reference card provided by Atmel. The sampling rate I am currently using is 11047and the DAC is providing 4 channels of 24bit. The HW interface is using DMA to copy data to RAM. By studying the audio data in hexdump or in Audacity I can verify that the sound looks good in intervals of about 30mS, then all channels are garbled for 30mS. This pattern repeat itself throughout the audio capture. I am not confident I have configured all ALSA parameters properly. How can I go about fixing/debugging this 30mS intverval problem? Below is the command I am using. Best regards William Juul # arecord -r 11047 -c 4 -f S24_LE -s 1 -A 100 -d 5 --buffer-size 16384 -F 21333 -v > test.wav Recording WAVE 'stdin' : Signed 24 bit Little Endian, Rate 11047 Hz, Channels 4 Plug PCM: Hardware PCM card 0 'AVR32 NGW100 external DAC' device 0 subdevice 0 pcm->setup: 1 stream : CAPTURE access : RW_INTERLEAVED format : S24_LE subformat : STD channels : 4 rate : 11047 exact rate : 11047 (11047/1) msbits : 32 buffer_size : 16384 period_size : 490 period_time : 44355 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 490 xfer_align : 490 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824

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Re: [alsa-devel] ASoC - Output widgets
by Mark Brown 26 Feb '08

26 Feb '08

On Tue, Feb 26, 2008 at 10:11:42AM +0100, Alexandre BOUIN wrote: > DAPM WM8900 snd_soc_dapm_stream_event > FM Transmitter: On in 2 out 1 > in static FMOUT_R > in static FMOUT_L > I initialize FM Transmitter like this : > snd_soc_dapm_set_endpoint(codec, "FM Transmitter", 0); > but it still enabled. Why ? According to the above trace your FM transmitter is still reporting as being connected. Have you checked to ensure that your your call to configure the transmitter is happening after your board widgets have been registered?

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Re: [alsa-devel] ASoC - Output widgets
by Mark Brown 26 Feb '08

26 Feb '08

On Mon, Feb 25, 2008 at 10:39:15AM +0100, Alexandre BOUIN wrote: > - 1 SND_SOC_DAPM_MIXER is connected to 2 SND_SOC_DAPM_PGA with power > control. > - each PGA is connected to an SND_SOC_DAPM_OUTPUT > - according to tosa example : > - first OUTPUT is connected to a SND_SOC_DAPM_HP defined as endpoint > - second OUTPUT is connected to a SND_SOC_DAPM_SPK defined as endpoint > this SPK is set disconnected in my tosa_ac97_init -like function Could you please post dmesg output with DAPM_DEBUG turned on which shows this happening?

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Re: [alsa-devel] [Alsa-devel] Questions about sound/isa/sb/sb8_main.c
by Clemens Ladisch 26 Feb '08

26 Feb '08

Rask Ingemann Lambertsen wrote: > I'm looking at sound/isa/sb/sb8_main.c from linux 2.6.20. In > snd_sb8_capture_prepare(), a line reads: > > if (chip->capture_format != SB_DSP_OUTPUT) { > > Shouldn't this be SB_DSP_INPUT instead? Yes. Now fixed in the ALSA tree. Thanks! Clemens

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[alsa-devel] POLLIN and POLLOUT event in pcm io plugins
by Pharaoh . 26 Feb '08

26 Feb '08

Hi, The structure snd_pcm_ioplug has following members: int poll_fd; /**< poll file descriptor */ unsigned int poll_events; /**< poll events */ and the associated events are POLLIN and POLLOUT, I suppose. I am unable to understand how these fields are getting used. There are many other plugins which dont use this. What is the significance of these fields? -pharaoh.

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[alsa-devel] Fwd: HW decoding abstracted in ALSA
by Harsha priya gupta 25 Feb '08

25 Feb '08

Hi, Thanks for your immediate reply. My hardware supports MP3 decoding in HW (in addition to few others). I am looking to write a driver such that when an MP3 file is played, the file is decoded using the HW decoder and played. Is there a possibility in current ALSA framework to do so? Thanks, Harsha. On Mon, Feb 25, 2008 at 5:33 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Mon, 25 Feb 2008 12:27:00 +0530, > Harsha priya gupta wrote: > > > > Hi, > > > > I am new to ALSA and audio stack in Linux. I have a sound card that can > do > > HW decoding as well. I am looking at writing an ALSA driver such that > > whatever may be the player or MMF above, it should use the HW decoder > and > > not the inherit SW decoder that is available with the player. > > > > Can someone help me with some pointers or existing like-drivers? > > Which format is in mind? In theoery, you can implement the h/w > decoder on the ALSA framework, but there are some restrictions: > > - ALSA PCM assumes the constant bitrate > - The format bit must be defined > - The specific app recognizes and supports that format > > In the case of AC3 or DTS, it's embedded in SPDIF frames so the PCM > core stuff doesn't take care of the above. But, in the case of native > (non-PCM) formats, the support can be difficult as is. > > > Takashi > -- -Harsha -- -Harsha

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[alsa-devel] Patch for alsa-info.sh - AC97 including regs
by Travis Place 25 Feb '08

25 Feb '08

Ok, ive added the +regs file aswell, but decided to store its output in a seperate tmp file (before being added to the complete output). This way, we can change how its output if we'd like, with less stuffing around (and all tmp files are cleaned up after each run, anyway). -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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Re: [alsa-devel] hdsp rpm support
by kgrill＠chello.at 25 Feb '08

25 Feb '08

I started to work on this about a year ago, but had to stop because my son took the card to Istanbul where he spent a year studying. I'll get back to it in the next days, and depending on how much has changed in the meantime I may get something ready by the end of January. regards kg

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[alsa-devel] Patch for alsa-info.sh
by Travis Place 25 Feb '08

25 Feb '08

Sorry for the frequent patch interval.. I think this will do for a while.. Adds AC97 codec output if an AC97 card is in the system. -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] How does ALSA detect underrun in OSS emulation?
by Timur Tabi 25 Feb '08

25 Feb '08

Can someone explain to me the exact method that ALSA uses to detect underrun with OSS emulation? I have a driver that reports underrun on almost every period when I use OSS emulation and I'm playing with a non-supported sample rate. So I'm assuming that maybe my hardware is playing the audio too fast or too slow, and the driver is returning periods before ALSA expects them. What I don't understand is: how does ALSA know that a period was finished too early. Does it use a timer, or is it purely application driven? -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] [alsa-cvslog] alsa-lib: Fix the state in snd_pcm_ioplug_pause()
by Thierry Vignaud 25 Feb '08

25 Feb '08

Takashi Iwai <tiwai(a)suse.de> writes: > changeset: 2423:7388ad3d21ea > tag: tip > user: tiwai > date: Mon Feb 25 15:11:46 2008 +0100 > files: src/pcm/pcm_ioplug.c > description: > Fix the state in snd_pcm_ioplug_pause() > > The states[] in snd_pcm_ioplug_pause() has wrong values. They should be > swapped. ALSA bug#3796: > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3796 Aren't some bits missing from this bug report? (s/o->data->state = states[enable]/io->data->state = states[prev]) > diff -r 6d83f965a9d9 -r 7388ad3d21ea src/pcm/pcm_ioplug.c > --- a/src/pcm/pcm_ioplug.c Mon Feb 25 14:58:45 2008 +0100 > +++ b/src/pcm/pcm_ioplug.c Mon Feb 25 15:11:46 2008 +0100 > @@ -484,7 +484,7 @@ static int snd_pcm_ioplug_pause(snd_pcm_ > { > ioplug_priv_t *io = pcm->private_data; > static snd_pcm_state_t states[2] = { > - SND_PCM_STATE_PAUSED, SND_PCM_STATE_RUNNING > + SND_PCM_STATE_RUNNING, SND_PCM_STATE_PAUSED > }; > int prev, err; > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog

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[alsa-devel] [PATCH] hdsp: Remove all eol tabs/spaces and space stuffing
by Andrew Paprocki 25 Feb '08

25 Feb '08

I have one of these cards and was looking through the file and saw lots of eol tabs/spaces as well as some space stuffing on various lines. diff -r 8087414481e8 -r 87b21007d0f9 pci/rme9652/hdsp.c --- a/pci/rme9652/hdsp.c Sat Feb 23 11:34:12 2008 +0100 +++ b/pci/rme9652/hdsp.c Sat Feb 23 14:01:54 2008 -0500 @@ -90,12 +90,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11 #define HDSP_inputBufferAddress 36 #define HDSP_controlRegister 64 #define HDSP_interruptConfirmation 96 -#define HDSP_outputEnable 128 +#define HDSP_outputEnable 128 #define HDSP_control2Reg 256 -#define HDSP_midiDataOut0 352 -#define HDSP_midiDataOut1 356 -#define HDSP_fifoData 368 -#define HDSP_inputEnable 384 +#define HDSP_midiDataOut0 352 +#define HDSP_midiDataOut1 356 +#define HDSP_fifoData 368 +#define HDSP_inputEnable 384 /* Read registers. These are defined as byte-offsets from the iobase value */ @@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11 /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ @@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11 26*3 values are read in ss mode 14*3 in ds mode, with no gap between values */ -#define HDSP_9652_peakBase 7164 +#define HDSP_9652_peakBase 7164 #define HDSP_9652_rmsBase 4096 /* c.f. the hdsp_9632_meters_t struct */ @@ -155,7 +155,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11 #define HDSP_9652_ENABLE_MIXER 0x800 #define HDSP_TDO 0x10000000 -#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0) +#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0) #define HDSP_S_LOAD (HDSP_PROGRAM|HDSP_CONFIG_MODE_1) /* Control Register bits */ @@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11 #define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */ #define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */ #define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */ -#define HDSP_SyncRef2 (1<<13) -#define HDSP_SPDIFInputSelect0 (1<<14) -#define HDSP_SPDIFInputSelect1 (1<<15) -#define HDSP_SyncRef0 (1<<16) +#define HDSP_SyncRef2 (1<<13) +#define HDSP_SPDIFInputSelect0 (1<<14) +#define HDSP_SPDIFInputSelect1 (1<<15) +#define HDSP_SyncRef0 (1<<16) #define HDSP_SyncRef1 (1<<17) -#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ +#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ #define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */ #define HDSP_Midi0InterruptEnable (1<<22) #define HDSP_Midi1InterruptEnable (1<<23) @@ -189,7 +189,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11 #define HDSP_DAGain1 (1<<28) #define HDSP_PhoneGain0 (1<<29) #define HDSP_PhoneGain1 (1<<30) -#define HDSP_QuadSpeed (1<<31) +#define HDSP_QuadSpeed (1<<31) #define HDSP_ADGainMask (HDSP_ADGain0|HDSP_ADGain1) #define HDSP_ADGainMinus10dBV HDSP_ADGainMask @@ -312,9 +312,9 @@ MODULE_FIRMWARE("digiface_firmware_rev11 #define HDSP_SPDIFErrorFlag (1<<25) #define HDSP_BufferID (1<<26) #define HDSP_TimecodeSync (1<<27) -#define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ +#define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ #define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */ -#define HDSP_midi0IRQPending (1<<30) +#define HDSP_midi0IRQPending (1<<30) #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) @@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11 #define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the Multiface still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -441,7 +441,7 @@ struct hdsp { spinlock_t lock; struct snd_pcm_substream *capture_substream; struct snd_pcm_substream *playback_substream; - struct hdsp_midi midi[2]; + struct hdsp_midi midi[2]; struct tasklet_struct midi_tasklet; int use_midi_tasklet; int precise_ptr; @@ -452,15 +452,15 @@ struct hdsp { int clock_source_locked; char *card_name; /* digiface/multiface */ enum HDSP_IO_Type io_type; /* ditto, but for code use */ - unsigned short firmware_rev; + unsigned short firmware_rev; unsigned short state; /* stores state bits */ u32 firmware_cache[24413]; /* this helps recover from accidental iobox power failure */ - size_t period_bytes; /* guess what this is */ + size_t period_bytes; /* guess what this is */ unsigned char max_channels; unsigned char qs_in_channels; /* quad speed mode for H9632 */ unsigned char ds_in_channels; unsigned char ss_in_channels; /* different for multiface/digiface */ - unsigned char qs_out_channels; + unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; @@ -477,13 +477,13 @@ struct hdsp { int dev; int irq; unsigned long port; - void __iomem *iobase; + void __iomem *iobase; struct snd_card *card; struct snd_pcm *pcm; struct snd_hwdep *hwdep; struct pci_dev *pci; struct snd_kcontrol *spdif_ctl; - unsigned short mixer_matrix[HDSP_MATRIX_MIXER_SIZE]; + unsigned short mixer_matrix[HDSP_MATRIX_MIXER_SIZE]; unsigned int dds_value; /* last value written to freq register */ }; @@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_C static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ /* Analog */ - 0, 1, 2, 3, 4, 5, 6, 7, + 0, 1, 2, 3, 4, 5, 6, 7, /* ADAT 2 */ - 16, 17, 18, 19, 20, 21, 22, 23, + 16, 17, 18, 19, 20, 21, 22, 23, /* SPDIF */ 24, 25, -1, -1, -1, -1, -1, -1, -1, -1 @@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MA /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ - -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; @@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MA /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ @@ -587,7 +587,7 @@ static struct pci_device_id snd_hdsp_ids static struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, - .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, + .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, .subvendor = PCI_ANY_ID, .subdevice = PCI_ANY_ID, }, /* RME Hammerfall-DSP */ @@ -670,19 +670,19 @@ static int snd_hdsp_load_firmware_from_c unsigned long flags; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + snd_printk ("Hammerfall-DSP: loading firmware\n"); hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); return -EIO; } - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - + for (i = 0; i < 24413; ++i) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { @@ -692,10 +692,10 @@ static int snd_hdsp_load_firmware_from_c } ssleep(3); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); - return -EIO; + return -EIO; } #ifdef SNDRV_BIG_ENDIAN @@ -705,15 +705,15 @@ static int snd_hdsp_load_firmware_from_c #endif hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); - + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); spin_lock_irqsave(&hdsp->lock, flags); snd_hdsp_set_defaults(hdsp); - spin_unlock_irqrestore(&hdsp->lock, flags); - } - + spin_unlock_irqrestore(&hdsp->lock, flags); + } + hdsp->state |= HDSP_FirmwareLoaded; return 0; @@ -722,7 +722,7 @@ static int hdsp_get_iobox_version (struc static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) @@ -738,7 +738,7 @@ static int hdsp_get_iobox_version (struc hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); } else { hdsp->io_type = Digiface; - } + } } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) @@ -786,13 +786,13 @@ static int hdsp_check_for_firmware (stru static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout) -{ +{ int i; /* the fifoStatus registers reports on how many words are available in the command FIFO. */ - + for (i = 0; i < timeout; i++) { if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count) @@ -824,11 +824,11 @@ static int hdsp_write_gain(struct hdsp * if (addr >= HDSP_MATRIX_MIXER_SIZE) return -1; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) { /* from martin bjornsen: - + "You can only write dwords to the mixer memory which contain two mixer values in the low and high @@ -847,7 +847,7 @@ static int hdsp_write_gain(struct hdsp * hdsp->mixer_matrix[addr] = data; - + /* `addr' addresses a 16-bit wide address, but the address space accessed via hdsp_write uses byte offsets. put another way, addr @@ -856,17 +856,17 @@ static int hdsp_write_gain(struct hdsp * to access 0 to 2703 ... */ ad = addr/2; - - hdsp_write (hdsp, 4096 + (ad*4), - (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + + + hdsp_write (hdsp, 4096 + (ad*4), + (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + hdsp->mixer_matrix[addr&0x7fe]); - + return 0; } else { ad = (addr << 16) + data; - + if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT)) return -1; @@ -902,7 +902,7 @@ static int hdsp_spdif_sample_rate(struct if (status & HDSP_SPDIFErrorFlag) return 0; - + switch (rate_bits) { case HDSP_spdifFrequency32KHz: return 32000; case HDSP_spdifFrequency44_1KHz: return 44100; @@ -910,13 +910,13 @@ static int hdsp_spdif_sample_rate(struct case HDSP_spdifFrequency64KHz: return 64000; case HDSP_spdifFrequency88_2KHz: return 88200; case HDSP_spdifFrequency96KHz: return 96000; - case HDSP_spdifFrequency128KHz: + case HDSP_spdifFrequency128KHz: if (hdsp->io_type == H9632) return 128000; break; - case HDSP_spdifFrequency176_4KHz: + case HDSP_spdifFrequency176_4KHz: if (hdsp->io_type == H9632) return 176400; break; - case HDSP_spdifFrequency192KHz: + case HDSP_spdifFrequency192KHz: if (hdsp->io_type == H9632) return 192000; break; default: @@ -1027,7 +1027,7 @@ static void hdsp_set_dds_value(struct hd { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -1053,35 +1053,35 @@ static int hdsp_set_rate(struct hdsp *hd there is no need for it (e.g. during module initialization). */ - - if (!(hdsp->control_register & HDSP_ClockModeMaster)) { + + if (!(hdsp->control_register & HDSP_ClockModeMaster)) { if (called_internally) { /* request from ctl or card initialization */ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n"); return -1; - } else { + } else { /* hw_param request while in AutoSync mode */ int external_freq = hdsp_external_sample_rate(hdsp); int spdif_freq = hdsp_spdif_sample_rate(hdsp); - + if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n"); else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) - snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); else if (rate != external_freq) { snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n"); return -1; - } - } + } + } } current_rate = hdsp->system_sample_rate; /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -1089,7 +1089,7 @@ static int hdsp_set_rate(struct hdsp *hd if (rate > 96000 && hdsp->io_type != H9632) return -EINVAL; - + switch (rate) { case 32000: if (current_rate > 48000) @@ -1179,7 +1179,7 @@ static int hdsp_set_rate(struct hdsp *hd break; } } - + hdsp->system_sample_rate = rate; return 0; @@ -1245,16 +1245,16 @@ static int snd_hdsp_midi_output_write (s unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output) { if (!snd_rawmidi_transmit_empty (hmidi->output)) { if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]); } } @@ -1325,14 +1325,14 @@ static void snd_hdsp_midi_output_timer(u { struct hdsp_midi *hmidi = (struct hdsp_midi *) data; unsigned long flags; - + snd_hdsp_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1501,7 +1501,7 @@ static int snd_hdsp_control_spdif_get(st static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif); return 0; } @@ -1511,7 +1511,7 @@ static int snd_hdsp_control_spdif_put(st struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif; @@ -1530,7 +1530,7 @@ static int snd_hdsp_control_spdif_stream static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream); return 0; } @@ -1540,7 +1540,7 @@ static int snd_hdsp_control_spdif_stream struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif_stream; @@ -1602,7 +1602,7 @@ static int snd_hdsp_get_spdif_in(struct static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp); return 0; } @@ -1612,7 +1612,7 @@ static int snd_hdsp_put_spdif_in(struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3); @@ -1649,7 +1649,7 @@ static int snd_hdsp_get_spdif_out(struct static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp); return 0; } @@ -1659,7 +1659,7 @@ static int snd_hdsp_put_spdif_out(struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1693,7 +1693,7 @@ static int snd_hdsp_get_spdif_profession static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp); return 0; } @@ -1703,7 +1703,7 @@ static int snd_hdsp_put_spdif_profession struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1737,7 +1737,7 @@ static int snd_hdsp_get_spdif_emphasis(s static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp); return 0; } @@ -1747,7 +1747,7 @@ static int snd_hdsp_put_spdif_emphasis(s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1781,7 +1781,7 @@ static int snd_hdsp_get_spdif_nonaudio(s static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp); return 0; } @@ -1791,7 +1791,7 @@ static int snd_hdsp_put_spdif_nonaudio(s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1828,7 +1828,7 @@ static int snd_hdsp_get_spdif_sample_rat static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_spdif_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1858,7 +1858,7 @@ static int snd_hdsp_get_spdif_sample_rat ucontrol->value.enumerated.item[0] = 9; break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1882,7 +1882,7 @@ static int snd_hdsp_get_system_sample_ra static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate; return 0; } @@ -1899,7 +1899,7 @@ static int snd_hdsp_info_autosync_sample static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; @@ -1912,7 +1912,7 @@ static int snd_hdsp_get_autosync_sample_ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_external_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1940,9 +1940,9 @@ static int snd_hdsp_get_autosync_sample_ break; case 192000: ucontrol->value.enumerated.item[0] = 9; - break; + break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1968,7 +1968,7 @@ static int snd_hdsp_info_system_clock_mo static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Master", "Slave" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 2; @@ -1981,7 +1981,7 @@ static int snd_hdsp_get_system_clock_mod static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp); return 0; } @@ -2018,7 +2018,7 @@ static int hdsp_clock_source(struct hdsp case 192000: return 9; default: - return 3; + return 3; } } else { return 0; @@ -2032,7 +2032,7 @@ static int hdsp_set_clock_source(struct case HDSP_CLOCK_SOURCE_AUTOSYNC: if (hdsp_external_sample_rate(hdsp) != 0) { if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) { - hdsp->control_register &= ~HDSP_ClockModeMaster; + hdsp->control_register &= ~HDSP_ClockModeMaster; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); return 0; } @@ -2043,7 +2043,7 @@ static int hdsp_set_clock_source(struct break; case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ: rate = 44100; - break; + break; case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ: rate = 48000; break; @@ -2078,13 +2078,13 @@ static int snd_hdsp_info_clock_source(st { static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; if (hdsp->io_type == H9632) uinfo->value.enumerated.items = 10; else - uinfo->value.enumerated.items = 7; + uinfo->value.enumerated.items = 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2094,7 +2094,7 @@ static int snd_hdsp_get_clock_source(str static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp); return 0; } @@ -2104,7 +2104,7 @@ static int snd_hdsp_put_clock_source(str struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2130,7 +2130,7 @@ static int snd_hdsp_get_clock_source_loc static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; return 0; } @@ -2165,7 +2165,7 @@ static int hdsp_da_gain(struct hdsp *hds case HDSP_DAGainMinus10dBV: return 2; default: - return 1; + return 1; } } @@ -2180,8 +2180,8 @@ static int hdsp_set_da_gain(struct hdsp hdsp->control_register |= HDSP_DAGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_DAGainMinus10dBV; - break; + hdsp->control_register |= HDSP_DAGainMinus10dBV; + break; default: return -1; @@ -2193,7 +2193,7 @@ static int snd_hdsp_info_da_gain(struct static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2206,7 +2206,7 @@ static int snd_hdsp_get_da_gain(struct s static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp); return 0; } @@ -2216,7 +2216,7 @@ static int snd_hdsp_put_da_gain(struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2250,7 +2250,7 @@ static int hdsp_ad_gain(struct hdsp *hds case HDSP_ADGainLowGain: return 2; default: - return 1; + return 1; } } @@ -2262,11 +2262,11 @@ static int hdsp_set_ad_gain(struct hdsp hdsp->control_register |= HDSP_ADGainMinus10dBV; break; case 1: - hdsp->control_register |= HDSP_ADGainPlus4dBu; + hdsp->control_register |= HDSP_ADGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_ADGainLowGain; - break; + hdsp->control_register |= HDSP_ADGainLowGain; + break; default: return -1; @@ -2278,7 +2278,7 @@ static int snd_hdsp_info_ad_gain(struct static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2291,7 +2291,7 @@ static int snd_hdsp_get_ad_gain(struct s static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp); return 0; } @@ -2301,7 +2301,7 @@ static int snd_hdsp_put_ad_gain(struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2335,7 +2335,7 @@ static int hdsp_phone_gain(struct hdsp * case HDSP_PhoneGainMinus12dB: return 2; default: - return 0; + return 0; } } @@ -2347,11 +2347,11 @@ static int hdsp_set_phone_gain(struct hd hdsp->control_register |= HDSP_PhoneGain0dB; break; case 1: - hdsp->control_register |= HDSP_PhoneGainMinus6dB; + hdsp->control_register |= HDSP_PhoneGainMinus6dB; break; case 2: - hdsp->control_register |= HDSP_PhoneGainMinus12dB; - break; + hdsp->control_register |= HDSP_PhoneGainMinus12dB; + break; default: return -1; @@ -2363,7 +2363,7 @@ static int snd_hdsp_info_phone_gain(stru static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2376,7 +2376,7 @@ static int snd_hdsp_get_phone_gain(struc static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp); return 0; } @@ -2386,7 +2386,7 @@ static int snd_hdsp_put_phone_gain(struc struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2432,7 +2432,7 @@ static int snd_hdsp_get_xlr_breakout_cab static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp); return 0; } @@ -2442,7 +2442,7 @@ static int snd_hdsp_put_xlr_breakout_cab struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2488,7 +2488,7 @@ static int snd_hdsp_get_aeb(struct snd_k static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp); return 0; } @@ -2498,7 +2498,7 @@ static int snd_hdsp_put_aeb(struct snd_k struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2576,7 +2576,7 @@ static int snd_hdsp_info_pref_sync_ref(s { static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2595,7 +2595,7 @@ static int snd_hdsp_info_pref_sync_ref(s uinfo->value.enumerated.items = 0; break; } - + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2605,7 +2605,7 @@ static int snd_hdsp_get_pref_sync_ref(st static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp); return 0; } @@ -2615,7 +2615,7 @@ static int snd_hdsp_put_pref_sync_ref(st struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; @@ -2664,7 +2664,7 @@ static int hdsp_autosync_ref(struct hdsp case HDSP_SelSyncRef_SPDIF: return HDSP_AUTOSYNC_FROM_SPDIF; case HDSP_SelSyncRefMask: - return HDSP_AUTOSYNC_FROM_NONE; + return HDSP_AUTOSYNC_FROM_NONE; case HDSP_SelSyncRef_ADAT1: return HDSP_AUTOSYNC_FROM_ADAT1; case HDSP_SelSyncRef_ADAT2: @@ -2680,7 +2680,7 @@ static int snd_hdsp_info_autosync_ref(st static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 7; @@ -2693,7 +2693,7 @@ static int snd_hdsp_get_autosync_ref(str static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp); return 0; } @@ -2727,7 +2727,7 @@ static int snd_hdsp_get_line_out(struct static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp_line_out(hdsp); spin_unlock_irq(&hdsp->lock); @@ -2739,7 +2739,7 @@ static int snd_hdsp_put_line_out(struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2773,7 +2773,7 @@ static int snd_hdsp_get_precise_pointer( static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->precise_ptr; spin_unlock_irq(&hdsp->lock); @@ -2785,7 +2785,7 @@ static int snd_hdsp_put_precise_pointer( struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2819,7 +2819,7 @@ static int snd_hdsp_get_use_midi_tasklet static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; spin_unlock_irq(&hdsp->lock); @@ -2831,7 +2831,7 @@ static int snd_hdsp_put_use_midi_tasklet struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2873,12 +2873,12 @@ static int snd_hdsp_get_mixer(struct snd source = ucontrol->value.integer.value[0]; destination = ucontrol->value.integer.value[1]; - + if (source >= hdsp->max_channels) addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination); else addr = hdsp_input_to_output_key(hdsp,source, destination); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr); spin_unlock_irq(&hdsp->lock); @@ -2926,7 +2926,7 @@ static int snd_hdsp_put_mixer(struct snd static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; + static char *texts[] = {"No Lock", "Lock", "Sync" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2971,7 +2971,7 @@ static int hdsp_spdif_sync_check(struct int status = hdsp_read(hdsp, HDSP_statusRegister); if (status & HDSP_SPDIFErrorFlag) return 0; - else { + else { if (status & HDSP_SPDIFSync) return 2; else @@ -3007,7 +3007,7 @@ static int hdsp_adatsync_sync_check(stru return 1; } else return 0; -} +} static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3025,17 +3025,17 @@ static int snd_hdsp_get_adatsync_sync_ch } static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx) -{ +{ int status = hdsp_read(hdsp, HDSP_statusRegister); - + if (status & (HDSP_Lock0>>idx)) { if (status & (HDSP_Sync0>>idx)) return 2; else - return 1; + return 1; } else return 0; -} +} static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3053,7 +3053,7 @@ static int snd_hdsp_get_adat_sync_check( break; case Multiface: case H9632: - if (offset >= 1) + if (offset >= 1) return -EINVAL; break; default: @@ -3115,7 +3115,7 @@ static int snd_hdsp_get_dds_offset(struc static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); return 0; } @@ -3125,7 +3125,7 @@ static int snd_hdsp_put_dds_offset(struc struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -3169,8 +3169,8 @@ static struct snd_kcontrol_new snd_hdsp_ .info = snd_hdsp_control_spdif_mask_info, .get = snd_hdsp_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | - IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_PROFESSIONAL | + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -3188,7 +3188,7 @@ HDSP_SPDIF_PROFESSIONAL("IEC958 Professi HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0), HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), -/* 'Sample Clock Source' complies with the alsa control naming scheme */ +/* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3240,7 +3240,7 @@ static int snd_hdsp_create_controls(stru return err; } } - + /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */ if (hdsp->io_type == H9632) { for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) { @@ -3259,7 +3259,7 @@ static int snd_hdsp_create_controls(stru } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3277,7 +3277,7 @@ snd_hdsp_proc_read(struct snd_info_entry if (hdsp_check_for_iobox (hdsp)) { snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n"); return; - } + } if (hdsp_check_for_firmware(hdsp, 0)) { if (hdsp->state & HDSP_FirmwareCached) { @@ -3298,7 +3298,7 @@ snd_hdsp_proc_read(struct snd_info_entry } } } - + status = hdsp_read(hdsp, HDSP_statusRegister); status2 = hdsp_read(hdsp, HDSP_status2Register); @@ -3362,17 +3362,17 @@ snd_hdsp_proc_read(struct snd_info_entry break; case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ: clock_source = "Internal 192 kHz"; - break; + break; default: - clock_source = "Error"; + clock_source = "Error"; } snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source); - + if (hdsp_system_clock_mode(hdsp)) system_clock_mode = "Slave"; else system_clock_mode = "Master"; - + switch (hdsp_pref_sync_ref (hdsp)) { case HDSP_SYNC_FROM_WORD: pref_sync_ref = "Word Clock"; @@ -3397,7 +3397,7 @@ snd_hdsp_proc_read(struct snd_info_entry break; } snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref); - + switch (hdsp_autosync_ref (hdsp)) { case HDSP_AUTOSYNC_FROM_WORD: autosync_ref = "Word Clock"; @@ -3410,7 +3410,7 @@ snd_hdsp_proc_read(struct snd_info_entry break; case HDSP_AUTOSYNC_FROM_NONE: autosync_ref = "None"; - break; + break; case HDSP_AUTOSYNC_FROM_ADAT1: autosync_ref = "ADAT1"; break; @@ -3425,14 +3425,14 @@ snd_hdsp_proc_read(struct snd_info_entry break; } snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref); - + snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp)); - + snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); - + snd_iprintf(buffer, "\n"); switch (hdsp_spdif_in(hdsp)) { @@ -3452,7 +3452,7 @@ snd_hdsp_proc_read(struct snd_info_entry snd_iprintf(buffer, "IEC958 input: ???\n"); break; } - + if (hdsp->control_register & HDSP_SPDIFOpticalOut) snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); else @@ -3510,13 +3510,13 @@ snd_hdsp_proc_read(struct snd_info_entry snd_iprintf (buffer, "SPDIF: No Lock\n"); else snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock"); - + x = status2 & HDSP_wc_sync; if (status2 & HDSP_wc_lock) snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock"); else snd_iprintf (buffer, "Word Clock: No Lock\n"); - + x = status & HDSP_TimecodeSync; if (status & HDSP_TimecodeLock) snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock"); @@ -3524,11 +3524,11 @@ snd_hdsp_proc_read(struct snd_info_entry snd_iprintf(buffer, "ADAT Sync: No Lock\n"); snd_iprintf(buffer, "\n"); - + /* Informations about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; - + switch (hdsp_ad_gain(hdsp)) { case 0: tmp = "-10 dBV"; @@ -3554,7 +3554,7 @@ snd_hdsp_proc_read(struct snd_info_entry break; } snd_iprintf(buffer, "DA Gain : %s\n", tmp); - + switch (hdsp_phone_gain(hdsp)) { case 0: tmp = "0 dB"; @@ -3568,8 +3568,8 @@ snd_hdsp_proc_read(struct snd_info_entry } snd_iprintf(buffer, "Phones Gain : %s\n", tmp); - snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); - + snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); + if (hdsp->control_register & HDSP_AnalogExtensionBoard) snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n"); else @@ -3632,18 +3632,18 @@ static int snd_hdsp_set_defaults(struct /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax Master clock mode maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - Enable line out. + Enable line out. */ - hdsp->control_register = HDSP_ClockModeMaster | - HDSP_SPDIFInputCoaxial | - hdsp_encode_latency(7) | + hdsp->control_register = HDSP_ClockModeMaster | + HDSP_SPDIFInputCoaxial | + hdsp_encode_latency(7) | HDSP_LineOut; - + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3661,7 +3661,7 @@ static int snd_hdsp_set_defaults(struct hdsp_compute_period_size(hdsp); /* silence everything */ - + for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i) hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN; @@ -3669,7 +3669,7 @@ static int snd_hdsp_set_defaults(struct if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN)) return -EIO; } - + /* H9632 specific defaults */ if (hdsp->io_type == H9632) { hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB); @@ -3687,12 +3687,12 @@ static void hdsp_midi_tasklet(unsigned l static void hdsp_midi_tasklet(unsigned long arg) { struct hdsp *hdsp = (struct hdsp *)arg; - + if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); if (hdsp->midi[1].pending) snd_hdsp_midi_input_read (&hdsp->midi[1]); -} +} static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) { @@ -3704,7 +3704,7 @@ static irqreturn_t snd_hdsp_interrupt(in unsigned int midi0status; unsigned int midi1status; int schedule = 0; - + status = hdsp_read(hdsp, HDSP_statusRegister); audio = status & HDSP_audioIRQPending; @@ -3718,15 +3718,15 @@ static irqreturn_t snd_hdsp_interrupt(in midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; - + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - + if (hdsp->playback_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); } - + if (midi0 && midi0status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ @@ -3767,11 +3767,11 @@ static char *hdsp_channel_buffer_locatio { int mapped_channel; - snd_assert(channel >= 0 && channel < hdsp->max_channels, return NULL); - + snd_assert(channel >= 0 && channel < hdsp->max_channels, return NULL); + if ((mapped_channel = hdsp->channel_map[channel]) < 0) return NULL; - + if (stream == SNDRV_PCM_STREAM_CAPTURE) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else @@ -3894,7 +3894,7 @@ static int snd_hdsp_hw_params(struct snd /* We're fine. */ spin_unlock_irq(&hdsp->lock); - return 0; + return 0; } else { spin_unlock_irq(&hdsp->lock); @@ -3958,7 +3958,7 @@ static int snd_hdsp_trigger(struct snd_p struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; int running; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; @@ -4052,10 +4052,10 @@ static struct snd_pcm_hardware snd_hdsp_ .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4081,10 +4081,10 @@ static struct snd_pcm_hardware snd_hdsp_ .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4163,7 +4163,7 @@ static int snd_hdsp_hw_rule_in_channels_ .max = hdsp->qs_in_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_in_channels, @@ -4194,7 +4194,7 @@ static int snd_hdsp_hw_rule_out_channels .max = hdsp->qs_out_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_out_channels, @@ -4290,7 +4290,7 @@ static int snd_hdsp_playback_open(struct snd_pcm_set_sync(substream); - runtime->hw = snd_hdsp_playback_subinfo; + runtime->hw = snd_hdsp_playback_subinfo; runtime->dma_area = hdsp->playback_buffer; runtime->dma_bytes = HDSP_DMA_AREA_BYTES; @@ -4311,8 +4311,8 @@ static int snd_hdsp_playback_open(struct if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; - } - + } + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -4529,7 +4529,7 @@ static int hdsp_get_peak(struct hdsp *hd hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_playbackRmsLevel + i * 8)) return -EFAULT; - if (copy_u64_le(&peak_rms->input_rms[i], + if (copy_u64_le(&peak_rms->input_rms[i], hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_inputRmsLevel + i * 8)) return -EFAULT; @@ -4539,7 +4539,7 @@ static int hdsp_get_peak(struct hdsp *hd static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; switch (cmd) { @@ -4564,9 +4564,9 @@ static int snd_hdsp_hwdep_ioctl(struct s struct hdsp_config_info info; unsigned long flags; int i; - + if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); + snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); return -EINVAL; } spin_lock_irqsave(&hdsp->lock, flags); @@ -4594,7 +4594,7 @@ static int snd_hdsp_hwdep_ioctl(struct s info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp); info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); - + } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4605,7 +4605,7 @@ static int snd_hdsp_hwdep_ioctl(struct s } case SNDRV_HDSP_IOCTL_GET_9632_AEB: { struct hdsp_9632_aeb h9632_aeb; - + if (hdsp->io_type != H9632) return -EINVAL; h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS; h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS; @@ -4616,7 +4616,7 @@ static int snd_hdsp_hwdep_ioctl(struct s case SNDRV_HDSP_IOCTL_GET_VERSION: { struct hdsp_version hdsp_version; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; if (hdsp->io_type == Undefined) { if ((err = hdsp_get_iobox_version(hdsp)) < 0) @@ -4625,14 +4625,14 @@ static int snd_hdsp_hwdep_ioctl(struct s hdsp_version.io_type = hdsp->io_type; hdsp_version.firmware_rev = hdsp->firmware_rev; if ((err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version)))) - return -EFAULT; + return -EFAULT; break; } case SNDRV_HDSP_IOCTL_UPLOAD_FIRMWARE: { struct hdsp_firmware __user *firmware; u32 __user *firmware_data; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */ if (hdsp->io_type == Undefined) return -EINVAL; @@ -4645,25 +4645,25 @@ static int snd_hdsp_hwdep_ioctl(struct s if (get_user(firmware_data, &firmware->firmware_data)) return -EFAULT; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0) return -EFAULT; - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; - - snd_hdsp_initialize_channels(hdsp); + + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - + if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n"); return err; @@ -4710,10 +4710,10 @@ static int snd_hdsp_create_hwdep(struct { struct snd_hwdep *hw; int err; - + if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0) return err; - + hdsp->hwdep = hw; hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); @@ -4721,7 +4721,7 @@ static int snd_hdsp_create_hwdep(struct hw->ops.open = snd_hdsp_hwdep_dummy_op; hw->ops.ioctl = snd_hdsp_hwdep_ioctl; hw->ops.release = snd_hdsp_hwdep_dummy_op; - + return 0; } @@ -4747,31 +4747,31 @@ static int snd_hdsp_create_pcm(struct sn static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp) { - hdsp->control2_register |= HDSP_9652_ENABLE_MIXER; + hdsp->control2_register |= HDSP_9652_ENABLE_MIXER; hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); } static int snd_hdsp_enable_io (struct hdsp *hdsp) { int i; - + if (hdsp_fifo_wait (hdsp, 0, 100)) { snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n"); return -EIO; } - + for (i = 0; i < hdsp->max_channels; ++i) { hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1); hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1); } - + return 0; } static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { int status, aebi_channels, aebo_channels; - + switch (hdsp->io_type) { case Digiface: hdsp->card_name = "RME Hammerfall DSP + Digiface"; @@ -4784,7 +4784,7 @@ static void snd_hdsp_initialize_channels hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS; break; - + case H9632: status = hdsp_read(hdsp, HDSP_statusRegister); /* HDSP_AEBx bits are low when AEB are connected */ @@ -4804,9 +4804,9 @@ static void snd_hdsp_initialize_channels hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; - + default: - /* should never get here */ + /* should never get here */ break; } } @@ -4820,12 +4820,12 @@ static int snd_hdsp_create_alsa_devices( static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp) { int err; - + if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n"); return err; } - + if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n"); @@ -4856,19 +4856,19 @@ static int snd_hdsp_create_alsa_devices( snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n"); return err; } - + if (!(hdsp->state & HDSP_InitializationComplete)) { strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - + if ((err = snd_card_register(card)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n"); return err; } hdsp->state |= HDSP_InitializationComplete; } - + return 0; } @@ -4879,7 +4879,7 @@ static int hdsp_request_fw_loader(struct const char *fwfile; const struct firmware *fw; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp->io_type == Undefined) { @@ -4888,7 +4888,7 @@ static int hdsp_request_fw_loader(struct if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; } - + /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { case Multiface: @@ -4922,12 +4922,12 @@ static int hdsp_request_fw_loader(struct memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache)); release_firmware(fw); - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; @@ -4974,14 +4974,14 @@ static int __devinit snd_hdsp_create(str hdsp->max_channels = 26; hdsp->card = card; - + spin_lock_init(&hdsp->lock); tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); - + pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; - + /* From Martin Bjoernsen : "It is important that the card's latency timer register in the PCI configuration space is set to a value much larger @@ -4990,7 +4990,7 @@ static int __devinit snd_hdsp_create(str to its maximum 255 to avoid problems with some computers." */ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF); - + strcpy(card->driver, "H-DSP"); strcpy(card->mixername, "Xilinx FPGA"); @@ -5004,7 +5004,7 @@ static int __devinit snd_hdsp_create(str } else { hdsp->card_name = "RME HDSP 9632"; hdsp->max_channels = 16; - is_9632 = 1; + is_9632 = 1; } if ((err = pci_enable_device(pci)) < 0) @@ -5033,7 +5033,7 @@ static int __devinit snd_hdsp_create(str if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) return err; - + if (!is_9652 && !is_9632) { /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); @@ -5056,35 +5056,35 @@ static int __devinit snd_hdsp_create(str return err; return 0; } else { - snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; - else + else hdsp->io_type = Digiface; } } - + if ((err = snd_hdsp_enable_io(hdsp)) != 0) return err; - + if (is_9652) hdsp->io_type = H9652; - + if (is_9632) hdsp->io_type = H9632; if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; - + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - hdsp->state |= HDSP_FirmwareLoaded; + hdsp->state |= HDSP_FirmwareLoaded; if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0) return err; - return 0; + return 0; } static int snd_hdsp_free(struct hdsp *hdsp) @@ -5100,13 +5100,13 @@ static int snd_hdsp_free(struct hdsp *hd free_irq(hdsp->irq, (void *)hdsp); snd_hdsp_free_buffers(hdsp); - + if (hdsp->iobase) iounmap(hdsp->iobase); if (hdsp->port) pci_release_regions(hdsp->pci); - + pci_disable_device(hdsp->pci); return 0; } @@ -5149,7 +5149,7 @@ static int __devinit snd_hdsp_probe(stru } strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); if ((err = snd_card_register(card)) < 0) {

3 5

[alsa-devel] Patch for alsa-info.sh
by Travis Place 25 Feb '08

25 Feb '08

The below patch should stop alsa-info.sh from running amixer on cards that dont exist (ie, if the user has no card0, but has a card1) I realise its a generally a rare case, and cards are _usually_ consecutive, but we might aswell cover all bases if we can. diff -r 7907cfbab11c alsa-info.sh --- a/alsa-info.sh Mon Feb 18 12:13:23 2008 +0100 +++ b/alsa-info.sh Mon Feb 25 15:16:16 2008 +1000 @@ -1,6 +1,6 @@ #!/bin/bash -SCRIPT_VERSION=0.4.37 +SCRIPT_VERSION=0.4.38 CHANGELOG="http://hg.alsa-project.org/alsa/log/tip/alsa-info.sh" ################################################################################# @@ -98,7 +98,7 @@ withamixer() { echo "!!Amixer output" >> $FILE echo "!!-------------" >> $FILE echo "" >> $FILE - for i in `seq 0 $LAST_CARD`;do + for i in `grep "]: " /proc/asound/cards | awk -F ' ' '{ print $1} '` ; do CARD_NAME=`grep "^ $i" /tmp/alsainfo/alsacards.tmp|awk {'print $2'}` echo "!!-------Mixer controls for card $i $CARD_NAME]" >> $FILE echo "" >>$FILE -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 1

Re: [alsa-devel] ASoC - Output widgets
by Mark Brown 25 Feb '08

25 Feb '08

On Fri, Feb 22, 2008 at 05:39:54PM +0100, Alexandre BOUIN wrote: > Using ASoC layer, I'd like to perform this example behaviour with my widgets > : > - 1 output mixer MIXOUT is connected to 2 outputs OUT1 and OUT2 > - OUT1 and OUT2 can be powered independently > This seems easy, but atm while one is powered, the other is powered too. I may be misunderstanding what you're doing here but I suspect that you need to use snd_soc_dapm_set_endpoint() to mark the outputs that are not in use as disconnected. If you do that then then DAPM will not power up the relevant output path. You can do this at any point at run time - see sound/soc/pxa/tosa.c for an example of this.

2 1

[alsa-devel] [PATCH] hda: STAC927x invalid association value
by Matthew Ranostay 25 Feb '08

25 Feb '08

STAC_DELL_BIOS quirks were setting the association value wrong for port 0x0f, which prevented it from being included in hp_outs[]. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 30701aaf404c pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Feb 20 12:13:29 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 15:01:35 2008 -0700 @@ -3630,7 +3630,7 @@ static int patch_stac927x(struct hda_cod break; case STAC_DELL_BIOS: /* correct the front output jack as a hp out */ - stac92xx_set_config_reg(codec, 0x0f, 0x02270110); + stac92xx_set_config_reg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); /* fallthru */

4 8

Re: [alsa-devel] installation steps for us-122l tascam driver
by Federico Briata 25 Feb '08

25 Feb '08

follow step-by-step instruction to compile everything. I'm using debian lenny, for ubuntu should be the same ------- download patch for us-122l--------- mkdir /usr/src/us-122l cd /usr/src/us-122l wget -c http://cache.gmane.org//gmane/linux/alsa/devel/49344-001.bin -O 0001-Driver-for-the-US-122L-usb-soundcard.patch wget -c http://cache.gmane.org//gmane/linux/alsa/devel/49344-002.bin -O 0001-Midi-in-support-for-the-US-122L-usb-soundcard.patch wget -c http://cache.gmane.org//gmane/linux/alsa/devel/49344-003.bin -O 0001-usb-stream-for-isochrone-usb-i-o.patch wget -c http://marc.info/?l=linux-usb-devel&m=119444869917392&q=raw -O ehci-hcd_complete_iso_urbs_ASAP.patch wget -c http://cache.gmane.org//gmane/linux/alsa/devel/49345-001.bin -O jackd-usb_stream-driver.diff.bz2 -------recompile the kernel--------- apt-get install debhelper kernel-package libncurses5-dev fakeroot cd /usr/src wget http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.23.1.tar.bz2 wget http://www.kernel.org/pub/linux/kernel/projects/rt/older/patch-2.6.23.1-rt1…. this is only if u want lowlatency) tar xjvf linux-2.6.23.1.tar.bz2 cd linux-2.6.23.1 bzcat ../patch-2.6.23.1-rt11.bz2 |patch -p1 (Nb. this is only if u want lowlatency) from patch -p1 < ../us-122l/0001-Driver-for-the-US-122L-usb-soundcard.patch patch -p1 < ../us-122l/0001- Midi-in-support-for-the-US-122L-usb-soundcard.patch patch -p1 < ../us-122l/0001-usb-stream-for-isochrone-usb-i-o.patch patch -p1 < ../us-122l/ehci-hcd_complete_iso_urbs_ASAP.patch make-kpkg clean cp /boot/config-2.6.22-3-amd64 .config make oldconfig - say 'yes' when u found your hardware and 'no' for unnecessary driver or press just enter if u r not sure make menuconfig - add driver for us-122l Device Drivers -> Sound -> Advanced Linux Sound Architecture -> USB devices -> Tascam US-122L USB driver - add tweaks for Alsa low-latency, u can get from this howto http://alsa-project.org/main/index.php/Low_latency_howto fakeroot make-kpkg --append-to-version -lowlatency --revision=1 --initrd kernel_image dpkg -i ../linux-image-2.6.23.1-rt11-lowlatency_1_amd64.deb update-initramfs -c -k 2.6.23.1-rt11-lowlatency ----------recompile jackd 0.103------------ apt-get install dh-buildinfo libasound2-dev libsndfile1-dev doxygen libcap-dev libreadline5- dev libraw1394-dev libfreebob0-dev chrpath cdbs type-handling cd /usr/src wget -c http://ftp.de.debian.org/debian/pool/main/j/jack-audio-connection-kit/jack-… wget -c http://ftp.de.debian.org/debian/pool/main/j/jack-audio-connection-kit/jack-… wget -c http://ftp.de.debian.org/debian/pool/main/j/jack-audio-connection-kit/jack-… cp ../us-122l/jackd-usb_stream-driver.diff.bz2 jack-audio-connection-kit-0.103.0/debian/patches/ dpkg-source -x jack-audio-connection-kit_0.103.0-6.dsc cd jack-audio-connection-kit-0.103.0/ fakeroot dpkg-buildpackage -b -uc dpkg -i ../*jack*.deb ----end Unfortunately when i plug the interface i get some error in kernel log. usbcore: registered new interface driver snd-usb-us122l usb 5-4: USB disconnect, address 5 usb 5-4: new high speed USB device using ehci_hcd and address 6 usb 5-4: configuration #1 chosen from 1 choice snd_us122l_probe:380:ffff810023435800:0 snd-usb-us122l: probe of 5-4:1.0 failed with error -5 snd_us122l_probe:380:ffff81002a9a3000:1 us122l_create_card:302 us122l_start:225 us122l_start:228 us122l_start:230 playback_prep_freqn:58 idle=-18 ul=1518 ps=1536 playback_prep_freqn:58 idle=0 ul=1554 ps=1536 i_capture_start:566: 0 0 0 30 36 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 30 36 36 36 30 36 30 36 30 36 30 36 30 36 playback_prep_freqn:58 idle=-12 ul=1524 ps=1536 playback_prep_freqn:58 idle=-30 ul=1518 ps=1536 stream_start:510 253 0 stream_start:510 254 0 i_capture_start:558 us122l_start error -14 snd-usb-us122l: probe of 5-4:1.1 failed with error -5 $ lsmod Module Size Used by nls_iso8859_1 9600 0 nls_cp437 11392 0 vfat 18176 0 fat 59056 1 vfat mmc_block 18816 0 tifm_sd 16392 0 usb_storage 91424 0 snd_rtctimer 8400 0 snd_usb_us122l 23008 0 snd_usb_lib 22272 1 snd_usb_us122l snd_seq_dummy 8452 2 snd_seq_oss 37568 0 snd_seq_midi 13376 0 snd_seq_midi_event 12416 2 snd_seq_oss,snd_seq_midi snd_seq 59872 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_rawmidi 31136 2 snd_usb_lib,snd_seq_midi snd_seq_device 12948 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq,snd_rawmidi snd_hwdep 15240 1 snd_usb_us122l i915 31744 2 drm 101928 3 i915 binfmt_misc 17676 1 rfcomm 50344 4 l2cap 33024 9 rfcomm ppdev 13704 0 lp 17448 0 ipv6 301576 27 acpi_cpufreq 14224 1 cpufreq_powersave 6272 0 cpufreq_userspace 9604 0 cpufreq_stats 10384 0 cpufreq_ondemand 13328 0 freq_table 9600 3 acpi_cpufreq,cpufreq_stats,cpufreq_ondemand cpufreq_conservative 12424 0 fuse 50608 1 tun 16384 0 usbserial 42704 0 loop 24324 0 wlan_scan_sta 18688 1 ath_rate_sample 17664 1 ath_pci 99760 0 wlan 206488 4 wlan_scan_sta,ath_rate_sample,ath_pci ath_hal 223088 3 ath_rate_sample,ath_pci snd_hda_intel 334756 1 irtty_sir 13312 0 sir_dev 22280 1 irtty_sir joydev 15744 0 snd_pcm_oss 48928 0 pcmcia 47008 0 firmware_class 15744 1 pcmcia snd_mixer_oss 21760 1 snd_pcm_oss tsdev 13440 0 parport_pc 42408 1 snd_pcm 90888 2 snd_hda_intel,snd_pcm_oss parport 45708 3 ppdev,lp,parport_pc iTCO_wdt 17360 0 usbhid 34384 0 tifm_7xx1 12800 0 sdhci 23180 0 irda 202956 2 irtty_sir,sir_dev hid 33408 1 usbhid snd_timer 29704 3 snd_rtctimer,snd_seq,snd_pcm tifm_core 16648 2 tifm_sd,tifm_7xx1 serio_raw 11908 0 yenta_socket 31628 2 rsrc_nonstatic 16000 1 yenta_socket pcmcia_core 47268 3 pcmcia,yenta_socket,rsrc_nonstatic crc_ccitt 6656 1 irda mmc_core 37000 3 mmc_block,tifm_sd,sdhci psmouse 45852 0 battery 18064 0 button 13856 0 container 9856 0 ac 10760 0 video 26004 0 output 8704 1 video hci_usb 21532 2 bluetooth 68228 7 rfcomm,l2cap,hci_usb snd 67000 13 snd_usb_us122l,snd_seq_oss,snd_seq,snd_rawmidi,snd_seq_device,snd_hwdep,snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 13968 1 snd intel_agp 31776 1 snd_page_alloc 15376 2 snd_hda_intel,snd_pcm evdev 15360 6 ext3 137744 3 jbd 67152 1 ext3 mbcache 14336 1 ext3 dm_mirror 28544 0 dm_snapshot 22096 0 dm_mod 66168 9 dm_mirror,dm_snapshot sd_mod 33792 3 generic 10372 0 [permanent] ide_core 146832 2 usb_storage,generic ata_piix 24452 2 ata_generic 13188 0 libata 131344 2 ata_piix,ata_generic scsi_mod 164888 3 usb_storage,sd_mod,libata ehci_hcd 38412 0 tg3 111876 0 uhci_hcd 30112 0 thermal 22304 0 processor 43592 2 acpi_cpufreq,thermal fan 10120 0 Arch: Hp compaq nc4400 (amd64) any idea? plz help!! regards, Federico

2 1

Re: [alsa-devel] [Alsa-user] ESS 1688
by Rene Herman 24 Feb '08

24 Feb '08

On 24-02-08 10:50, Rene Herman wrote: > Not yet, but I confirm the bug. I'll try and see of I can locate > anything. No ETA available. Can you try if applying the attached makes things work as you'd expect with respect to the recording levels? Over here it seems to, although I get the feeling we might as well rename the control "Distortion Levels" as keeping it at 0 is the only sensible thing. Even if this works, it's not a final fix, just to confirm. Given that you posted to alsa-user... if you're not setup to apply this patch (to any semi recent source) and recompile the driver I hope someone can help. With respect to the duplex issue -- there seem to be no docs for 688 and 1688 which is a bit of a problem. Rene. diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 5c26d49..eb35267 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -368,9 +368,11 @@ static int snd_es1688_playback_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); + int tmp; chip->dma_size = size; spin_lock_irqsave(&chip->reg_lock, flags); + tmp = snd_es1688_read(chip, 0xb4); snd_es1688_reset(chip); snd_es1688_set_rate(chip, substream); snd_es1688_write(chip, 0xb8, 4); /* auto init DMA mode */ @@ -403,6 +405,7 @@ static int snd_es1688_playback_prepare(struct snd_pcm_substream *substream) } snd_es1688_write(chip, 0xb1, (snd_es1688_read(chip, 0xb1) & 0x0f) | 0x50); snd_es1688_write(chip, 0xb2, (snd_es1688_read(chip, 0xb2) & 0x0f) | 0x50); + snd_es1688_write(chip, 0xb4, tmp); snd_es1688_dsp_command(chip, ES1688_DSP_CMD_SPKON); spin_unlock_irqrestore(&chip->reg_lock, flags); /* --- */ @@ -429,9 +432,11 @@ static int snd_es1688_capture_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); + int tmp; chip->dma_size = size; spin_lock_irqsave(&chip->reg_lock, flags); + tmp = snd_es1688_read(chip, 0xb4); snd_es1688_reset(chip); snd_es1688_set_rate(chip, substream); snd_es1688_dsp_command(chip, ES1688_DSP_CMD_SPKOFF); @@ -461,6 +466,7 @@ static int snd_es1688_capture_prepare(struct snd_pcm_substream *substream) } snd_es1688_write(chip, 0xb1, (snd_es1688_read(chip, 0xb1) & 0x0f) | 0x50); snd_es1688_write(chip, 0xb2, (snd_es1688_read(chip, 0xb2) & 0x0f) | 0x50); + snd_es1688_write(chip, 0xb4, tmp); spin_unlock_irqrestore(&chip->reg_lock, flags); /* --- */ count = -count;

1 0

[alsa-devel] [PATCH] ALSA: update the Kconfig files to be easier to use
by Andrew Paprocki 24 Feb '08

24 Feb '08

This update to the Kconfig files allows the ability to select/deselect entire submenus, starting with the root "Sound device support" on the main kernel configuration page. Inside sound support, ALSA/OSS can be easily enabled/disabled. Inside ALSA, entire classes of devices can be easily enabled/disabled at the ALSA menu (e.g. ISA, PCI, USB devices) without having to manually disable individual cards within the submenus. Modified menus: main, isa, pci, pcmcia, soc, spi, usb Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 39df17389b57 -r a60a39d8e6fe Kconfig --- a/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,11 +1,9 @@ # sound/Config.in # -menu "Sound" +menuconfig SOUND + tristate "Sound device support" depends on HAS_IOMEM - -config SOUND - tristate "Sound card support" help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. Be sure to have all the information @@ -33,21 +31,22 @@ config SOUND Kernel patches and supporting utilities to do that are in the pcsp package, available at <ftp://ftp.infradead.org/pub/pcsp/>. +if SOUND + source "sound/oss/dmasound/Kconfig" if !M68K -menu "Advanced Linux Sound Architecture" - depends on SOUND!=n - -config SND - tristate "Advanced Linux Sound Architecture" - depends on SOUND - help +menuconfig SND + tristate "Advanced Linux Sound Architecture" + depends on SOUND!=n + help Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture), the new base sound system. For more information, see <http://www.alsa-project.org/> + +if SND source "sound/core/Kconfig" @@ -63,9 +62,7 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" -if SPI source "sound/spi/Kconfig" -endif source "sound/mips/Kconfig" @@ -85,22 +82,21 @@ source "sound/parisc/Kconfig" source "sound/soc/Kconfig" -endmenu +endif # SND -menu "Open Sound System" +menuconfig SOUND_PRIME + tristate "Open Sound System (DEPRECATED)" depends on SOUND!=n - -config SOUND_PRIME - tristate "Open Sound System (DEPRECATED)" - depends on SOUND help Say 'Y' or 'M' to enable Open Sound System drivers. +if SOUND_PRIME + source "sound/oss/Kconfig" -endmenu +endif # SOUND_PRIME -endif +endif # !M68K config AC97_BUS tristate @@ -110,4 +106,4 @@ config AC97_BUS sound although they're sharing the AC97 bus. Concerned drivers should "select" this. -endmenu +endif # SOUND diff -r 39df17389b57 -r a60a39d8e6fe isa/Kconfig --- a/isa/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/isa/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,4 +1,13 @@ # ALSA ISA drivers + +menuconfig SND_ISA + bool "ISA devices" + depends on SND!=n && ISA && ISA_DMA_API + default y + help + Support for sound devices connected via the ISA bus. + +if SND_ISA config SND_AD1848_LIB tristate @@ -21,14 +30,10 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON -menu "ISA devices" - depends on SND!=n && ISA && ISA_DMA_API - config SND_ADLIB tristate "AdLib FM card" - depends on SND - select SND_OPL3_LIB - help + select SND_OPL3_LIB + ---help--- Say Y here to include support for AdLib FM cards. To compile this driver as a module, choose M here: the module @@ -36,12 +41,12 @@ config SND_ADLIB config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" - depends on SND && PNP && ISA - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_PCM - help + depends on PNP + select ISAPNP + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + ---help--- Say Y here to include support for Analog Devices SoundPort AD1816A or compatible sound chips. @@ -50,9 +55,8 @@ config SND_AD1816A config SND_AD1848 tristate "Generic AD1848/CS4248 driver" - depends on SND - select SND_AD1848_LIB - help + select SND_AD1848_LIB + ---help--- Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. @@ -64,12 +68,12 @@ config SND_AD1848 config SND_ALS100 tristate "Avance Logic ALS100/ALS120" - depends on SND && PNP && ISA - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help + depends on PNP + select ISAPNP + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB16_DSP + ---help--- Say Y here to include support for soundcards based on Avance Logic ALS100, ALS110, ALS120 and ALS200 chips. @@ -78,12 +82,12 @@ config SND_ALS100 config SND_AZT2320 tristate "Aztech Systems AZT2320" - depends on SND && PNP && ISA - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_CS4231_LIB - help + depends on PNP + select ISAPNP + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_CS4231_LIB + ---help--- Say Y here to include support for soundcards based on the Aztech Systems AZT2320 chip. @@ -92,10 +96,9 @@ config SND_AZT2320 config SND_CMI8330 tristate "C-Media CMI8330" - depends on SND - select SND_AD1848_LIB - select SND_SB16_DSP - help + select SND_AD1848_LIB + select SND_SB16_DSP + ---help--- Say Y here to include support for soundcards based on the C-Media CMI8330 chip. @@ -104,10 +107,9 @@ config SND_CMI8330 config SND_CS4231 tristate "Generic Cirrus Logic CS4231 driver" - depends on SND - select SND_MPU401_UART - select SND_CS4231_LIB - help + select SND_MPU401_UART + select SND_CS4231_LIB + ---help--- Say Y here to include support for CS4231 chips from Cirrus Logic - Crystal Semiconductors. @@ -116,11 +118,10 @@ config SND_CS4231 config SND_CS4232 tristate "Generic Cirrus Logic CS4232 driver" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_CS4231_LIB - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_CS4231_LIB + ---help--- Say Y here to include support for CS4232 chips from Cirrus Logic - Crystal Semiconductors. @@ -129,11 +130,10 @@ config SND_CS4232 config SND_CS4236 tristate "Generic Cirrus Logic CS4236+ driver" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_CS4231_LIB - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_CS4231_LIB + ---help--- Say Y to include support for CS4235,CS4236,CS4237B,CS4238B, CS4239 chips from Cirrus Logic - Crystal Semiconductors. @@ -142,12 +142,12 @@ config SND_CS4236 config SND_DT019X tristate "Diamond Technologies DT-019X, Avance Logic ALS-007" - depends on SND && PNP && ISA - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help + depends on PNP + select ISAPNP + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB16_DSP + ---help--- Say Y here to include support for soundcards based on the Diamond Technologies DT-019X or Avance Logic ALS-007 chips. @@ -156,11 +156,11 @@ config SND_DT019X config SND_ES968 tristate "Generic ESS ES968 driver" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_MPU401_UART select SND_SB8_DSP - help + ---help--- Say Y here to include support for ESS AudioDrive ES968 chips. To compile this driver as a module, choose M here: the module @@ -168,11 +168,10 @@ config SND_ES968 config SND_ES1688 tristate "Generic ESS ES688/ES1688 driver" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_PCM - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + ---help--- Say Y here to include support for ESS AudioDrive ES688 or ES1688 chips. @@ -181,11 +180,10 @@ config SND_ES1688 config SND_ES18XX tristate "Generic ESS ES18xx driver" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_PCM - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + ---help--- Say Y here to include support for ESS AudioDrive ES18xx chips. To compile this driver as a module, choose M here: the module @@ -193,11 +191,11 @@ config SND_ES18XX config SND_SC6000 tristate "Gallant SC-6000, Audio Excel DSP 16" - depends on SND && HAS_IOPORT - select SND_AD1848_LIB - select SND_OPL3_LIB - select SND_MPU401_UART - help + depends on HAS_IOPORT + select SND_AD1848_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + ---help--- Say Y here to include support for Gallant SC-6000 card and clones: Audio Excel DSP 16 and Zoltrix AV302. @@ -209,11 +207,10 @@ config SND_GUS_SYNTH config SND_GUSCLASSIC tristate "Gravis UltraSound Classic" - depends on SND - select SND_RAWMIDI - select SND_PCM - select SND_GUS_SYNTH - help + select SND_RAWMIDI + select SND_PCM + select SND_GUS_SYNTH + ---help--- Say Y here to include support for Gravis UltraSound Classic soundcards. @@ -222,12 +219,11 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - depends on SND select SND_HWDEP select SND_MPU401_UART select SND_PCM select SND_GUS_SYNTH - help + ---help--- Say Y here to include support for Gravis UltraSound Extreme soundcards. @@ -236,11 +232,10 @@ config SND_GUSEXTREME config SND_GUSMAX tristate "Gravis UltraSound MAX" - depends on SND - select SND_RAWMIDI - select SND_CS4231_LIB - select SND_GUS_SYNTH - help + select SND_RAWMIDI + select SND_CS4231_LIB + select SND_GUS_SYNTH + ---help--- Say Y here to include support for Gravis UltraSound MAX soundcards. @@ -249,11 +244,11 @@ config SND_GUSMAX config SND_INTERWAVE tristate "AMD InterWave, Gravis UltraSound PnP" - depends on SND && PNP && ISA - select SND_RAWMIDI - select SND_CS4231_LIB - select SND_GUS_SYNTH - help + depends on PNP + select SND_RAWMIDI + select SND_CS4231_LIB + select SND_GUS_SYNTH + ---help--- Say Y here to include support for AMD InterWave based soundcards (Gravis UltraSound Plug & Play, STB SoundRage32, MED3210, Dynasonic Pro, Panasonic PCA761AW). @@ -263,11 +258,11 @@ config SND_INTERWAVE config SND_INTERWAVE_STB tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)" - depends on SND && PNP && ISA - select SND_RAWMIDI - select SND_CS4231_LIB - select SND_GUS_SYNTH - help + depends on PNP + select SND_RAWMIDI + select SND_CS4231_LIB + select SND_GUS_SYNTH + ---help--- Say Y here to include support for AMD InterWave based soundcards with a TEA6330T bass and treble regulator (UltraSound 32-Pro). @@ -277,11 +272,10 @@ config SND_INTERWAVE_STB config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_CS4231_LIB - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_CS4231_LIB + ---help--- Say Y here to include support for Yamaha OPL3-SA2 and OPL3-SA3 chips. @@ -290,12 +284,11 @@ config SND_OPL3SA2 config SND_OPTI92X_AD1848 tristate "OPTi 82C92x - AD1848" - depends on SND select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART select SND_AD1848_LIB - help + ---help--- Say Y here to include support for soundcards based on Opti 82C92x or OTI-601 chips and using an AD1848 codec. @@ -304,12 +297,11 @@ config SND_OPTI92X_AD1848 config SND_OPTI92X_CS4231 tristate "OPTi 82C92x - CS4231" - depends on SND select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART select SND_CS4231_LIB - help + ---help--- Say Y here to include support for soundcards based on Opti 82C92x chips and using a CS4231 codec. @@ -318,11 +310,10 @@ config SND_OPTI92X_CS4231 config SND_OPTI93X tristate "OPTi 82C93x" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_PCM - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + ---help--- Say Y here to include support for soundcards based on Opti 82C93x chips. @@ -331,12 +322,11 @@ config SND_OPTI93X config SND_MIRO tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver" - depends on SND select SND_OPL4_LIB select SND_CS4231_LIB select SND_MPU401_UART select SND_PCM - help + ---help--- Say 'Y' or 'M' to include support for Miro miroSOUND PCM1 pro, miroSOUND PCM12 and miroSOUND PCM20 Radio soundcards. @@ -345,11 +335,10 @@ config SND_MIRO config SND_SB8 tristate "Sound Blaster 1.0/2.0/Pro (8-bit)" - depends on SND select SND_OPL3_LIB select SND_RAWMIDI select SND_SB8_DSP - help + ---help--- Say Y here to include support for Creative Sound Blaster 1.0/ 2.0/Pro (8-bit) or 100% compatible soundcards. @@ -358,11 +347,10 @@ config SND_SB8 config SND_SB16 tristate "Sound Blaster 16 (PnP)" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB16_DSP + ---help--- Say Y here to include support for Sound Blaster 16 soundcards (including the Plug and Play version). @@ -371,11 +359,10 @@ config SND_SB16 config SND_SBAWE tristate "Sound Blaster AWE (32,64) (PnP)" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB16_DSP + ---help--- Say Y here to include support for Sound Blaster AWE soundcards (including the Plug and Play version). @@ -386,7 +373,7 @@ config SND_SB16_CSP bool "Sound Blaster 16/AWE CSP support" depends on (SND_SB16 || SND_SBAWE) && (BROKEN || !PPC) select FW_LOADER if !SND_SB16_CSP_FIRMWARE_IN_KERNEL - help + ---help--- Say Y here to include support for the CSP core. This special coprocessor can do variable tasks like various compression and decompression algorithms. @@ -395,16 +382,15 @@ config SND_SB16_CSP_FIRMWARE_IN_KERNEL bool "In-kernel firmware for SB16 CSP" depends on SND_SB16_CSP default y - help + ---help--- Say Y here to include the static firmware built in the kernel for the SB16 CSP controller. If you choose N here, you need to install the firmware files from the alsa-firmware package. config SND_SGALAXY tristate "Aztech Sound Galaxy" - depends on SND - select SND_AD1848_LIB - help + select SND_AD1848_LIB + ---help--- Say Y here to include support for Aztech Sound Galaxy soundcards. @@ -413,11 +399,10 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape PnP driver" - depends on SND select SND_HWDEP select SND_MPU401_UART select SND_CS4231_LIB - help + ---help--- Say Y here to include support for Ensoniq SoundScape PnP soundcards. @@ -426,12 +411,11 @@ config SND_SSCAPE config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" - depends on SND select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB - help + ---help--- Say Y here to include support for Turtle Beach Maui, Tropez and Tropez+ soundcards based on the Wavefront chip. @@ -442,10 +426,10 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL bool "In-kernel firmware for Wavefront" depends on SND_WAVEFRONT default y - help + ---help--- Say Y here to include the static firmware for FX DSP built in the kernel for the Wavefront driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. -endmenu +endif # SND_ISA diff -r 39df17389b57 -r a60a39d8e6fe pci/Kconfig --- a/pci/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/pci/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,13 +1,18 @@ # ALSA PCI drivers -menu "PCI devices" - depends on SND!=n && PCI +menuconfig SND_PCI + bool "PCI devices" + depends on SND!=n && PCI + default y + help + Support for sound devices connected via the PCI bus. + +if SND_PCI config SND_AD1889 tristate "Analog Devices AD1889" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated AC97 sound device found in particular on the Hewlett-Packard [BCJ]-xxx0 class PA-RISC workstations, using the AD1819 codec. @@ -17,11 +22,10 @@ config SND_AD1889 config SND_ALS300 tristate "Avance Logic ALS300/ALS300+" - depends on SND - select SND_PCM - select SND_AC97_CODEC - select SND_OPL3_LIB - help + select SND_PCM + select SND_AC97_CODEC + select SND_OPL3_LIB + ---help--- Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ To compile this driver as a module, choose M here: the module @@ -29,12 +33,12 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM select SND_SB_COMMON - help + ---help--- Say Y here to include support for soundcards based on Avance Logic ALS4000 chips. @@ -43,10 +47,9 @@ config SND_ALS4000 config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller (M1535/M1535D/M1535+/M1535D+ south bridges). Newer chipsets @@ -57,9 +60,8 @@ config SND_ALI5451 config SND_ATIIXP tristate "ATI IXP AC97 Controller" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated AC97 sound device on motherboards with ATI chipsets (ATI IXP 150/200/250/ 300/400). @@ -69,9 +71,8 @@ config SND_ATIIXP config SND_ATIIXP_MODEM tristate "ATI IXP Modem" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated MC97 modem on motherboards with ATI chipsets (ATI IXP 150/200/250). @@ -80,10 +81,9 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for Aureal Advantage soundcards. Supported features: Hardware Mixer, SRC, EQ and SPDIF output. @@ -95,10 +95,9 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for Aureal Vortex soundcards. Supported features: Hardware Mixer and SRC. For more info, email @@ -109,10 +108,9 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for Aureal Vortex 2 soundcards. Supported features: Hardware Mixer, SRC, EQ and SPDIF output. @@ -124,8 +122,7 @@ config SND_AU8830 config SND_AW2 tristate "Emagic Audiowerk 2" - depends on SND - help + ---help--- Say Y here to include support for Emagic Audiowerk 2 soundcards. Supported features: Analog and SPDIF output. Analog or SPDIF input. @@ -139,11 +136,11 @@ config SND_AW2 config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" - depends on SND && EXPERIMENTAL - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_PCM - help + depends on EXPERIMENTAL + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + ---help--- Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -152,9 +149,8 @@ config SND_AZT3328 config SND_BT87X tristate "Bt87x Audio Capture" - depends on SND - select SND_PCM - help + select SND_PCM + ---help--- If you want to record audio from TV cards based on Brooktree Bt878/Bt879 chips, say Y here and read <file:Documentation/sound/alsa/Bt87x.txt>. @@ -165,7 +161,7 @@ config SND_BT87X_OVERCLOCK config SND_BT87X_OVERCLOCK bool "Bt87x Audio overclocking" depends on SND_BT87X - help + ---help--- Say Y here if 448000 Hz isn't enough for you and you want to record from the analog input with up to 1792000 Hz. @@ -174,11 +170,10 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER - help + ---help--- Say Y here to include support for the Sound Blaster Audigy LS and Live 24bit. @@ -187,11 +182,10 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_PCM - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + ---help--- If you want to use soundcards based on C-Media CMI8338, CMI8738, CMI8768 or CMI8770 chips, say Y here and read <file:Documentation/sound/alsa/CMIPCI.txt>. @@ -201,15 +195,13 @@ config SND_CMIPCI config SND_OXYGEN_LIB tristate - depends on SND select SND_PCM select SND_MPU401_UART config SND_OXYGEN tristate "C-Media 8788 (Oxygen)" - depends on SND select SND_OXYGEN_LIB - help + ---help--- Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: * Asound A-8788 @@ -225,11 +217,10 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" - depends on SND - select SND_OPL3_LIB - select SND_RAWMIDI - select SND_AC97_CODEC - help + select SND_OPL3_LIB + select SND_RAWMIDI + select SND_AC97_CODEC + ---help--- Say Y here to include support for Cirrus Logic CS4281 chips. To compile this driver as a module, choose M here: the module @@ -237,10 +228,9 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" - depends on SND - select SND_RAWMIDI - select SND_AC97_CODEC - help + select SND_RAWMIDI + select SND_AC97_CODEC + ---help--- Say Y here to include support for Cirrus Logic CS4610/CS4612/ CS4614/CS4615/CS4622/CS4624/CS4630/CS4280 chips. @@ -251,16 +241,16 @@ config SND_CS46XX_NEW_DSP bool "Cirrus Logic (Sound Fusion) New DSP support" depends on SND_CS46XX default y - help + ---help--- Say Y here to use a new DSP image for SPDIF and dual codecs. This works better than the old code, so say Y. config SND_CS5530 tristate "CS5530 Audio" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_SB16_DSP - help + ---help--- Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. To compile this driver as a module, choose M here: the module @@ -268,10 +258,10 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on SND && X86 && !X86_64 - select SND_PCM - select SND_AC97_CODEC - help + depends on X86 && !X86_64 + select SND_PCM + select SND_AC97_CODEC + ---help--- Say Y here to include support for audio on CS5535 chips. It is referred to as NS CS5535 IO or AMD CS5535 IO companion in various literature. This driver also supports the CS5536 audio @@ -286,10 +276,9 @@ config SND_CS5535AUDIO config SND_DARLA20 tristate "(Echoaudio) Darla20" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Darla. To compile this driver as a module, choose M here: the module @@ -297,10 +286,9 @@ config SND_DARLA20 config SND_GINA20 tristate "(Echoaudio) Gina20" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Gina. To compile this driver as a module, choose M here: the module @@ -308,11 +296,10 @@ config SND_GINA20 config SND_LAYLA20 tristate "(Echoaudio) Layla20" - depends on SND - select FW_LOADER - select SND_RAWMIDI - select SND_PCM - help + select FW_LOADER + select SND_RAWMIDI + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Layla. To compile this driver as a module, choose M here: the module @@ -320,10 +307,9 @@ config SND_LAYLA20 config SND_DARLA24 tristate "(Echoaudio) Darla24" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Darla24. To compile this driver as a module, choose M here: the module @@ -331,10 +317,9 @@ config SND_DARLA24 config SND_GINA24 tristate "(Echoaudio) Gina24" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Gina24. To compile this driver as a module, choose M here: the module @@ -342,11 +327,10 @@ config SND_GINA24 config SND_LAYLA24 tristate "(Echoaudio) Layla24" - depends on SND - select FW_LOADER - select SND_RAWMIDI - select SND_PCM - help + select FW_LOADER + select SND_RAWMIDI + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Layla24. To compile this driver as a module, choose M here: the module @@ -354,11 +338,10 @@ config SND_LAYLA24 config SND_MONA tristate "(Echoaudio) Mona" - depends on SND - select FW_LOADER - select SND_RAWMIDI - select SND_PCM - help + select FW_LOADER + select SND_RAWMIDI + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Mona. To compile this driver as a module, choose M here: the module @@ -366,11 +349,10 @@ config SND_MONA config SND_MIA tristate "(Echoaudio) Mia" - depends on SND - select FW_LOADER - select SND_RAWMIDI - select SND_PCM - help + select FW_LOADER + select SND_RAWMIDI + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Mia and Mia-midi. To compile this driver as a module, choose M here: the module @@ -378,11 +360,10 @@ config SND_MIA config SND_ECHO3G tristate "(Echoaudio) 3G cards" - depends on SND - select FW_LOADER - select SND_RAWMIDI - select SND_PCM - help + select FW_LOADER + select SND_RAWMIDI + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Gina3G and Layla3G. To compile this driver as a module, choose M here: the module @@ -390,10 +371,9 @@ config SND_ECHO3G config SND_INDIGO tristate "(Echoaudio) Indigo" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Indigo. To compile this driver as a module, choose M here: the module @@ -401,10 +381,9 @@ config SND_INDIGO config SND_INDIGOIO tristate "(Echoaudio) Indigo IO" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Indigo IO. To compile this driver as a module, choose M here: the module @@ -412,10 +391,9 @@ config SND_INDIGOIO config SND_INDIGODJ tristate "(Echoaudio) Indigo DJ" - depends on SND - select FW_LOADER - select SND_PCM - help + select FW_LOADER + select SND_PCM + ---help--- Say 'Y' or 'M' to include support for Echoaudio Indigo DJ. To compile this driver as a module, choose M here: the module @@ -423,12 +401,11 @@ config SND_INDIGODJ config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" - depends on SND select FW_LOADER select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC - help + ---help--- Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -441,10 +418,9 @@ config SND_EMU10K1 config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" - depends on SND - select SND_AC97_CODEC - select SND_RAWMIDI - help + select SND_AC97_CODEC + select SND_RAWMIDI + ---help--- Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -453,10 +429,9 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" - depends on SND - select SND_RAWMIDI - select SND_PCM - help + select SND_RAWMIDI + select SND_PCM + ---help--- Say Y here to include support for Ensoniq AudioPCI ES1370 chips. To compile this driver as a module, choose M here: the module @@ -464,10 +439,9 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" - depends on SND - select SND_RAWMIDI - select SND_AC97_CODEC - help + select SND_RAWMIDI + select SND_AC97_CODEC + ---help--- Say Y here to include support for Ensoniq AudioPCI ES1371 chips and Sound Blaster PCI 64 or 128 soundcards. @@ -476,11 +450,10 @@ config SND_ENS1371 config SND_ES1938 tristate "ESS ES1938/1946/1969 (Solo-1)" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -489,10 +462,9 @@ config SND_ES1938 config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -501,11 +473,10 @@ config SND_ES1968 config SND_FM801 tristate "ForteMedia FM801" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on the ForteMedia FM801 chip. @@ -515,7 +486,7 @@ config SND_FM801_TEA575X_BOOL config SND_FM801_TEA575X_BOOL bool "ForteMedia FM801 + TEA5757 tuner" depends on SND_FM801 - help + ---help--- Say Y here to include support for soundcards based on the ForteMedia FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media Forte SF256-PCS-02) into the snd-fm801 driver. @@ -531,10 +502,9 @@ config SND_FM801_TEA575X config SND_HDA_INTEL tristate "Intel HD Audio" - depends on SND select SND_PCM select SND_VMASTER - help + ---help--- Say Y here to include support for Intel "High Definition Audio" (Azalia) motherboard devices. @@ -545,7 +515,7 @@ config SND_HDA_HWDEP bool "Build hwdep interface for HD-audio driver" depends on SND_HDA_INTEL select SND_HWDEP - help + ---help--- Say Y here to build a hwdep interface for HD-audio driver. This interface can be used for out-of-band communication with codecs for debugging purposes. @@ -554,7 +524,7 @@ config SND_HDA_CODEC_REALTEK bool "Build Realtek HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. @@ -562,7 +532,7 @@ config SND_HDA_CODEC_ANALOG bool "Build Analog Device HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include Analog Device HD-audio codec support in snd-hda-intel driver, such as AD1986A. @@ -570,7 +540,7 @@ config SND_HDA_CODEC_SIGMATEL bool "Build IDT/Sigmatel HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. @@ -578,7 +548,7 @@ config SND_HDA_CODEC_VIA bool "Build VIA HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include VIA HD-audio codec support in snd-hda-intel driver, such as VT1708. @@ -586,7 +556,7 @@ config SND_HDA_CODEC_ATIHDMI bool "Build ATI HDMI HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include ATI HDMI HD-audio codec support in snd-hda-intel driver, such as ATI RS600 HDMI. @@ -594,7 +564,7 @@ config SND_HDA_CODEC_CONEXANT bool "Build Conexant HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include Conexant HD-audio codec support in snd-hda-intel driver, such as CX20549. @@ -602,7 +572,7 @@ config SND_HDA_CODEC_CMEDIA bool "Build C-Media HD-audio codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include C-Media HD-audio codec support in snd-hda-intel driver, such as CMI9880. @@ -610,7 +580,7 @@ config SND_HDA_CODEC_SI3054 bool "Build Silicon Labs 3054 HD-modem codec support" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to include Silicon Labs 3054 HD-modem codec (and compatibles) support in snd-hda-intel driver. @@ -618,14 +588,14 @@ config SND_HDA_GENERIC bool "Enable generic HD-audio codec parser" depends on SND_HDA_INTEL default y - help + ---help--- Say Y here to enable the generic HD-audio codec parser in snd-hda-intel driver. config SND_HDA_POWER_SAVE bool "Aggressive power-saving on HD-audio" depends on SND_HDA_INTEL && EXPERIMENTAL - help + ---help--- Say Y here to enable more aggressive power-saving mode on HD-audio driver. The power-saving timeout can be configured via power_save option or over sysfs on-the-fly. @@ -634,17 +604,16 @@ config SND_HDA_POWER_SAVE_DEFAULT int "Default time-out for HD-audio power-save mode" depends on SND_HDA_POWER_SAVE default 0 - help + ---help--- The default time-out value in seconds for HD-audio automatic power-save mode. 0 means to disable the power-save mode. config SND_HDSP tristate "RME Hammerfall DSP Audio" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM - help + ---help--- Say Y here to include support for RME Hammerfall DSP Audio soundcards. @@ -653,11 +622,10 @@ config SND_HDSP config SND_HDSPM tristate "RME Hammerfall DSP MADI" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM - help + ---help--- Say Y here to include support for RME Hammerfall DSP MADI soundcards. @@ -666,9 +634,8 @@ config SND_HDSPM config SND_HIFIER tristate "TempoTec HiFier Fantasia" - depends on SND select SND_OXYGEN_LIB - help + ---help--- Say Y here to include support for the MediaTek/TempoTec HiFier Fantasia sound card. @@ -677,10 +644,9 @@ config SND_HIFIER config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -694,10 +660,9 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on ICE/VT1724/1720 (Envy24HT/PT) chips. @@ -711,9 +676,8 @@ config SND_ICE1724 config SND_INTEL8X0 tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated AC97 sound device on motherboards with Intel/SiS/nVidia/AMD chipsets, or ALi chipsets using the M5455 Audio Controller. (There is a @@ -724,9 +688,8 @@ config SND_INTEL8X0 config SND_INTEL8X0M tristate "Intel/SiS/nVidia/AMD MC97 Modem" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated MC97 modem on motherboards with Intel/SiS/nVidia/AMD chipsets. @@ -735,10 +698,9 @@ config SND_INTEL8X0M config SND_KORG1212 tristate "Korg 1212 IO" - depends on SND select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL select SND_PCM - help + ---help--- Say Y here to include support for Korg 1212IO soundcards. To compile this driver as a module, choose M here: the module @@ -748,17 +710,16 @@ config SND_KORG1212_FIRMWARE_IN_KERNEL bool "In-kernel firmware for Korg1212 driver" depends on SND_KORG1212 default y - help + ---help--- Say Y here to include the static firmware built in the kernel for the Korg1212 driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" - depends on SND select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL select SND_AC97_CODEC - help + ---help--- Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -769,17 +730,16 @@ config SND_MAESTRO3_FIRMWARE_IN_KERNEL bool "In-kernel firmware for Maestro3 driver" depends on SND_MAESTRO3 default y - help + ---help--- Say Y here to include the static firmware built in the kernel for the Maestro3 driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. config SND_MIXART tristate "Digigram miXart" - depends on SND select SND_HWDEP select SND_PCM - help + ---help--- If you want to use Digigram miXart soundcards, say Y here and read <file:Documentation/sound/alsa/MIXART.txt>. @@ -788,9 +748,8 @@ config SND_MIXART config SND_NM256 tristate "NeoMagic NM256AV/ZX" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for NeoMagic NM256AV/ZX chips. To compile this driver as a module, choose M here: the module @@ -798,10 +757,9 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" - depends on SND select SND_PCM select SND_HWDEP - help + ---help--- Say Y here to include support for Digigram PCXHR boards. To compile this driver as a module, choose M here: the module @@ -809,12 +767,11 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" - depends on SND - select FW_LOADER - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_AC97_CODEC - help + select FW_LOADER + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say 'Y' or 'M' to include support for Conexant Riptide chip. To compile this driver as a module, choose M here: the module @@ -822,9 +779,8 @@ config SND_RIPTIDE config SND_RME32 tristate "RME Digi32, 32/8, 32 PRO" - depends on SND - select SND_PCM - help + select SND_PCM + ---help--- Say Y to include support for RME Digi32, Digi32 PRO and Digi32/8 (Sek'd Prodif32, Prodif96 and Prodif Gold) audio devices. @@ -834,9 +790,8 @@ config SND_RME32 config SND_RME96 tristate "RME Digi96, 96/8, 96/8 PRO" - depends on SND - select SND_PCM - help + select SND_PCM + ---help--- Say Y here to include support for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST soundcards. @@ -845,9 +800,8 @@ config SND_RME96 config SND_RME9652 tristate "RME Digi9652 (Hammerfall)" - depends on SND - select SND_PCM - help + select SND_PCM + ---help--- Say Y here to include support for RME Hammerfall (RME Digi9652/Digi9636) soundcards. @@ -856,9 +810,9 @@ config SND_RME9652 config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" - depends on SND && X86 && !X86_64 - select SND_AC97_CODEC - help + depends on X86 && !X86_64 + select SND_AC97_CODEC + ---help--- Say Y here to include support for the SiS 7019 Audio Accelerator. To compile this driver as a module, choose M here: the module @@ -866,11 +820,10 @@ config SND_SIS7019 config SND_SONICVIBES tristate "S3 SonicVibes" - depends on SND - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -879,10 +832,9 @@ config SND_SONICVIBES config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. @@ -891,10 +843,9 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" - depends on SND - select SND_MPU401_UART - select SND_AC97_CODEC - help + select SND_MPU401_UART + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated AC97 sound device on motherboards with VIA chipsets. @@ -903,9 +854,8 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" - depends on SND - select SND_AC97_CODEC - help + select SND_AC97_CODEC + ---help--- Say Y here to include support for the integrated MC97 modem on motherboards with VIA chipsets. @@ -914,9 +864,8 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 200 (Xonar)" - depends on SND select SND_OXYGEN_LIB - help + ---help--- Say Y here to include support for sound cards based on the Asus AV200 chip, i.e., Xonar D2 and Xonar D2X. @@ -925,9 +874,8 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" - depends on SND select SND_VX_LIB - help + ---help--- Say Y here to include support for Digigram VX222 soundcards. To compile this driver as a module, choose M here: the module @@ -935,12 +883,11 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" - depends on SND select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - help + ---help--- Say Y here to include support for Yamaha PCI audio chips - YMF724, YMF724F, YMF740, YMF740C, YMF744, YMF754. @@ -951,7 +898,7 @@ config SND_YMFPCI_FIRMWARE_IN_KERNEL bool "In-kernel firmware for YMFPCI driver" depends on SND_YMFPCI default y - help + ---help--- Say Y here to include the static firmware built in the kernel for the YMFPCI driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. @@ -960,7 +907,7 @@ config SND_AC97_POWER_SAVE bool "AC97 Power-Saving Mode" depends on SND_AC97_CODEC && EXPERIMENTAL default n - help + ---help--- Say Y here to enable the aggressive power-saving support of AC97 codecs. In this mode, the power-mode is dynamically controlled at each open/close. @@ -973,8 +920,8 @@ config SND_AC97_POWER_SAVE_DEFAULT int "Default time-out for AC97 power-save mode" depends on SND_AC97_POWER_SAVE default 0 - help + ---help--- The default time-out value in seconds for AC97 automatic power-save mode. 0 means to disable the power-save mode. -endmenu +endif # SND_PCI diff -r 39df17389b57 -r a60a39d8e6fe pcmcia/Kconfig --- a/pcmcia/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/pcmcia/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,11 +1,16 @@ # ALSA PCMCIA drivers -menu "PCMCIA devices" +menuconfig SND_PCMCIA + bool "PCMCIA devices" depends on SND!=n && PCMCIA + default y + help + Support for sound devices connected via a PCMCIA interface. + +if SND_PCMCIA config SND_VXPOCKET tristate "Digigram VXpocket" - depends on SND && PCMCIA select SND_VX_LIB help Say Y here to include support for Digigram VXpocket and @@ -16,7 +21,6 @@ config SND_VXPOCKET config SND_PDAUDIOCF tristate "Sound Core PDAudioCF" - depends on SND && PCMCIA select SND_PCM help Say Y here to include support for Sound Core PDAudioCF @@ -25,4 +29,4 @@ config SND_PDAUDIOCF To compile this driver as a module, choose M here: the module will be called snd-pdaudiocf. -endmenu +endif # SND_PCMCIA diff -r 39df17389b57 -r a60a39d8e6fe soc/Kconfig --- a/soc/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/soc/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,27 +1,24 @@ -# -# SoC audio configuration -# +# ALSA SoC audio configuration -menu "System on Chip audio support" - depends on SND!=n +menuconfig SND_SOC + tristate "SoC (System on Chip) devices" + depends on SND!=n + select SND_PCM + help + If you want ASoC support, you should say Y here and also to the + specific driver for your SoC platform below. + + ASoC provides power efficient ALSA support for embedded battery + powered SoC based systems like PDAs, phones and personal media + players. + + This ASoC audio support can also be built as a module. If so, the + module will be called snd-soc-core. + +if SND_SOC config SND_SOC_AC97_BUS bool - -config SND_SOC - tristate "ALSA for SoC audio support" - depends on SND - select SND_PCM - ---help--- - - If you want ASoC support, you should say Y here and also to the - specific driver for your SoC platform below. - - ASoC provides power efficient ALSA support for embedded battery powered - SoC based systems like PDA's, Phones and Personal Media Players. - - This ASoC audio support can also be built as a module. If so, the module - will be called snd-soc-core. # All the supported Soc's source "sound/soc/at91/Kconfig" @@ -34,4 +31,4 @@ source "sound/soc/davinci/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" -endmenu +endif # SND_SOC diff -r 39df17389b57 -r a60a39d8e6fe spi/Kconfig --- a/spi/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/spi/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,13 +1,19 @@ -#SPI drivers +# ALSA SPI drivers -menu "SPI devices" - depends on SND != n +menuconfig SND_SPI + bool "SPI devices" + depends on SND!=n && SPI + default y + help + Support for sound devices connected via the SPI bus. + +if SND_SPI config SND_AT73C213 tristate "Atmel AT73C213 DAC driver" depends on ATMEL_SSC select SND_PCM - help + ---help--- Say Y here if you want to use the Atmel AT73C213 external DAC. This DAC can be found on Atmel development boards. @@ -22,10 +28,10 @@ config SND_AT73C213_TARGET_BITRATE depends on SND_AT73C213 default "48000" range 8000 50000 - help + ---help--- Sets the target bitrate for the bitrate calculator in the driver. Limited by hardware to be between 8000 Hz and 50000 Hz. Set to 48000 Hz by default. -endmenu +endif # SND_SPI diff -r 39df17389b57 -r a60a39d8e6fe usb/Kconfig --- a/usb/Kconfig Fri Feb 22 17:57:02 2008 +0100 +++ b/usb/Kconfig Sat Feb 23 01:04:07 2008 -0500 @@ -1,15 +1,20 @@ # ALSA USB drivers -menu "USB devices" - depends on SND!=n && USB!=n +menuconfig SND_USB + bool "USB devices" + depends on SND!=n && USB!=n + default y + help + Support for sound devices connected via the USB bus. + +if SND_USB config SND_USB_AUDIO tristate "USB Audio/MIDI driver" - depends on SND && USB select SND_HWDEP select SND_RAWMIDI select SND_PCM - help + ---help--- Say Y here to include support for USB audio and USB MIDI devices. @@ -18,11 +23,11 @@ config SND_USB_AUDIO config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" - depends on SND && USB && (X86 || PPC || ALPHA) + depends on (X86 || PPC || ALPHA) select SND_HWDEP select SND_RAWMIDI select SND_PCM - help + ---help--- Say Y here to include support for Tascam USB Audio/MIDI interfaces or controllers US-122, US-224 and US-428. @@ -31,11 +36,10 @@ config SND_USB_USX2Y config SND_USB_CAIAQ tristate "Native Instruments USB audio devices" - depends on SND && USB select SND_HWDEP select SND_RAWMIDI select SND_PCM - help + ---help--- Say Y here to include support for caiaq USB audio interfaces, namely: @@ -50,10 +54,10 @@ config SND_USB_CAIAQ will be called snd-usb-caiaq. config SND_USB_CAIAQ_INPUT - bool "enable input device for controllers" + bool "Enable input device for controllers" depends on SND_USB_CAIAQ depends on INPUT=y || INPUT=SND_USB_CAIAQ - help + ---help--- Say Y here to support input controllers like buttons, knobs, alpha dials and analog pedals on the following products: @@ -63,5 +67,4 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 -endmenu - +endif # SND_USB

2 4

[alsa-devel] about PCM devices
by Deephay 24 Feb '08

24 Feb '08

Greetings all, I am very new to ALSA developing, some times I found my program get a "snd_pcm_open with device 'default': Device or resource busy" error if some other applications was using ALSA at the mean time (for example virtualbox), but totem player can co-exist with the ALSA-using application very well, what is the reason? I tried play a wav file when there is another ALSA-using application such as totem : aplay -D default test.wav everything went fine, but with "-D hw:0,0", "-D plughw:0,0", the device will be busy, with "-D hw:0,1" or "-D plughw:0,1", the device won't be busy but you can hear nothing, the documentation explains: "The default device is equal to plug plugin with hw plugin as slave.", so why "-D plughw:0,0" won't work? Thanks very much! Cheers, Deephay

1 0

[alsa-devel] Adapt eeepc p710 mixer for virtual master control
by Herton Ronaldo Krzesinski 23 Feb '08

23 Feb '08

Adapt eeepc p710 mixer for virtual master control Currently eeepc p710 mixer doesn't define any slave volume control. This is incompatible with current virtual master control code, that will return -ENOENT in this case. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f7dd96..bda09ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12989,8 +12989,8 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), I'll explain the issue a bit more: after commit 2134ea4f37d36addbe86d4901f6c67a22a5db006 "[ALSA] hda-codec - Add virtual master controls", alsamixer for example doesn't work anymore on eeepc p710: "alsamixer: function snd_mixer_load failed: No such file or directory" I figured out that the problem was with the virtual master controls code and the eeepc p710 mixer that doesn't have any slave volume control. Given eeepc hardware and my understading in reality this virtual master doesn't seem much useful (you have only one playback source - LineOut Playback), only for master switch it is useful as you have Speaker and HeadPhone switches (Headphone is name LineOut Playback Switch in this case) as slave switches. But as only master switch doesn't seem good (it would be strange change vmaster code to accept only a master switch) and I didn't want to refactor the eeepc mixer, this patch seemed a good solution, please comment on it, what do you think? I also thought of making code to just disable vmaster for this model, I don't know what could be the better solution. Something should be done also for the other eeepc ep20 model mixer, it will not suffer the same bug, but the way its mixer is currently seems to conflict with the virtual master control idea (may be "MuteCtrl" would be obsoleted). -- []'s Herton

2 2

[alsa-devel] emu10k1, how to setup DMA for 24bit playback?
by Ctirad Fertr 23 Feb '08

23 Feb '08

Hello, I'm trying to add 24bit playback for E-mu cards into emu10k driver. These cards use an FPGA, which is 32bit internally and it is feeded via 10k2 EMU32 BUSes. Unfortunatelly, the emu10k DMA cannot handle more than 16bit samples, so kind of tricky stuff is needed. The already working 24bit multichannel capture uses an DSP code, which splits an 32 bit sample into two halfs, sent it over two DMA channels and then alsa lib (???) joins it back into S32_LE sample. I'm trying to use same technique in reverse way using a pair of A_FXBUSes for one playback chanel, but I'm not able to set the DMA the way I need. Can someone with emu10k knowledge help me? ---- Ctirad

2 2

[alsa-devel] asihpi - how much further?
by Eliot Blennerhassett 23 Feb '08

23 Feb '08

Hi Takashi, My boss has started asking how long is it going to take to get our driver into mainline. I.e. a list of things we need to do, that when all those items have been dealt with, then the driver is accepted. Is it possible to make such a list? thanks and regards Eliot Blennerhassett www.audioscience.com P.S. Heres a quote that I feel has some relevance: http://linux-foundation.org/weblogs/openvoices/linus-torvalds-part-ii/ Jim Zemlin: But in a sense, over time, you win because you set the bar high and you encourage people to get it into the main line. Linus Torvalds: One of the problems is we have people who have so high criteria for what is acceptable or not that it scares away people who want to do new code and do new experiments. We mustn’t set the bar that high. New code, new drivers, there will be problems and I’d rather take them and then improve them than expect driver authors, especially when they stand outside the main tree and feel kind of like outsiders even though maybe they really are part of the same whole development community, but they feel like outsiders because their driver hasn’t made it into the tree yet. And then asking them to jump through hoops and make their driver perfect when they’re standing there alone and don’t have help; I think that’s unfair. And there are people in the kernel community that feel that way that things have to be just right before you can accept them and I’m much more of a laissez-faire kind of person. We don’t want to accept bad things, but on the other hand, hey, everything starts from less-than-perfect roots and it’s much better to accept things that work but may not be perfect and then improve on them when we can all improve on them and all the different vendors can fix the small nagging issues they have instead of keeping them at arm’s length until they’re perfect because maybe they’ll never be perfect without help.

3 5

[alsa-devel] make[3]: *** [sound/drivers/opl3/opl3_synth.o] Error 1
by Vegard Nossum 23 Feb '08

23 Feb '08

Hi. make randconfig (v2.6.25-rc2 + unrelated patches) found this: CC [M] sound/drivers/opl3/opl3_synth.o sound/drivers/opl3/opl3_synth.c: In function ‘snd_opl3_find_patch’: sound/drivers/opl3/opl3_synth.c:308: error: ‘OPL3_PATCH_HASH_SIZE’ undeclared (first use in this function) sound/drivers/opl3/opl3_synth.c:308: error: (Each undeclared identifier is reported only once sound/drivers/opl3/opl3_synth.c:308: error: for each function it appears in.) sound/drivers/opl3/opl3_synth.c:311: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:323: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:324: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c: In function ‘snd_opl3_clear_patches’: sound/drivers/opl3/opl3_synth.c:335: error: ‘OPL3_PATCH_HASH_SIZE’ undeclared (first use in this function) sound/drivers/opl3/opl3_synth.c:337: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ sound/drivers/opl3/opl3_synth.c:342: error: ‘struct snd_opl3’ has no member named ‘patch_table’ make[3]: *** [sound/drivers/opl3/opl3_synth.o] Error 1 make[2]: *** [sound/drivers/opl3] Error 2 make[1]: *** [sound/drivers] Error 2 make: *** [sound] Error 2 Config: # # Automatically generated make config: don't edit # Linux kernel version: 2.6.25-rc2 # Fri Feb 22 16:46:32 2008 # # CONFIG_64BIT is not set CONFIG_X86_32=y # CONFIG_X86_64 is not set CONFIG_X86=y # CONFIG_GENERIC_LOCKBREAK is not set CONFIG_GENERIC_TIME=y CONFIG_GENERIC_CMOS_UPDATE=y CONFIG_CLOCKSOURCE_WATCHDOG=y CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y CONFIG_LOCKDEP_SUPPORT=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_HAVE_LATENCYTOP_SUPPORT=y CONFIG_SEMAPHORE_SLEEPERS=y CONFIG_FAST_CMPXCHG_LOCAL=y CONFIG_MMU=y CONFIG_ZONE_DMA=y CONFIG_QUICKLIST=y CONFIG_GENERIC_ISA_DMA=y CONFIG_GENERIC_IOMAP=y CONFIG_GENERIC_BUG=y CONFIG_GENERIC_HWEIGHT=y # CONFIG_GENERIC_GPIO is not set CONFIG_ARCH_MAY_HAVE_PC_FDC=y CONFIG_DMI=y # CONFIG_RWSEM_GENERIC_SPINLOCK is not set CONFIG_RWSEM_XCHGADD_ALGORITHM=y # CONFIG_ARCH_HAS_ILOG2_U32 is not set # CONFIG_ARCH_HAS_ILOG2_U64 is not set CONFIG_ARCH_HAS_CPU_IDLE_WAIT=y CONFIG_GENERIC_CALIBRATE_DELAY=y # CONFIG_GENERIC_TIME_VSYSCALL is not set CONFIG_ARCH_HAS_CPU_RELAX=y # CONFIG_HAVE_SETUP_PER_CPU_AREA is not set CONFIG_ARCH_HIBERNATION_POSSIBLE=y CONFIG_ARCH_SUSPEND_POSSIBLE=y # CONFIG_ZONE_DMA32 is not set CONFIG_ARCH_POPULATES_NODE_MAP=y # CONFIG_AUDIT_ARCH is not set CONFIG_ARCH_SUPPORTS_AOUT=y CONFIG_GENERIC_HARDIRQS=y CONFIG_GENERIC_IRQ_PROBE=y CONFIG_GENERIC_PENDING_IRQ=y CONFIG_X86_SMP=y CONFIG_X86_32_SMP=y CONFIG_X86_HT=y CONFIG_X86_BIOS_REBOOT=y CONFIG_X86_TRAMPOLINE=y CONFIG_KTIME_SCALAR=y CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" # # General setup # # CONFIG_EXPERIMENTAL is not set CONFIG_LOCK_KERNEL=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_LOCALVERSION="" CONFIG_LOCALVERSION_AUTO=y # CONFIG_SWAP is not set # CONFIG_SYSVIPC is not set CONFIG_BSD_PROCESS_ACCT=y # CONFIG_BSD_PROCESS_ACCT_V3 is not set CONFIG_TASKSTATS=y # CONFIG_TASK_DELAY_ACCT is not set CONFIG_TASK_XACCT=y CONFIG_TASK_IO_ACCOUNTING=y CONFIG_AUDIT=y CONFIG_AUDITSYSCALL=y CONFIG_AUDIT_TREE=y # CONFIG_IKCONFIG is not set CONFIG_LOG_BUF_SHIFT=14 # CONFIG_CGROUPS is not set # CONFIG_GROUP_SCHED is not set CONFIG_RELAY=y # CONFIG_NAMESPACES is not set # CONFIG_BLK_DEV_INITRD is not set CONFIG_SYSCTL=y CONFIG_EMBEDDED=y CONFIG_UID16=y CONFIG_SYSCTL_SYSCALL=y CONFIG_KALLSYMS=y CONFIG_KALLSYMS_ALL=y # CONFIG_KALLSYMS_EXTRA_PASS is not set # CONFIG_HOTPLUG is not set CONFIG_PRINTK=y CONFIG_BUG=y CONFIG_ELF_CORE=y CONFIG_COMPAT_BRK=y CONFIG_BASE_FULL=y CONFIG_FUTEX=y CONFIG_ANON_INODES=y CONFIG_EPOLL=y # CONFIG_SIGNALFD is not set # CONFIG_TIMERFD is not set CONFIG_EVENTFD=y # CONFIG_SHMEM is not set # CONFIG_VM_EVENT_COUNTERS is not set # CONFIG_SLAB is not set # CONFIG_SLUB is not set CONFIG_SLOB=y # CONFIG_PROFILING is not set CONFIG_MARKERS=y CONFIG_HAVE_OPROFILE=y CONFIG_KPROBES=y CONFIG_HAVE_KPROBES=y CONFIG_RT_MUTEXES=y CONFIG_TINY_SHMEM=y CONFIG_BASE_SMALL=0 CONFIG_MODULES=y CONFIG_MODULE_UNLOAD=y # CONFIG_MODVERSIONS is not set # CONFIG_MODULE_SRCVERSION_ALL is not set # CONFIG_KMOD is not set CONFIG_STOP_MACHINE=y CONFIG_BLOCK=y # CONFIG_LBD is not set # CONFIG_LSF is not set # # IO Schedulers # CONFIG_IOSCHED_NOOP=y CONFIG_IOSCHED_AS=m CONFIG_IOSCHED_DEADLINE=y # CONFIG_IOSCHED_CFQ is not set # CONFIG_DEFAULT_AS is not set CONFIG_DEFAULT_DEADLINE=y # CONFIG_DEFAULT_CFQ is not set # CONFIG_DEFAULT_NOOP is not set CONFIG_DEFAULT_IOSCHED="deadline" CONFIG_CLASSIC_RCU=y # CONFIG_PREEMPT_RCU is not set # # Processor type and features # # CONFIG_TICK_ONESHOT is not set # CONFIG_NO_HZ is not set # CONFIG_HIGH_RES_TIMERS is not set CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_SMP=y # CONFIG_X86_PC is not set # CONFIG_X86_ELAN is not set # CONFIG_X86_VOYAGER is not set CONFIG_X86_NUMAQ=y # CONFIG_X86_SUMMIT is not set # CONFIG_X86_BIGSMP is not set # CONFIG_X86_VISWS is not set # CONFIG_X86_GENERICARCH is not set # CONFIG_X86_ES7000 is not set # CONFIG_X86_RDC321X is not set # CONFIG_X86_VSMP is not set # CONFIG_SCHED_NO_NO_OMIT_FRAME_POINTER is not set # CONFIG_PARAVIRT_GUEST is not set # CONFIG_M386 is not set # CONFIG_M486 is not set # CONFIG_M586 is not set # CONFIG_M586TSC is not set CONFIG_M586MMX=y # CONFIG_M686 is not set # CONFIG_MPENTIUMII is not set # CONFIG_MPENTIUMIII is not set # CONFIG_MPENTIUMM is not set # CONFIG_MPENTIUM4 is not set # CONFIG_MK6 is not set # CONFIG_MK7 is not set # CONFIG_MK8 is not set # CONFIG_MCRUSOE is not set # CONFIG_MEFFICEON is not set # CONFIG_MWINCHIPC6 is not set # CONFIG_MWINCHIP2 is not set # CONFIG_MWINCHIP3D is not set # CONFIG_MGEODEGX1 is not set # CONFIG_MGEODE_LX is not set # CONFIG_MCYRIXIII is not set # CONFIG_MVIAC3_2 is not set # CONFIG_MVIAC7 is not set # CONFIG_MPSC is not set # CONFIG_MCORE2 is not set # CONFIG_GENERIC_CPU is not set CONFIG_X86_GENERIC=y CONFIG_X86_CMPXCHG=y CONFIG_X86_L1_CACHE_SHIFT=7 CONFIG_X86_XADD=y # CONFIG_X86_PPRO_FENCE is not set CONFIG_X86_F00F_BUG=y CONFIG_X86_WP_WORKS_OK=y CONFIG_X86_INVLPG=y CONFIG_X86_BSWAP=y CONFIG_X86_POPAD_OK=y CONFIG_X86_ALIGNMENT_16=y CONFIG_X86_GOOD_APIC=y CONFIG_X86_INTEL_USERCOPY=y CONFIG_X86_MINIMUM_CPU_FAMILY=4 CONFIG_HPET_TIMER=y CONFIG_HPET_EMULATE_RTC=y # CONFIG_IOMMU_HELPER is not set CONFIG_NR_CPUS=32 # CONFIG_SCHED_SMT is not set CONFIG_SCHED_MC=y # CONFIG_PREEMPT_NONE is not set # CONFIG_PREEMPT_VOLUNTARY is not set CONFIG_PREEMPT=y CONFIG_RCU_TRACE=y CONFIG_X86_LOCAL_APIC=y CONFIG_X86_IO_APIC=y CONFIG_X86_MCE=y CONFIG_X86_MCE_NONFATAL=m # CONFIG_X86_MCE_P4THERMAL is not set CONFIG_VM86=y CONFIG_TOSHIBA=y CONFIG_I8K=y # CONFIG_X86_REBOOTFIXUPS is not set CONFIG_MICROCODE=m CONFIG_MICROCODE_OLD_INTERFACE=y CONFIG_X86_MSR=m CONFIG_X86_CPUID=m # CONFIG_NOHIGHMEM is not set # CONFIG_HIGHMEM4G is not set CONFIG_HIGHMEM64G=y CONFIG_PAGE_OFFSET=0xC0000000 CONFIG_HIGHMEM=y CONFIG_X86_PAE=y CONFIG_NUMA=y CONFIG_NODES_SHIFT=4 CONFIG_HAVE_ARCH_BOOTMEM_NODE=y CONFIG_ARCH_HAVE_MEMORY_PRESENT=y CONFIG_NEED_NODE_MEMMAP_SIZE=y CONFIG_HAVE_ARCH_ALLOC_REMAP=y CONFIG_ARCH_DISCONTIGMEM_ENABLE=y CONFIG_ARCH_DISCONTIGMEM_DEFAULT=y CONFIG_ARCH_SPARSEMEM_ENABLE=y CONFIG_ARCH_SELECT_MEMORY_MODEL=y CONFIG_SELECT_MEMORY_MODEL=y # CONFIG_FLATMEM_MANUAL is not set CONFIG_DISCONTIGMEM_MANUAL=y # CONFIG_SPARSEMEM_MANUAL is not set CONFIG_DISCONTIGMEM=y CONFIG_FLAT_NODE_MEM_MAP=y CONFIG_NEED_MULTIPLE_NODES=y CONFIG_HAVE_MEMORY_PRESENT=y CONFIG_SPARSEMEM_STATIC=y # CONFIG_SPARSEMEM_VMEMMAP_ENABLE is not set CONFIG_SPLIT_PTLOCK_CPUS=4 CONFIG_MIGRATION=y CONFIG_RESOURCES_64BIT=y CONFIG_ZONE_DMA_FLAG=1 CONFIG_BOUNCE=y CONFIG_NR_QUICK=1 CONFIG_VIRT_TO_BUS=y CONFIG_HIGHPTE=y # CONFIG_MATH_EMULATION is not set CONFIG_MTRR=y CONFIG_IRQBALANCE=y CONFIG_HZ_100=y # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set CONFIG_HZ=100 # CONFIG_SCHED_HRTICK is not set # CONFIG_KEXEC is not set CONFIG_PHYSICAL_START=0x1000000 CONFIG_PHYSICAL_ALIGN=0x100000 CONFIG_COMPAT_VDSO=y CONFIG_ARCH_ENABLE_MEMORY_HOTPLUG=y # CONFIG_HAVE_ARCH_EARLY_PFN_TO_NID is not set # # Power management options # # CONFIG_PM is not set # # CPU Frequency scaling # CONFIG_CPU_FREQ=y CONFIG_CPU_FREQ_TABLE=y CONFIG_CPU_FREQ_DEBUG=y # CONFIG_CPU_FREQ_STAT is not set CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE=y # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set CONFIG_CPU_FREQ_GOV_PERFORMANCE=y CONFIG_CPU_FREQ_GOV_POWERSAVE=m # CONFIG_CPU_FREQ_GOV_USERSPACE is not set CONFIG_CPU_FREQ_GOV_ONDEMAND=y # CONFIG_CPU_FREQ_GOV_CONSERVATIVE is not set # # CPUFreq processor drivers # CONFIG_X86_POWERNOW_K6=y CONFIG_X86_POWERNOW_K7=y CONFIG_X86_POWERNOW_K8=m CONFIG_X86_GX_SUSPMOD=m CONFIG_X86_SPEEDSTEP_CENTRINO=m CONFIG_X86_SPEEDSTEP_CENTRINO_TABLE=y # CONFIG_X86_SPEEDSTEP_ICH is not set CONFIG_X86_P4_CLOCKMOD=y # CONFIG_X86_LONGRUN is not set # # shared options # CONFIG_X86_SPEEDSTEP_LIB=y CONFIG_CPU_IDLE=y CONFIG_CPU_IDLE_GOV_LADDER=y # # Bus options (PCI etc.) # CONFIG_PCI=y # CONFIG_PCI_GOBIOS is not set CONFIG_PCI_GOMMCONFIG=y # CONFIG_PCI_GODIRECT is not set # CONFIG_PCI_GOANY is not set CONFIG_PCI_DOMAINS=y # CONFIG_PCIEPORTBUS is not set CONFIG_ARCH_SUPPORTS_MSI=y # CONFIG_PCI_MSI is not set CONFIG_PCI_LEGACY=y CONFIG_HT_IRQ=y CONFIG_ISA_DMA_API=y CONFIG_ISA=y CONFIG_EISA=y # CONFIG_EISA_PCI_EISA is not set CONFIG_EISA_NAMES=y # CONFIG_MCA is not set CONFIG_SCx200=y CONFIG_SCx200HR_TIMER=y CONFIG_K8_NB=y # # Executable file formats / Emulations # CONFIG_BINFMT_ELF=y CONFIG_BINFMT_AOUT=m # CONFIG_BINFMT_MISC is not set # # Networking # CONFIG_NET=y # # Networking options # CONFIG_PACKET=y # CONFIG_PACKET_MMAP is not set CONFIG_UNIX=y CONFIG_XFRM=y # CONFIG_XFRM_USER is not set CONFIG_NET_KEY=m CONFIG_INET=y CONFIG_IP_MULTICAST=y CONFIG_IP_ADVANCED_ROUTER=y # CONFIG_ASK_IP_FIB_HASH is not set CONFIG_IP_FIB_TRIE=y # CONFIG_IP_FIB_HASH is not set # CONFIG_IP_FIB_TRIE_STATS is not set # CONFIG_IP_MULTIPLE_TABLES is not set # CONFIG_IP_ROUTE_MULTIPATH is not set CONFIG_IP_ROUTE_VERBOSE=y # CONFIG_IP_PNP is not set # CONFIG_NET_IPIP is not set CONFIG_NET_IPGRE=m # CONFIG_NET_IPGRE_BROADCAST is not set CONFIG_IP_MROUTE=y CONFIG_IP_PIMSM_V1=y CONFIG_IP_PIMSM_V2=y CONFIG_SYN_COOKIES=y CONFIG_INET_AH=y CONFIG_INET_ESP=m # CONFIG_INET_IPCOMP is not set # CONFIG_INET_XFRM_TUNNEL is not set CONFIG_INET_TUNNEL=m CONFIG_INET_XFRM_MODE_TRANSPORT=y CONFIG_INET_XFRM_MODE_TUNNEL=m CONFIG_INET_XFRM_MODE_BEET=m CONFIG_INET_LRO=y # CONFIG_INET_DIAG is not set # CONFIG_TCP_CONG_ADVANCED is not set CONFIG_TCP_CONG_CUBIC=y CONFIG_DEFAULT_TCP_CONG="cubic" CONFIG_IPV6=y # CONFIG_IPV6_PRIVACY is not set CONFIG_IPV6_ROUTER_PREF=y CONFIG_INET6_AH=m # CONFIG_INET6_ESP is not set # CONFIG_INET6_IPCOMP is not set # CONFIG_INET6_XFRM_TUNNEL is not set CONFIG_INET6_TUNNEL=m # CONFIG_INET6_XFRM_MODE_TRANSPORT is not set # CONFIG_INET6_XFRM_MODE_TUNNEL is not set CONFIG_INET6_XFRM_MODE_BEET=m CONFIG_IPV6_SIT=m CONFIG_IPV6_TUNNEL=m CONFIG_NETWORK_SECMARK=y # CONFIG_NETFILTER is not set CONFIG_ATM=y CONFIG_ATM_CLIP=m # CONFIG_ATM_CLIP_NO_ICMP is not set CONFIG_ATM_LANE=y # CONFIG_ATM_MPOA is not set # CONFIG_ATM_BR2684 is not set CONFIG_BRIDGE=y CONFIG_VLAN_8021Q=m CONFIG_DECNET=y CONFIG_LLC=y CONFIG_LLC2=m # CONFIG_IPX is not set CONFIG_ATALK=y CONFIG_DEV_APPLETALK=m # CONFIG_LTPC is not set CONFIG_COPS=m # CONFIG_COPS_DAYNA is not set # CONFIG_COPS_TANGENT is not set # CONFIG_IPDDP is not set # CONFIG_NET_SCHED is not set # # Network testing # # CONFIG_HAMRADIO is not set CONFIG_CAN=m # CONFIG_CAN_RAW is not set CONFIG_CAN_BCM=m # # CAN Device Drivers # # CONFIG_CAN_VCAN is not set CONFIG_CAN_DEBUG_DEVICES=y # CONFIG_IRDA is not set # CONFIG_BT is not set # # Wireless # CONFIG_CFG80211=y CONFIG_NL80211=y CONFIG_WIRELESS_EXT=y # CONFIG_MAC80211 is not set CONFIG_IEEE80211=y CONFIG_IEEE80211_DEBUG=y CONFIG_IEEE80211_CRYPT_WEP=y # CONFIG_IEEE80211_CRYPT_CCMP is not set CONFIG_IEEE80211_CRYPT_TKIP=y # CONFIG_RFKILL is not set # # Device Drivers # # # Generic Driver Options # CONFIG_STANDALONE=y CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=y # CONFIG_SYS_HYPERVISOR is not set CONFIG_CONNECTOR=y CONFIG_PROC_EVENTS=y CONFIG_MTD=m # CONFIG_MTD_DEBUG is not set CONFIG_MTD_CONCAT=m # CONFIG_MTD_PARTITIONS is not set # # User Modules And Translation Layers # CONFIG_MTD_CHAR=m CONFIG_MTD_BLKDEVS=m # CONFIG_MTD_BLOCK is not set # CONFIG_MTD_BLOCK_RO is not set # CONFIG_FTL is not set # CONFIG_NFTL is not set # CONFIG_INFTL is not set CONFIG_RFD_FTL=m # CONFIG_SSFDC is not set CONFIG_MTD_OOPS=m # # RAM/ROM/Flash chip drivers # CONFIG_MTD_CFI=m # CONFIG_MTD_JEDECPROBE is not set CONFIG_MTD_GEN_PROBE=m CONFIG_MTD_CFI_ADV_OPTIONS=y # CONFIG_MTD_CFI_NOSWAP is not set CONFIG_MTD_CFI_BE_BYTE_SWAP=y # CONFIG_MTD_CFI_LE_BYTE_SWAP is not set CONFIG_MTD_CFI_GEOMETRY=y CONFIG_MTD_MAP_BANK_WIDTH_1=y CONFIG_MTD_MAP_BANK_WIDTH_2=y # CONFIG_MTD_MAP_BANK_WIDTH_4 is not set CONFIG_MTD_MAP_BANK_WIDTH_8=y # CONFIG_MTD_MAP_BANK_WIDTH_16 is not set # CONFIG_MTD_MAP_BANK_WIDTH_32 is not set # CONFIG_MTD_CFI_I1 is not set CONFIG_MTD_CFI_I2=y CONFIG_MTD_CFI_I4=y CONFIG_MTD_CFI_I8=y CONFIG_MTD_OTP=y CONFIG_MTD_CFI_INTELEXT=m # CONFIG_MTD_CFI_AMDSTD is not set CONFIG_MTD_CFI_STAA=m CONFIG_MTD_CFI_UTIL=m CONFIG_MTD_RAM=m CONFIG_MTD_ROM=m # CONFIG_MTD_ABSENT is not set # # Mapping drivers for chip access # CONFIG_MTD_COMPLEX_MAPPINGS=y # CONFIG_MTD_PHYSMAP is not set CONFIG_MTD_SC520CDP=m # CONFIG_MTD_TS5500 is not set CONFIG_MTD_PCI=m # CONFIG_MTD_INTEL_VR_NOR is not set CONFIG_MTD_PLATRAM=m # # Self-contained MTD device drivers # # CONFIG_MTD_PMC551 is not set # CONFIG_MTD_SLRAM is not set CONFIG_MTD_PHRAM=m # CONFIG_MTD_MTDRAM is not set CONFIG_MTD_BLOCK2MTD=m # # Disk-On-Chip Device Drivers # CONFIG_MTD_DOC2000=m # CONFIG_MTD_DOC2001 is not set CONFIG_MTD_DOC2001PLUS=m CONFIG_MTD_DOCPROBE=m CONFIG_MTD_DOCECC=m # CONFIG_MTD_DOCPROBE_ADVANCED is not set CONFIG_MTD_DOCPROBE_ADDRESS=0 # CONFIG_MTD_NAND is not set CONFIG_MTD_NAND_IDS=m CONFIG_MTD_ONENAND=m CONFIG_MTD_ONENAND_VERIFY_WRITE=y # CONFIG_MTD_ONENAND_OTP is not set # CONFIG_MTD_ONENAND_2X_PROGRAM is not set # # UBI - Unsorted block images # CONFIG_MTD_UBI=m CONFIG_MTD_UBI_WL_THRESHOLD=4096 CONFIG_MTD_UBI_BEB_RESERVE=1 # CONFIG_MTD_UBI_GLUEBI is not set # # UBI debugging options # # CONFIG_PARPORT is not set # CONFIG_PNP is not set CONFIG_BLK_DEV=y CONFIG_BLK_DEV_FD=y # CONFIG_BLK_DEV_XD is not set CONFIG_BLK_CPQ_DA=y # CONFIG_BLK_CPQ_CISS_DA is not set CONFIG_BLK_DEV_DAC960=y # CONFIG_BLK_DEV_COW_COMMON is not set CONFIG_BLK_DEV_LOOP=y CONFIG_BLK_DEV_CRYPTOLOOP=m CONFIG_BLK_DEV_NBD=y CONFIG_BLK_DEV_SX8=y # CONFIG_BLK_DEV_UB is not set CONFIG_BLK_DEV_RAM=y CONFIG_BLK_DEV_RAM_COUNT=16 CONFIG_BLK_DEV_RAM_SIZE=4096 # CONFIG_BLK_DEV_XIP is not set CONFIG_CDROM_PKTCDVD=m CONFIG_CDROM_PKTCDVD_BUFFERS=8 CONFIG_ATA_OVER_ETH=y # CONFIG_MISC_DEVICES is not set CONFIG_HAVE_IDE=y # CONFIG_IDE is not set # # SCSI device support # CONFIG_RAID_ATTRS=m # CONFIG_SCSI is not set # CONFIG_SCSI_DMA is not set # CONFIG_SCSI_NETLINK is not set # CONFIG_ATA is not set # CONFIG_MD is not set # CONFIG_FUSION is not set # # IEEE 1394 (FireWire) support # # # An alternative FireWire stack is available with EXPERIMENTAL=y # CONFIG_IEEE1394=y # # Subsystem Options # # CONFIG_IEEE1394_VERBOSEDEBUG is not set # # Controllers # CONFIG_IEEE1394_PCILYNX=m CONFIG_IEEE1394_OHCI1394=y # # Protocols # # CONFIG_IEEE1394_VIDEO1394 is not set # # SBP-2 support (for storage devices) requires SCSI # # CONFIG_IEEE1394_ETH1394_ROM_ENTRY is not set CONFIG_IEEE1394_DV1394=m CONFIG_IEEE1394_RAWIO=y # CONFIG_I2O is not set CONFIG_MACINTOSH_DRIVERS=y # CONFIG_MAC_EMUMOUSEBTN is not set CONFIG_NETDEVICES=y CONFIG_NETDEVICES_MULTIQUEUE=y CONFIG_DUMMY=y CONFIG_BONDING=y CONFIG_EQUALIZER=m CONFIG_TUN=y # CONFIG_VETH is not set CONFIG_ARCNET=m CONFIG_ARCNET_1201=m # CONFIG_ARCNET_1051 is not set CONFIG_ARCNET_RAW=m CONFIG_ARCNET_CAP=m CONFIG_ARCNET_COM90xx=m CONFIG_ARCNET_COM90xxIO=m # CONFIG_ARCNET_RIM_I is not set CONFIG_ARCNET_COM20020=m # CONFIG_ARCNET_COM20020_ISA is not set CONFIG_ARCNET_COM20020_PCI=m # CONFIG_PHYLIB is not set CONFIG_NET_ETHERNET=y CONFIG_MII=y CONFIG_HAPPYMEAL=m CONFIG_SUNGEM=y # CONFIG_CASSINI is not set CONFIG_NET_VENDOR_3COM=y CONFIG_EL1=m CONFIG_EL2=y CONFIG_ELPLUS=y # CONFIG_EL3 is not set # CONFIG_3C515 is not set CONFIG_VORTEX=y CONFIG_TYPHOON=m CONFIG_LANCE=m # CONFIG_NET_VENDOR_SMC is not set # CONFIG_NET_VENDOR_RACAL is not set CONFIG_NET_TULIP=y CONFIG_TULIP=y # CONFIG_TULIP_MMIO is not set # CONFIG_TULIP_NAPI is not set CONFIG_DE4X5=y CONFIG_WINBOND_840=y CONFIG_DM9102=y CONFIG_ULI526X=y CONFIG_DEPCA=m # CONFIG_HP100 is not set # CONFIG_NET_ISA is not set # CONFIG_IBM_NEW_EMAC_ZMII is not set # CONFIG_IBM_NEW_EMAC_RGMII is not set # CONFIG_IBM_NEW_EMAC_TAH is not set # CONFIG_IBM_NEW_EMAC_EMAC4 is not set CONFIG_NET_PCI=y CONFIG_PCNET32=y # CONFIG_PCNET32_NAPI is not set CONFIG_AMD8111_ETH=y # CONFIG_AMD8111E_NAPI is not set CONFIG_ADAPTEC_STARFIRE=y # CONFIG_APRICOT is not set CONFIG_B44=m CONFIG_B44_PCI_AUTOSELECT=y CONFIG_B44_PCICORE_AUTOSELECT=y CONFIG_B44_PCI=y # CONFIG_FORCEDETH is not set # CONFIG_CS89x0 is not set CONFIG_EEPRO100=y # CONFIG_E100 is not set # CONFIG_FEALNX is not set CONFIG_NATSEMI=m CONFIG_NE2K_PCI=y # CONFIG_8139TOO is not set CONFIG_R6040=m CONFIG_SIS900=m CONFIG_EPIC100=y CONFIG_SUNDANCE=m # CONFIG_SUNDANCE_MMIO is not set CONFIG_TLAN=y CONFIG_VIA_RHINE=y # CONFIG_VIA_RHINE_MMIO is not set CONFIG_VIA_RHINE_NAPI=y # CONFIG_NETDEV_1000 is not set CONFIG_NETDEV_10000=y CONFIG_CHELSIO_T1=m CONFIG_CHELSIO_T1_1G=y CONFIG_CHELSIO_T1_NAPI=y # CONFIG_CHELSIO_T3 is not set CONFIG_IXGBE=y # CONFIG_IXGB is not set CONFIG_S2IO=m CONFIG_MYRI10GE=y # CONFIG_NETXEN_NIC is not set CONFIG_NIU=m CONFIG_MLX4_CORE=m # CONFIG_MLX4_DEBUG is not set CONFIG_TEHUTI=m CONFIG_BNX2X=y CONFIG_TR=y CONFIG_IBMTR=m # CONFIG_IBMOL is not set # CONFIG_IBMLS is not set CONFIG_3C359=y CONFIG_TMS380TR=y CONFIG_TMSPCI=m CONFIG_SKISA=m CONFIG_PROTEON=y # CONFIG_ABYSS is not set CONFIG_SMCTR=y # # Wireless LAN # # CONFIG_WLAN_PRE80211 is not set # CONFIG_WLAN_80211 is not set # # USB Network Adapters # # CONFIG_USB_KAWETH is not set CONFIG_USB_PEGASUS=m # CONFIG_USB_USBNET is not set # CONFIG_WAN is not set # CONFIG_ATM_DRIVERS is not set CONFIG_FDDI=y CONFIG_DEFXX=m # CONFIG_DEFXX_MMIO is not set # CONFIG_SKFP is not set # CONFIG_PPP is not set # CONFIG_SLIP is not set # CONFIG_NETPOLL is not set # CONFIG_NET_POLL_CONTROLLER is not set CONFIG_ISDN=y CONFIG_ISDN_I4L=m # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y CONFIG_ISDN_TTY_FAX=y # # ISDN feature submodules # # CONFIG_ISDN_DIVERSION is not set # # ISDN4Linux hardware drivers # # # Passive cards # CONFIG_ISDN_DRV_HISAX=m # # D-channel protocol features # # CONFIG_HISAX_EURO is not set # CONFIG_HISAX_1TR6 is not set # CONFIG_HISAX_NI1 is not set CONFIG_HISAX_MAX_CARDS=8 # # HiSax supported cards # CONFIG_HISAX_16_0=y CONFIG_HISAX_16_3=y # CONFIG_HISAX_TELESPCI is not set CONFIG_HISAX_S0BOX=y # CONFIG_HISAX_AVM_A1 is not set CONFIG_HISAX_FRITZPCI=y # CONFIG_HISAX_AVM_A1_PCMCIA is not set # CONFIG_HISAX_ELSA is not set # CONFIG_HISAX_IX1MICROR2 is not set CONFIG_HISAX_DIEHLDIVA=y # CONFIG_HISAX_ASUSCOM is not set CONFIG_HISAX_TELEINT=y CONFIG_HISAX_HFCS=y CONFIG_HISAX_SEDLBAUER=y CONFIG_HISAX_SPORTSTER=y # CONFIG_HISAX_MIC is not set CONFIG_HISAX_NETJET=y CONFIG_HISAX_NETJET_U=y CONFIG_HISAX_NICCY=y # CONFIG_HISAX_ISURF is not set # CONFIG_HISAX_HSTSAPHIR is not set # CONFIG_HISAX_BKM_A4T is not set CONFIG_HISAX_SCT_QUADRO=y # CONFIG_HISAX_GAZEL is not set # CONFIG_HISAX_HFC_PCI is not set # CONFIG_HISAX_W6692 is not set # CONFIG_HISAX_HFC_SX is not set # CONFIG_HISAX_ENTERNOW_PCI is not set # CONFIG_HISAX_DEBUG is not set # # HiSax PCMCIA card service modules # # # HiSax sub driver modules # # # Active cards # CONFIG_ISDN_DRV_ICN=m # CONFIG_ISDN_DRV_PCBIT is not set # CONFIG_ISDN_DRV_SC is not set # CONFIG_ISDN_DRV_ACT2000 is not set # CONFIG_ISDN_DRV_GIGASET is not set CONFIG_ISDN_CAPI=m # CONFIG_ISDN_DRV_AVMB1_VERBOSE_REASON is not set # CONFIG_CAPI_TRACE is not set # CONFIG_ISDN_CAPI_CAPI20 is not set CONFIG_ISDN_CAPI_CAPIDRV=m # # CAPI hardware drivers # CONFIG_CAPI_AVM=y CONFIG_ISDN_DRV_AVMB1_B1ISA=m # CONFIG_ISDN_DRV_AVMB1_B1PCI is not set # CONFIG_ISDN_DRV_AVMB1_T1ISA is not set # CONFIG_ISDN_DRV_AVMB1_B1PCMCIA is not set # CONFIG_ISDN_DRV_AVMB1_T1PCI is not set CONFIG_ISDN_DRV_AVMB1_C4=m CONFIG_CAPI_EICON=y CONFIG_PHONE=m # CONFIG_PHONE_IXJ is not set # # Input device support # CONFIG_INPUT=y CONFIG_INPUT_FF_MEMLESS=y CONFIG_INPUT_POLLDEV=m # # Userland interfaces # CONFIG_INPUT_MOUSEDEV=y # CONFIG_INPUT_MOUSEDEV_PSAUX is not set CONFIG_INPUT_MOUSEDEV_SCREEN_X=1024 CONFIG_INPUT_MOUSEDEV_SCREEN_Y=768 CONFIG_INPUT_JOYDEV=y CONFIG_INPUT_EVDEV=m CONFIG_INPUT_EVBUG=m # # Input Device Drivers # # CONFIG_INPUT_KEYBOARD is not set CONFIG_INPUT_MOUSE=y CONFIG_MOUSE_PS2=y # CONFIG_MOUSE_PS2_ALPS is not set CONFIG_MOUSE_PS2_LOGIPS2PP=y CONFIG_MOUSE_PS2_SYNAPTICS=y # CONFIG_MOUSE_PS2_LIFEBOOK is not set CONFIG_MOUSE_PS2_TRACKPOINT=y # CONFIG_MOUSE_PS2_TOUCHKIT is not set CONFIG_MOUSE_SERIAL=m CONFIG_MOUSE_APPLETOUCH=m CONFIG_MOUSE_INPORT=m CONFIG_MOUSE_ATIXL=y CONFIG_MOUSE_LOGIBM=y CONFIG_MOUSE_PC110PAD=m CONFIG_MOUSE_VSXXXAA=y CONFIG_INPUT_JOYSTICK=y CONFIG_JOYSTICK_ANALOG=y CONFIG_JOYSTICK_A3D=m CONFIG_JOYSTICK_ADI=m CONFIG_JOYSTICK_COBRA=y CONFIG_JOYSTICK_GF2K=m # CONFIG_JOYSTICK_GRIP is not set CONFIG_JOYSTICK_GRIP_MP=y CONFIG_JOYSTICK_GUILLEMOT=y CONFIG_JOYSTICK_INTERACT=y # CONFIG_JOYSTICK_SIDEWINDER is not set CONFIG_JOYSTICK_TMDC=y # CONFIG_JOYSTICK_IFORCE is not set CONFIG_JOYSTICK_WARRIOR=y CONFIG_JOYSTICK_MAGELLAN=y CONFIG_JOYSTICK_SPACEORB=m CONFIG_JOYSTICK_SPACEBALL=m CONFIG_JOYSTICK_STINGER=m CONFIG_JOYSTICK_TWIDJOY=m CONFIG_JOYSTICK_JOYDUMP=y CONFIG_JOYSTICK_XPAD=y # CONFIG_JOYSTICK_XPAD_FF is not set # CONFIG_JOYSTICK_XPAD_LEDS is not set CONFIG_INPUT_TABLET=y # CONFIG_TABLET_USB_ACECAD is not set # CONFIG_TABLET_USB_AIPTEK is not set CONFIG_TABLET_USB_GTCO=m CONFIG_TABLET_USB_KBTAB=m CONFIG_TABLET_USB_WACOM=y # CONFIG_INPUT_TOUCHSCREEN is not set # CONFIG_INPUT_MISC is not set # # Hardware I/O ports # CONFIG_SERIO=y CONFIG_SERIO_I8042=y CONFIG_SERIO_SERPORT=y CONFIG_SERIO_CT82C710=y CONFIG_SERIO_PCIPS2=y CONFIG_SERIO_LIBPS2=y CONFIG_SERIO_RAW=y CONFIG_GAMEPORT=y CONFIG_GAMEPORT_NS558=y # CONFIG_GAMEPORT_L4 is not set # CONFIG_GAMEPORT_EMU10K1 is not set CONFIG_GAMEPORT_FM801=y # # Character devices # # CONFIG_VT is not set # CONFIG_SERIAL_NONSTANDARD is not set # # Serial drivers # CONFIG_SERIAL_8250=y CONFIG_SERIAL_8250_CONSOLE=y CONFIG_FIX_EARLYCON_MEM=y CONFIG_SERIAL_8250_PCI=m CONFIG_SERIAL_8250_NR_UARTS=4 CONFIG_SERIAL_8250_RUNTIME_UARTS=4 # CONFIG_SERIAL_8250_EXTENDED is not set # # Non-8250 serial port support # CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y CONFIG_SERIAL_JSM=m CONFIG_UNIX98_PTYS=y CONFIG_LEGACY_PTYS=y CONFIG_LEGACY_PTY_COUNT=256 # CONFIG_IPMI_HANDLER is not set # CONFIG_HW_RANDOM is not set CONFIG_NVRAM=y CONFIG_RTC=m CONFIG_GEN_RTC=y CONFIG_GEN_RTC_X=y CONFIG_DTLK=y CONFIG_R3964=m CONFIG_APPLICOM=y # CONFIG_MWAVE is not set CONFIG_SCx200_GPIO=m CONFIG_PC8736x_GPIO=m CONFIG_NSC_GPIO=y CONFIG_CS5535_GPIO=m CONFIG_RAW_DRIVER=y CONFIG_MAX_RAW_DEVS=256 # CONFIG_HANGCHECK_TIMER is not set CONFIG_DEVPORT=y CONFIG_I2C=m CONFIG_I2C_BOARDINFO=y # CONFIG_I2C_CHARDEV is not set # # I2C Algorithms # CONFIG_I2C_ALGOBIT=m # CONFIG_I2C_ALGOPCF is not set # CONFIG_I2C_ALGOPCA is not set # # I2C Hardware Bus support # # CONFIG_I2C_ALI1535 is not set CONFIG_I2C_ALI15X3=m CONFIG_I2C_AMD756=m CONFIG_I2C_AMD8111=m # CONFIG_I2C_I801 is not set CONFIG_I2C_I810=m CONFIG_I2C_PIIX4=m # CONFIG_I2C_NFORCE2 is not set CONFIG_I2C_PARPORT_LIGHT=m # CONFIG_I2C_PROSAVAGE is not set # CONFIG_I2C_SAVAGE4 is not set CONFIG_I2C_SIMTEC=m CONFIG_SCx200_I2C=m CONFIG_SCx200_I2C_SCL=12 CONFIG_SCx200_I2C_SDA=13 # CONFIG_SCx200_ACB is not set CONFIG_I2C_SIS5595=m # CONFIG_I2C_SIS630 is not set CONFIG_I2C_SIS96X=m # CONFIG_I2C_TINY_USB is not set CONFIG_I2C_VIAPRO=m # CONFIG_I2C_VOODOO3 is not set # CONFIG_I2C_PCA_ISA is not set # # Miscellaneous I2C Chip support # CONFIG_PCF8575=m CONFIG_TPS65010=m CONFIG_I2C_DEBUG_CORE=y CONFIG_I2C_DEBUG_ALGO=y # CONFIG_I2C_DEBUG_BUS is not set CONFIG_I2C_DEBUG_CHIP=y # # SPI support # # CONFIG_SPI is not set # CONFIG_SPI_MASTER is not set CONFIG_W1=m CONFIG_W1_CON=y # # 1-wire Bus Masters # # CONFIG_W1_MASTER_MATROX is not set CONFIG_W1_MASTER_DS2490=m # # 1-wire Slaves # CONFIG_W1_SLAVE_THERM=m # CONFIG_W1_SLAVE_SMEM is not set # CONFIG_W1_SLAVE_DS2433 is not set CONFIG_W1_SLAVE_DS2760=m # CONFIG_POWER_SUPPLY is not set # CONFIG_HWMON is not set CONFIG_THERMAL=y CONFIG_WATCHDOG=y CONFIG_WATCHDOG_NOWAYOUT=y # # Watchdog Device Drivers # CONFIG_SOFT_WATCHDOG=m CONFIG_ACQUIRE_WDT=m CONFIG_ADVANTECH_WDT=y # CONFIG_ALIM1535_WDT is not set # CONFIG_ALIM7101_WDT is not set CONFIG_SC520_WDT=y CONFIG_EUROTECH_WDT=y # CONFIG_IB700_WDT is not set # CONFIG_IBMASR is not set # CONFIG_WAFER_WDT is not set CONFIG_I6300ESB_WDT=y CONFIG_ITCO_WDT=m CONFIG_ITCO_VENDOR_SUPPORT=y # CONFIG_IT8712F_WDT is not set CONFIG_SC1200_WDT=m # CONFIG_SCx200_WDT is not set CONFIG_PC87413_WDT=m CONFIG_60XX_WDT=y CONFIG_SBC8360_WDT=m CONFIG_SBC7240_WDT=m # CONFIG_CPU5_WDT is not set CONFIG_SMSC37B787_WDT=m # CONFIG_W83627HF_WDT is not set CONFIG_W83697HF_WDT=m CONFIG_W83877F_WDT=y # CONFIG_W83977F_WDT is not set # CONFIG_MACHZ_WDT is not set CONFIG_SBC_EPX_C3_WATCHDOG=y # # ISA-based Watchdog Cards # CONFIG_PCWATCHDOG=y # CONFIG_MIXCOMWD is not set CONFIG_WDT=m # CONFIG_WDT_501 is not set # # PCI-based Watchdog Cards # CONFIG_PCIPCWATCHDOG=y CONFIG_WDTPCI=y # CONFIG_WDT_501_PCI is not set # # USB-based Watchdog Cards # CONFIG_USBPCWATCHDOG=m # # Sonics Silicon Backplane # CONFIG_SSB_POSSIBLE=y CONFIG_SSB=y CONFIG_SSB_PCIHOST_POSSIBLE=y CONFIG_SSB_PCIHOST=y # CONFIG_SSB_SILENT is not set CONFIG_SSB_DEBUG=y CONFIG_SSB_DRIVER_PCICORE_POSSIBLE=y CONFIG_SSB_DRIVER_PCICORE=y # # Multifunction device drivers # CONFIG_MFD_SM501=y # # Multimedia devices # CONFIG_VIDEO_DEV=y CONFIG_VIDEO_V4L1=y CONFIG_VIDEO_V4L1_COMPAT=y CONFIG_VIDEO_V4L2=y CONFIG_VIDEO_CAPTURE_DRIVERS=y CONFIG_VIDEO_ADV_DEBUG=y # CONFIG_VIDEO_HELPER_CHIPS_AUTO is not set # # Encoders/decoders and other helper chips # # # Audio decoders # # CONFIG_VIDEO_TVAUDIO is not set CONFIG_VIDEO_TDA7432=m # CONFIG_VIDEO_TDA9840 is not set # CONFIG_VIDEO_TDA9875 is not set CONFIG_VIDEO_TEA6415C=m # CONFIG_VIDEO_TEA6420 is not set # CONFIG_VIDEO_MSP3400 is not set CONFIG_VIDEO_CS53L32A=m CONFIG_VIDEO_WM8775=m CONFIG_VIDEO_WM8739=m # # Video decoders # # CONFIG_VIDEO_BT819 is not set CONFIG_VIDEO_BT856=m CONFIG_VIDEO_BT866=m # CONFIG_VIDEO_KS0127 is not set CONFIG_VIDEO_OV7670=m # CONFIG_VIDEO_TCM825X is not set CONFIG_VIDEO_SAA7110=m CONFIG_VIDEO_SAA7111=m # CONFIG_VIDEO_SAA7114 is not set # CONFIG_VIDEO_SAA711X is not set CONFIG_VIDEO_SAA7191=m # CONFIG_VIDEO_TVP5150 is not set # CONFIG_VIDEO_VPX3220 is not set # # Video and audio decoders # # # MPEG video encoders # # # Video encoders # # CONFIG_VIDEO_SAA7127 is not set # CONFIG_VIDEO_SAA7185 is not set CONFIG_VIDEO_ADV7170=m CONFIG_VIDEO_ADV7175=m # # Video improvement chips # CONFIG_VIDEO_UPD64031A=m CONFIG_VIDEO_UPD64083=m CONFIG_VIDEO_VIVI=y CONFIG_VIDEO_BT848=m CONFIG_VIDEO_BT848_DVB=y # CONFIG_VIDEO_SAA6588 is not set CONFIG_VIDEO_PMS=y CONFIG_VIDEO_CPIA=y CONFIG_VIDEO_CPIA_USB=y CONFIG_VIDEO_CPIA2=m # CONFIG_VIDEO_SAA5246A is not set # CONFIG_VIDEO_SAA5249 is not set # CONFIG_TUNER_3036 is not set CONFIG_VIDEO_ZORAN_ZR36060=m CONFIG_VIDEO_ZORAN=m # CONFIG_VIDEO_ZORAN_BUZ is not set CONFIG_VIDEO_ZORAN_DC10=m CONFIG_VIDEO_ZORAN_DC30=m # CONFIG_VIDEO_ZORAN_LML33 is not set CONFIG_VIDEO_ZORAN_LML33R10=m # CONFIG_VIDEO_SAA7134 is not set # CONFIG_VIDEO_MXB is not set CONFIG_VIDEO_DPC=m # CONFIG_VIDEO_HEXIUM_ORION is not set # CONFIG_VIDEO_HEXIUM_GEMINI is not set # CONFIG_VIDEO_CX88 is not set # CONFIG_VIDEO_CX23885 is not set # CONFIG_VIDEO_CAFE_CCIC is not set CONFIG_V4L_USB_DRIVERS=y # CONFIG_VIDEO_EM28XX is not set CONFIG_VIDEO_USBVISION=m CONFIG_VIDEO_USBVIDEO=m CONFIG_USB_IBMCAM=m CONFIG_USB_KONICAWC=m # CONFIG_USB_QUICKCAM_MESSENGER is not set # CONFIG_USB_ET61X251 is not set CONFIG_VIDEO_OVCAMCHIP=m # CONFIG_USB_W9968CF is not set CONFIG_USB_OV511=m # CONFIG_USB_SE401 is not set # CONFIG_USB_SN9C102 is not set # CONFIG_USB_STV680 is not set CONFIG_USB_ZC0301=y CONFIG_USB_PWC=m CONFIG_USB_PWC_DEBUG=y CONFIG_USB_ZR364XX=m # CONFIG_RADIO_ADAPTERS is not set CONFIG_DVB_CORE=m CONFIG_DVB_CORE_ATTACH=y # CONFIG_DVB_CAPTURE_DRIVERS is not set CONFIG_DVB_BT8XX=m CONFIG_VIDEO_SAA7146=m CONFIG_VIDEO_SAA7146_VV=m CONFIG_VIDEO_TUNER=m # CONFIG_VIDEO_TUNER_CUSTOMIZE is not set CONFIG_TUNER_XC2028=m CONFIG_TUNER_MT20XX=m CONFIG_TUNER_TDA8290=m CONFIG_TUNER_TEA5761=m CONFIG_TUNER_TEA5767=m CONFIG_TUNER_SIMPLE=m CONFIG_TUNER_TDA9887=m CONFIG_VIDEOBUF_GEN=y CONFIG_VIDEOBUF_DMA_SG=m CONFIG_VIDEOBUF_VMALLOC=y CONFIG_VIDEO_BTCX=m CONFIG_VIDEO_IR_I2C=m CONFIG_VIDEO_IR=m CONFIG_VIDEO_TVEEPROM=m # CONFIG_DAB is not set # # Graphics support # CONFIG_AGP=m # CONFIG_AGP_ALI is not set CONFIG_AGP_ATI=m CONFIG_AGP_AMD=m CONFIG_AGP_AMD64=m # CONFIG_AGP_INTEL is not set # CONFIG_AGP_NVIDIA is not set CONFIG_AGP_SIS=m CONFIG_AGP_SWORKS=m # CONFIG_AGP_VIA is not set # CONFIG_AGP_EFFICEON is not set CONFIG_DRM=m # CONFIG_DRM_TDFX is not set # CONFIG_DRM_R128 is not set # CONFIG_DRM_RADEON is not set # CONFIG_DRM_MGA is not set # CONFIG_DRM_SIS is not set # CONFIG_DRM_VIA is not set CONFIG_DRM_SAVAGE=m CONFIG_VGASTATE=m # CONFIG_VIDEO_OUTPUT_CONTROL is not set CONFIG_FB=m # CONFIG_FIRMWARE_EDID is not set CONFIG_FB_DDC=m CONFIG_FB_CFB_FILLRECT=m CONFIG_FB_CFB_COPYAREA=m CONFIG_FB_CFB_IMAGEBLIT=m # CONFIG_FB_CFB_REV_PIXELS_IN_BYTE is not set CONFIG_FB_SYS_FILLRECT=m CONFIG_FB_SYS_COPYAREA=m CONFIG_FB_SYS_IMAGEBLIT=m CONFIG_FB_SYS_FOPS=m CONFIG_FB_DEFERRED_IO=y # CONFIG_FB_SVGALIB is not set # CONFIG_FB_MACMODES is not set # CONFIG_FB_BACKLIGHT is not set CONFIG_FB_MODE_HELPERS=y CONFIG_FB_TILEBLITTING=y # # Frame buffer hardware drivers # # CONFIG_FB_CIRRUS is not set # CONFIG_FB_PM2 is not set # CONFIG_FB_CYBER2000 is not set CONFIG_FB_ARC=m CONFIG_FB_VGA16=m CONFIG_FB_UVESA=m # CONFIG_FB_HECUBA is not set # CONFIG_FB_HGA is not set # CONFIG_FB_S1D13XXX is not set # CONFIG_FB_NVIDIA is not set # CONFIG_FB_RIVA is not set CONFIG_FB_LE80578=m # CONFIG_FB_CARILLO_RANCH is not set CONFIG_FB_MATROX=m CONFIG_FB_MATROX_MILLENIUM=y CONFIG_FB_MATROX_MYSTIQUE=y CONFIG_FB_MATROX_G=y CONFIG_FB_MATROX_I2C=m # CONFIG_FB_MATROX_MAVEN is not set # CONFIG_FB_MATROX_MULTIHEAD is not set CONFIG_FB_RADEON=m CONFIG_FB_RADEON_I2C=y # CONFIG_FB_RADEON_BACKLIGHT is not set CONFIG_FB_RADEON_DEBUG=y # CONFIG_FB_ATY128 is not set # CONFIG_FB_ATY is not set # CONFIG_FB_S3 is not set # CONFIG_FB_SIS is not set CONFIG_FB_NEOMAGIC=m # CONFIG_FB_KYRO is not set # CONFIG_FB_3DFX is not set CONFIG_FB_VOODOO1=m # CONFIG_FB_VT8623 is not set CONFIG_FB_CYBLA=m # CONFIG_FB_TRIDENT is not set # CONFIG_FB_ARK is not set CONFIG_FB_SM501=m # CONFIG_FB_VIRTUAL is not set CONFIG_BACKLIGHT_LCD_SUPPORT=y # CONFIG_LCD_CLASS_DEVICE is not set CONFIG_BACKLIGHT_CLASS_DEVICE=y CONFIG_BACKLIGHT_CORGI=y CONFIG_BACKLIGHT_PROGEAR=y # # Display device support # CONFIG_DISPLAY_SUPPORT=m # # Display hardware drivers # CONFIG_VIDEO_SELECT=y CONFIG_LOGO=y # CONFIG_LOGO_LINUX_MONO is not set CONFIG_LOGO_LINUX_VGA16=y # CONFIG_LOGO_LINUX_CLUT224 is not set # # Sound # CONFIG_SOUND=y # # Advanced Linux Sound Architecture # CONFIG_SND=m CONFIG_SND_TIMER=m CONFIG_SND_PCM=m CONFIG_SND_HWDEP=m CONFIG_SND_RAWMIDI=m # CONFIG_SND_SEQUENCER is not set CONFIG_SND_OSSEMUL=y CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_PCM_OSS_PLUGINS=y CONFIG_SND_RTCTIMER=m # CONFIG_SND_DYNAMIC_MINORS is not set # CONFIG_SND_SUPPORT_OLD_API is not set CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y # CONFIG_SND_DEBUG_DETECT is not set # # Generic devices # CONFIG_SND_MPU401_UART=m CONFIG_SND_OPL3_LIB=m CONFIG_SND_OPL4_LIB=m CONFIG_SND_AC97_CODEC=m # CONFIG_SND_DUMMY is not set # CONFIG_SND_MTPAV is not set # CONFIG_SND_SERIAL_U16550 is not set # CONFIG_SND_MPU401 is not set CONFIG_SND_AD1848_LIB=m CONFIG_SND_CS4231_LIB=m CONFIG_SND_SB_COMMON=m CONFIG_SND_SB16_DSP=m # # ISA devices # CONFIG_SND_ADLIB=m # CONFIG_SND_AD1848 is not set # CONFIG_SND_CMI8330 is not set CONFIG_SND_CS4231=m # CONFIG_SND_CS4232 is not set CONFIG_SND_CS4236=m # CONFIG_SND_ES1688 is not set CONFIG_SND_ES18XX=m # CONFIG_SND_SC6000 is not set CONFIG_SND_GUS_SYNTH=m CONFIG_SND_GUSCLASSIC=m # CONFIG_SND_GUSEXTREME is not set # CONFIG_SND_GUSMAX is not set CONFIG_SND_OPL3SA2=m CONFIG_SND_OPTI92X_AD1848=m CONFIG_SND_OPTI92X_CS4231=m # CONFIG_SND_OPTI93X is not set # CONFIG_SND_MIRO is not set # CONFIG_SND_SB8 is not set CONFIG_SND_SB16=m CONFIG_SND_SBAWE=m CONFIG_SND_SB16_CSP=y CONFIG_SND_SB16_CSP_FIRMWARE_IN_KERNEL=y # CONFIG_SND_SGALAXY is not set # CONFIG_SND_SSCAPE is not set CONFIG_SND_WAVEFRONT=m # CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL is not set # # PCI devices # CONFIG_SND_AD1889=m # CONFIG_SND_ALS300 is not set CONFIG_SND_ALS4000=m # CONFIG_SND_ALI5451 is not set # CONFIG_SND_ATIIXP is not set # CONFIG_SND_ATIIXP_MODEM is not set # CONFIG_SND_AU8810 is not set # CONFIG_SND_AU8820 is not set CONFIG_SND_AU8830=m CONFIG_SND_BT87X=m # CONFIG_SND_BT87X_OVERCLOCK is not set CONFIG_SND_CA0106=m # CONFIG_SND_CMIPCI is not set CONFIG_SND_OXYGEN_LIB=m # CONFIG_SND_OXYGEN is not set # CONFIG_SND_CS4281 is not set # CONFIG_SND_CS46XX is not set CONFIG_SND_CS5530=m # CONFIG_SND_CS5535AUDIO is not set CONFIG_SND_DARLA20=m CONFIG_SND_GINA20=m # CONFIG_SND_LAYLA20 is not set # CONFIG_SND_DARLA24 is not set CONFIG_SND_GINA24=m # CONFIG_SND_LAYLA24 is not set # CONFIG_SND_MONA is not set # CONFIG_SND_MIA is not set # CONFIG_SND_ECHO3G is not set CONFIG_SND_INDIGO=m # CONFIG_SND_INDIGOIO is not set # CONFIG_SND_INDIGODJ is not set # CONFIG_SND_EMU10K1 is not set # CONFIG_SND_EMU10K1X is not set CONFIG_SND_ENS1370=m # CONFIG_SND_ENS1371 is not set CONFIG_SND_ES1938=m # CONFIG_SND_ES1968 is not set # CONFIG_SND_FM801 is not set # CONFIG_SND_HDA_INTEL is not set # CONFIG_SND_HDSP is not set CONFIG_SND_HDSPM=m CONFIG_SND_HIFIER=m CONFIG_SND_ICE1712=m # CONFIG_SND_ICE1724 is not set # CONFIG_SND_INTEL8X0 is not set # CONFIG_SND_INTEL8X0M is not set # CONFIG_SND_KORG1212 is not set CONFIG_SND_MAESTRO3=m # CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL is not set # CONFIG_SND_MIXART is not set CONFIG_SND_NM256=m # CONFIG_SND_PCXHR is not set CONFIG_SND_RIPTIDE=m CONFIG_SND_RME32=m # CONFIG_SND_RME96 is not set CONFIG_SND_RME9652=m # CONFIG_SND_SIS7019 is not set CONFIG_SND_SONICVIBES=m # CONFIG_SND_TRIDENT is not set # CONFIG_SND_VIA82XX is not set # CONFIG_SND_VIA82XX_MODEM is not set CONFIG_SND_VIRTUOSO=m # CONFIG_SND_VX222 is not set CONFIG_SND_YMFPCI=m # CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL is not set # # USB devices # # CONFIG_SND_USB_AUDIO is not set # CONFIG_SND_USB_USX2Y is not set CONFIG_SND_USB_CAIAQ=m # CONFIG_SND_USB_CAIAQ_INPUT is not set # # System on Chip audio support # CONFIG_SND_SOC=m # # SoC Audio support for SuperH # # # ALSA SoC audio for Freescale SOCs # # # Open Sound System # CONFIG_SOUND_PRIME=m # CONFIG_SOUND_TRIDENT is not set 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 is not set CONFIG_AC97_BUS=m CONFIG_HID_SUPPORT=y CONFIG_HID=y CONFIG_HID_DEBUG=y # CONFIG_HIDRAW is not set # # USB Input Devices # CONFIG_USB_HID=y # CONFIG_USB_HIDINPUT_POWERBOOK is not set CONFIG_USB_HIDDEV=y CONFIG_USB_SUPPORT=y CONFIG_USB_ARCH_HAS_HCD=y CONFIG_USB_ARCH_HAS_OHCI=y CONFIG_USB_ARCH_HAS_EHCI=y CONFIG_USB=y # CONFIG_USB_DEBUG is not set # CONFIG_USB_ANNOUNCE_NEW_DEVICES is not set # # Miscellaneous USB options # # CONFIG_USB_DEVICEFS is not set # CONFIG_USB_DEVICE_CLASS is not set # # USB Host Controller Drivers # CONFIG_USB_EHCI_HCD=m CONFIG_USB_ISP116X_HCD=m # CONFIG_USB_OHCI_HCD is not set # CONFIG_USB_UHCI_HCD is not set # CONFIG_USB_U132_HCD is not set CONFIG_USB_SL811_HCD=m CONFIG_USB_R8A66597_HCD=y # # USB Device Class drivers # CONFIG_USB_ACM=y CONFIG_USB_PRINTER=y # # NOTE: USB_STORAGE enables SCSI, and 'SCSI disk support' # # # may also be needed; see USB_STORAGE Help for more information # # CONFIG_USB_LIBUSUAL is not set # # USB Imaging devices # # CONFIG_USB_MON is not set # # USB port drivers # CONFIG_USB_SERIAL=y CONFIG_USB_EZUSB=y # CONFIG_USB_SERIAL_GENERIC is not set CONFIG_USB_SERIAL_AIRPRIME=m CONFIG_USB_SERIAL_BELKIN=y # CONFIG_USB_SERIAL_CH341 is not set CONFIG_USB_SERIAL_WHITEHEAT=y CONFIG_USB_SERIAL_DIGI_ACCELEPORT=m CONFIG_USB_SERIAL_EMPEG=m CONFIG_USB_SERIAL_FUNSOFT=m # CONFIG_USB_SERIAL_VISOR is not set CONFIG_USB_SERIAL_IPAQ=y CONFIG_USB_SERIAL_EDGEPORT=y CONFIG_USB_SERIAL_EDGEPORT_TI=m CONFIG_USB_SERIAL_GARMIN=m # CONFIG_USB_SERIAL_KEYSPAN_PDA is not set CONFIG_USB_SERIAL_KEYSPAN=m # CONFIG_USB_SERIAL_KEYSPAN_MPR is not set # CONFIG_USB_SERIAL_KEYSPAN_USA28 is not set CONFIG_USB_SERIAL_KEYSPAN_USA28X=y # CONFIG_USB_SERIAL_KEYSPAN_USA28XA is not set CONFIG_USB_SERIAL_KEYSPAN_USA28XB=y CONFIG_USB_SERIAL_KEYSPAN_USA19=y CONFIG_USB_SERIAL_KEYSPAN_USA18X=y # CONFIG_USB_SERIAL_KEYSPAN_USA19W is not set CONFIG_USB_SERIAL_KEYSPAN_USA19QW=y # CONFIG_USB_SERIAL_KEYSPAN_USA19QI is not set CONFIG_USB_SERIAL_KEYSPAN_USA49W=y CONFIG_USB_SERIAL_KEYSPAN_USA49WLC=y CONFIG_USB_SERIAL_KOBIL_SCT=m CONFIG_USB_SERIAL_MCT_U232=y CONFIG_USB_SERIAL_MOS7720=m CONFIG_USB_SERIAL_MOS7840=m CONFIG_USB_SERIAL_NAVMAN=y CONFIG_USB_SERIAL_PL2303=y CONFIG_USB_SERIAL_OTI6858=y CONFIG_USB_SERIAL_HP4X=y CONFIG_USB_SERIAL_SIERRAWIRELESS=y CONFIG_USB_SERIAL_TI=m # CONFIG_USB_SERIAL_XIRCOM is not set CONFIG_USB_SERIAL_OPTION=y CONFIG_USB_SERIAL_DEBUG=y # # USB Miscellaneous drivers # CONFIG_USB_EMI62=y CONFIG_USB_EMI26=m # CONFIG_USB_LCD is not set # CONFIG_USB_BERRY_CHARGE is not set CONFIG_USB_LED=y # CONFIG_USB_CYPRESS_CY7C63 is not set CONFIG_USB_CYTHERM=y CONFIG_USB_PHIDGET=y # CONFIG_USB_PHIDGETKIT is not set # CONFIG_USB_PHIDGETMOTORCONTROL is not set CONFIG_USB_PHIDGETSERVO=y # CONFIG_USB_IDMOUSE is not set CONFIG_USB_FTDI_ELAN=y CONFIG_USB_APPLEDISPLAY=y # CONFIG_USB_SISUSBVGA is not set CONFIG_USB_LD=y CONFIG_USB_TRANCEVIBRATOR=m # CONFIG_USB_IOWARRIOR is not set # CONFIG_USB_ATM is not set CONFIG_USB_GADGET=m CONFIG_USB_GADGET_DEBUG_FS=y CONFIG_USB_GADGET_SELECTED=y # CONFIG_USB_GADGET_AMD5536UDC is not set # CONFIG_USB_GADGET_ATMEL_USBA is not set # CONFIG_USB_GADGET_FSL_USB2 is not set # CONFIG_USB_GADGET_NET2280 is not set # CONFIG_USB_GADGET_PXA2XX is not set CONFIG_USB_GADGET_M66592=y CONFIG_USB_M66592=m # CONFIG_USB_GADGET_GOKU is not set # CONFIG_USB_GADGET_LH7A40X is not set # CONFIG_USB_GADGET_OMAP is not set # CONFIG_USB_GADGET_S3C2410 is not set # CONFIG_USB_GADGET_AT91 is not set # CONFIG_USB_GADGET_DUMMY_HCD is not set CONFIG_USB_GADGET_DUALSPEED=y # CONFIG_USB_ZERO is not set CONFIG_USB_ETH=m # CONFIG_USB_GADGETFS is not set CONFIG_USB_FILE_STORAGE=m # CONFIG_USB_FILE_STORAGE_TEST is not set CONFIG_USB_G_SERIAL=m # CONFIG_USB_MIDI_GADGET is not set CONFIG_USB_G_PRINTER=m CONFIG_MMC=y # CONFIG_MMC_DEBUG is not set CONFIG_MMC_UNSAFE_RESUME=y # # MMC/SD Card Drivers # # CONFIG_MMC_BLOCK is not set CONFIG_SDIO_UART=m # # MMC/SD Host Controller Drivers # CONFIG_MMC_SDHCI=m CONFIG_MMC_WBSD=y # CONFIG_MEMSTICK is not set CONFIG_NEW_LEDS=y CONFIG_LEDS_CLASS=y # # LED drivers # CONFIG_LEDS_NET48XX=m CONFIG_LEDS_WRAP=m # # LED Triggers # # CONFIG_LEDS_TRIGGERS is not set CONFIG_INFINIBAND=y CONFIG_INFINIBAND_USER_MAD=y CONFIG_INFINIBAND_USER_ACCESS=m CONFIG_INFINIBAND_USER_MEM=y CONFIG_INFINIBAND_ADDR_TRANS=y CONFIG_INFINIBAND_MTHCA=m CONFIG_INFINIBAND_MTHCA_DEBUG=y CONFIG_INFINIBAND_AMSO1100=y CONFIG_INFINIBAND_AMSO1100_DEBUG=y CONFIG_MLX4_INFINIBAND=m CONFIG_INFINIBAND_NES=m CONFIG_INFINIBAND_NES_DEBUG=y # CONFIG_INFINIBAND_IPOIB is not set # CONFIG_RTC_CLASS is not set # # Userspace I/O # # CONFIG_UIO is not set # # Firmware Drivers # CONFIG_EDD=y CONFIG_DELL_RBU=m CONFIG_DCDBAS=m # CONFIG_DMIID is not set # # File systems # CONFIG_EXT2_FS=m CONFIG_EXT2_FS_XATTR=y # CONFIG_EXT2_FS_POSIX_ACL is not set # CONFIG_EXT2_FS_SECURITY is not set CONFIG_EXT2_FS_XIP=y CONFIG_FS_XIP=y # CONFIG_EXT3_FS is not set CONFIG_FS_MBCACHE=m CONFIG_REISERFS_FS=y # CONFIG_REISERFS_CHECK is not set CONFIG_REISERFS_FS_XATTR=y CONFIG_REISERFS_FS_POSIX_ACL=y CONFIG_REISERFS_FS_SECURITY=y CONFIG_JFS_FS=m # CONFIG_JFS_POSIX_ACL is not set CONFIG_JFS_SECURITY=y CONFIG_JFS_DEBUG=y CONFIG_JFS_STATISTICS=y CONFIG_FS_POSIX_ACL=y CONFIG_XFS_FS=y # CONFIG_XFS_QUOTA is not set # CONFIG_XFS_SECURITY is not set # CONFIG_XFS_POSIX_ACL is not set CONFIG_XFS_RT=y CONFIG_DNOTIFY=y CONFIG_INOTIFY=y # CONFIG_INOTIFY_USER is not set # CONFIG_QUOTA is not set # CONFIG_AUTOFS_FS is not set CONFIG_AUTOFS4_FS=y # CONFIG_FUSE_FS is not set CONFIG_GENERIC_ACL=y # # CD-ROM/DVD Filesystems # CONFIG_ISO9660_FS=m CONFIG_JOLIET=y # CONFIG_ZISOFS is not set # CONFIG_UDF_FS is not set # # DOS/FAT/NT Filesystems # CONFIG_FAT_FS=y CONFIG_MSDOS_FS=y CONFIG_VFAT_FS=m CONFIG_FAT_DEFAULT_CODEPAGE=437 CONFIG_FAT_DEFAULT_IOCHARSET="iso8859-1" # CONFIG_NTFS_FS is not set # # Pseudo filesystems # # CONFIG_PROC_FS is not set # CONFIG_SYSFS is not set CONFIG_TMPFS=y CONFIG_TMPFS_POSIX_ACL=y # CONFIG_HUGETLBFS is not set # CONFIG_HUGETLB_PAGE is not set # # Miscellaneous filesystems # CONFIG_HFSPLUS_FS=m # CONFIG_JFFS2_FS is not set # CONFIG_CRAMFS is not set CONFIG_VXFS_FS=y # CONFIG_MINIX_FS is not set CONFIG_HPFS_FS=m CONFIG_QNX4FS_FS=m # CONFIG_ROMFS_FS is not set CONFIG_SYSV_FS=m CONFIG_UFS_FS=m CONFIG_UFS_DEBUG=y # CONFIG_NETWORK_FILESYSTEMS is not set # # Partition Types # CONFIG_PARTITION_ADVANCED=y # CONFIG_ACORN_PARTITION is not set CONFIG_OSF_PARTITION=y CONFIG_AMIGA_PARTITION=y CONFIG_ATARI_PARTITION=y CONFIG_MAC_PARTITION=y # CONFIG_MSDOS_PARTITION is not set # CONFIG_LDM_PARTITION is not set # CONFIG_SGI_PARTITION is not set CONFIG_ULTRIX_PARTITION=y # CONFIG_SUN_PARTITION is not set CONFIG_KARMA_PARTITION=y CONFIG_EFI_PARTITION=y # CONFIG_SYSV68_PARTITION is not set CONFIG_NLS=y CONFIG_NLS_DEFAULT="iso8859-1" CONFIG_NLS_CODEPAGE_437=y CONFIG_NLS_CODEPAGE_737=m CONFIG_NLS_CODEPAGE_775=y # CONFIG_NLS_CODEPAGE_850 is not set CONFIG_NLS_CODEPAGE_852=y CONFIG_NLS_CODEPAGE_855=m CONFIG_NLS_CODEPAGE_857=y CONFIG_NLS_CODEPAGE_860=m # CONFIG_NLS_CODEPAGE_861 is not set # CONFIG_NLS_CODEPAGE_862 is not set CONFIG_NLS_CODEPAGE_863=m # CONFIG_NLS_CODEPAGE_864 is not set # CONFIG_NLS_CODEPAGE_865 is not set CONFIG_NLS_CODEPAGE_866=m CONFIG_NLS_CODEPAGE_869=y CONFIG_NLS_CODEPAGE_936=y CONFIG_NLS_CODEPAGE_950=m CONFIG_NLS_CODEPAGE_932=y # CONFIG_NLS_CODEPAGE_949 is not set CONFIG_NLS_CODEPAGE_874=m CONFIG_NLS_ISO8859_8=m CONFIG_NLS_CODEPAGE_1250=y # CONFIG_NLS_CODEPAGE_1251 is not set # CONFIG_NLS_ASCII is not set CONFIG_NLS_ISO8859_1=m # CONFIG_NLS_ISO8859_2 is not set CONFIG_NLS_ISO8859_3=y CONFIG_NLS_ISO8859_4=y CONFIG_NLS_ISO8859_5=y # CONFIG_NLS_ISO8859_6 is not set CONFIG_NLS_ISO8859_7=y CONFIG_NLS_ISO8859_9=y CONFIG_NLS_ISO8859_13=m # CONFIG_NLS_ISO8859_14 is not set # CONFIG_NLS_ISO8859_15 is not set CONFIG_NLS_KOI8_R=m CONFIG_NLS_KOI8_U=y CONFIG_NLS_UTF8=y # # Kernel hacking # CONFIG_TRACE_IRQFLAGS_SUPPORT=y CONFIG_PRINTK_TIME=y # CONFIG_ENABLE_WARN_DEPRECATED is not set # CONFIG_ENABLE_MUST_CHECK is not set # CONFIG_MAGIC_SYSRQ is not set # CONFIG_UNUSED_SYMBOLS is not set CONFIG_DEBUG_FS=y CONFIG_HEADERS_CHECK=y # CONFIG_DEBUG_KERNEL is not set CONFIG_SCHED_DEBUG=y CONFIG_SCHEDSTATS=y CONFIG_STACKTRACE=y # CONFIG_DEBUG_BUGVERBOSE is not set CONFIG_FRAME_POINTER=y CONFIG_LATENCYTOP=y # CONFIG_PROVIDE_OHCI1394_DMA_INIT is not set # CONFIG_SAMPLES is not set # CONFIG_EARLY_PRINTK is not set CONFIG_X86_FIND_SMP_CONFIG=y CONFIG_X86_MPPARSE=y CONFIG_DOUBLEFAULT=y CONFIG_IO_DELAY_TYPE_0X80=0 CONFIG_IO_DELAY_TYPE_0XED=1 CONFIG_IO_DELAY_TYPE_UDELAY=2 CONFIG_IO_DELAY_TYPE_NONE=3 # CONFIG_IO_DELAY_0X80 is not set # CONFIG_IO_DELAY_0XED is not set # CONFIG_IO_DELAY_UDELAY is not set CONFIG_IO_DELAY_NONE=y CONFIG_DEFAULT_IO_DELAY_TYPE=3 # # Security options # # CONFIG_KEYS is not set CONFIG_CRYPTO=y CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_AEAD=y CONFIG_CRYPTO_BLKCIPHER=y CONFIG_CRYPTO_SEQIV=m CONFIG_CRYPTO_HASH=y CONFIG_CRYPTO_MANAGER=y CONFIG_CRYPTO_HMAC=y CONFIG_CRYPTO_NULL=m CONFIG_CRYPTO_MD4=y CONFIG_CRYPTO_MD5=y CONFIG_CRYPTO_SHA1=y CONFIG_CRYPTO_SHA256=y CONFIG_CRYPTO_SHA512=y # CONFIG_CRYPTO_WP512 is not set # CONFIG_CRYPTO_TGR192 is not set CONFIG_CRYPTO_GF128MUL=m CONFIG_CRYPTO_ECB=y CONFIG_CRYPTO_CBC=m # CONFIG_CRYPTO_PCBC is not set CONFIG_CRYPTO_CTR=m CONFIG_CRYPTO_GCM=m CONFIG_CRYPTO_CCM=m CONFIG_CRYPTO_CRYPTD=y CONFIG_CRYPTO_DES=m CONFIG_CRYPTO_FCRYPT=y CONFIG_CRYPTO_BLOWFISH=y # CONFIG_CRYPTO_TWOFISH is not set # CONFIG_CRYPTO_TWOFISH_586 is not set CONFIG_CRYPTO_SERPENT=y CONFIG_CRYPTO_AES=y CONFIG_CRYPTO_AES_586=y CONFIG_CRYPTO_CAST5=y CONFIG_CRYPTO_CAST6=m # CONFIG_CRYPTO_TEA is not set CONFIG_CRYPTO_ARC4=y CONFIG_CRYPTO_KHAZAD=m CONFIG_CRYPTO_ANUBIS=y # CONFIG_CRYPTO_SEED is not set CONFIG_CRYPTO_DEFLATE=y CONFIG_CRYPTO_MICHAEL_MIC=y CONFIG_CRYPTO_CRC32C=m CONFIG_CRYPTO_CAMELLIA=m # CONFIG_CRYPTO_TEST is not set CONFIG_CRYPTO_AUTHENC=y CONFIG_CRYPTO_LZO=y # CONFIG_CRYPTO_HW is not set CONFIG_HAVE_KVM=y CONFIG_VIRTUALIZATION=y CONFIG_VIRTIO=m CONFIG_VIRTIO_RING=m CONFIG_VIRTIO_BALLOON=m # # Library routines # CONFIG_BITREVERSE=y CONFIG_CRC_CCITT=m # CONFIG_CRC16 is not set # CONFIG_CRC_ITU_T is not set CONFIG_CRC32=y CONFIG_CRC7=y CONFIG_LIBCRC32C=m CONFIG_AUDIT_GENERIC=y CONFIG_ZLIB_INFLATE=y CONFIG_ZLIB_DEFLATE=y CONFIG_LZO_COMPRESS=y CONFIG_LZO_DECOMPRESS=y CONFIG_PLIST=y CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y

2 1

[alsa-devel] Fix num_adc_nids assignment for eeepc p701 config preset
by Herton Ronaldo Krzesinski 23 Feb '08

23 Feb '08

Fix num_adc_nids assignment for eeepc p701 config preset. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 586d98f..6f7dd96 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13388,7 +13388,7 @@ static struct alc_config_preset alc662_presets[] = { alc662_eeepc_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, -- []'s Herton

2 1

[alsa-devel] asihpi: Need help converting volatile to memory barriers
by Eliot Blennerhassett 22 Feb '08

22 Feb '08

Greetings, first, the meta-help request: Is there another place that I should be making this request? (lkml, kernel newbies, linux driver project etc)? Theres quite a bit written about barriers, but most seems to be assuming SMP situation or memory mapped devices. Not much about devices doing DMA. I.e I have read Documentation/memory-barriers.txt Now the actual question: Currently I have a driver that uses "volatile" - heres the relevant source. http://hg.alsa-project.org/alsa-driver/file/89222d702376/pci/asihpi/hpi6205… The "volatile" is applied to structures that are either read or written by device DMA. Certainly the driver in its current state doesn't work without volatile qualifier. (BTW the device doesn't use host interrupts) Structures ("interface") used for dma are allocated with dma_alloc_coherent() In the following, am I using the barriers correctly? 1) Reading something updated by DMA volatile struct bus_master_interface *interface; while (interface->ack != OK) { sleep(a while) [ device changes interface->ack by dma ] }; === after conversion struct bus_master_interface *interface; while (interface->ack != OK) { sleep(a while); rmb(); }; Here the volatile or rmb is needed or the loop gets optimised away. 2) Writing to memory, interrupt device volatile struct bus_master_interface *interface; interface->cmd = command; iowrite(device_interrupt, 1); === after conversion === struct bus_master_interface *interface; interface->cmd = command; wmb(); iowrite(device_interrupt, 1); [device reads interface->cmd by dma] Is the wmb() a guarantee that the command will be in memory visible to the device when the driver informs it of a new command? 3) One assumption I am making is that the compiler is not going to optimise across functions E.g. in the following scenario, is the compiler going to optimise the loop away without a rmb()? If not, is this because of something inherent in the C standard, or just because the optimiser isn't yet smart enough to see it? I.e. it might work now, but when whole-file-optimisation is introduced, it will fail? int get_ack(interface) { return interface->ack } ... while (get_ack(interface) != OK) { sleep(a while); } regards -- Eliot Blennerhassett www.audioscience.com

3 2

[alsa-devel] [PATCH] hda: Mic as output fix
by Matthew Ranostay 22 Feb '08

22 Feb '08

Added logic to check if AUTO_PIN_FRONT_MIC is available for output switch, if AUTO_PIN_MIC isn't. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r a5754a5e8133 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 21 14:11:09 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Feb 22 08:19:47 2008 -0700 @@ -2307,6 +2307,29 @@ static int create_controls(struct sigmat return 0; } +static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else if (spec->multiout.num_dacs > 4) { + printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); + return 1; + } else { + spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; + spec->multiout.num_dacs++; + } + return 0; +} + +static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (is_in_dac_nids(spec, nid)) + return 1; + if (spec->multiout.hp_nid == nid) + return 1; + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -2369,10 +2392,11 @@ static int stac92xx_auto_create_multi_ou if (spec->mic_switch) { unsigned int def_conf; - nid = cfg->input_pins[AUTO_PIN_MIC]; + unsigned int mic_pin = AUTO_PIN_MIC; +again: + nid = cfg->input_pins[mic_pin]; def_conf = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { @@ -2382,35 +2406,19 @@ static int stac92xx_auto_create_multi_ou err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (nid << 8) | 1); + nid = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (!check_in_dac_nids(spec, nid)) + add_spec_dacs(spec, nid); if (err < 0) return err; } + } else if (mic_pin == AUTO_PIN_MIC) { + mic_pin = AUTO_PIN_FRONT_MIC; + goto again; } } - return 0; -} - -static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (is_in_dac_nids(spec, nid)) - return 1; - if (spec->multiout.hp_nid == nid) - return 1; - return 0; -} - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else if (spec->multiout.num_dacs > 4) { - printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); - return 1; - } else { - spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; - spec->multiout.num_dacs++; - } return 0; }

2 1

[alsa-devel] [PATCH] RME 9632 fix at 192kHz
by Remy Bruno 22 Feb '08

22 Feb '08

Hello, I realized that one of my hdsp patches was not included in the hg tree, so I guess I forgot to submit it, sorry for submitting only now. Description: The bits indicating SPDIF frequency in the status register are not the same for the 9632 than for the other cards, because it also supports 192kHz. A specific bitmask has thus been added (used in hdsp_spdif_sample_rate()). The 9632 does not seem to report external sample rates greater than 96kHz. In this case, the best seems to report spdif rate when autosync reference is spdif. This also required to move function hdsp_spdif_sample_rate(). Signed-off-by: Remy Bruno <remy.bruno(a)trinnov.com> Regards, Remy

2 1

[alsa-devel] [PATCH] alsa-lib: issues in snd_seq_change_bit and some enhancements
by Aldrin Martoq 22 Feb '08

22 Feb '08

Hello hackers, I found two issues in sequencer part of alsa-lib while working with the python binding: 1) snd_seq_change_bit is not working as expected. 2) There is no sane way to clear the event_filter of snd_seq_client_info_t structure. Because the structure is opaque, we cannot memset to zero because the user doesn't know the size of the bitmap. The first attached patch is a demo of both issues. For 1) please apply snd_seq_change_bit.txt patch. For 2) I propose to clear the bitmap if NULL is passed, see snd_seq_client_info_set_event_filter.txt patch. Also, I think it makes no sense that the bitmap is returned or expected to be given in both snd_seq_client_info_{set,get}_event_filter; the client_info structure is opaque and the alsa-lib user can't/shouldn't known the bitmap size. The only sane way is to use the middle interface function snd_seq_set_client_event_filter(), but it only can be used for an opened sequencer's client_info. So, the last patch is a proposal for changing this: - mark as deprecated both snd_seq_client_info_{set,get}_event_filter - snd_seq_client_info_event_filter_clear() clears the event_filter - snd_seq_client_info_event_filter_add() add an event to the event filter - snd_seq_client_info_event_filter_del() removes an event from the event filter - snd_seq_client_info_event_filter_check() test if an event is present in the filtering Any comments? Thanks! -- Aldrin Martoq <amartoq(a)dcc.uchile.cl>

3 5

[alsa-devel] ASoC - Output widgets
by Alexandre BOUIN 22 Feb '08

22 Feb '08

Hi list, Using ASoC layer, I'd like to perform this example behaviour with my widgets : - 1 output mixer MIXOUT is connected to 2 outputs OUT1 and OUT2 - OUT1 and OUT2 can be powered independently This seems easy, but atm while one is powered, the other is powered too. Does someone have already made this widget behavior ? Regards, Alex.

1 0

[alsa-devel] Strange (HW?) bug with RME9632 + AO4S + Mobo DG33BU
by Remy Bruno 22 Feb '08

22 Feb '08

Hi, I came across a quite strange bug using the RME 9632 with an intel motherboard DG33BU, whose symptom is random xruns. Conditions: - Sound card = RME 9632. I tried several ones (recent and older ones), so this is not due to a defective card - AI4S/AO4S (daughter boards) pluged. The bug does not happen without these. I also tried several ones. - Sample rate does not matter (48kHz or 96kHz for example) - Last alsa driver/lib (from HG tree) - Linux kernel 2.6.23.9 - DG33BU motherboard. For example, on an Asus M2NPV-VM with the same cards/alsa/kernel, the bug does not happen. - And the most strange condition: send non-null samples to channels 1&3 or channels 2&4 of the daughter board (ie channels 13&15 or 14&16 at 48kHz). The bug does not happen if you send sound on channels 1&2 only, or 1 only, or 3 only for example, but it happens if you send on both 1&3. By analyzing what happens, I found the reason for the xrun. In the conditions of the bug, from time to time (say about 1 time over 1000), the memory space of the card (called "iobase" in the hdsp driver) is incorrectly read. The value 0xFFFFFFFF is read for any register instead of the correct values. I confirmed this behavior with the following experiment: - I wrote a programm permanently reading the memory space of the card, in an active loop, by mmap()ing /sys/bus/pci/devices/<my_card>/resource0, reading the first word, and reporting when 0xFFFFFFFF is read (BTW, this first register corresponds to the card status, and the value 0xFFFFFFFF is not a possible status value of the card from what I saw --- or at least this value is highly unlikely) This program effectively reports some 0xff-reads when the above conditions are met (ie when non-null samples are sent to channel 1&3 of AO4S). As I said before, not all reads give 0xFF, only some of them (but of course, this is enough to get problems!). As soon as the above conditions are not met (for example, outputing true silence on the AO4S), the program stops reporting problems. - To make sure the bug is not due to an alsa driver/library or program bug, I wanted to write non-null samples directly to the shared memory region without using snd_pcm_writeX(). In order to do so, I modified the code displaying /proc/asound/card0/hdsp in order to get the physical address of the playback buffer in addition to the virtual address already given. And then, by mmap()ing /dev/mem at the right offset, I got the possibility to write non-null values while the card is running. The effect is the same: while the alsa driver writes zeros, if I "overload" the buffer with non-null values at offset corresponding to channels 1&3 of the AO4S, I get several bad reads with my watching program (of course, the bad reads stop quickly because the alsa drivers continue to write zeros thus overwriting my non-null values and thus the bug conditions are not met any longer) I don't know exactly who is to be blamed here, because the card with the same driver/kernel on another motherboard works perfectly, and, indeed, no 0xFFFFFFFF reads are detected in this case. Needless to say, I tried several DG33BU mobos, several slots, several re-plugs of the PCI card, with the same behaviour. Maybe this can be due to some kind of PCI conflict? Anyway, the RME card is the only PCI/PCI-express card on the mobo. For now, the fix is truly wonderful: in hdsp_read(), keep on reading while 0xFFFFFFFF is read... With a maximum of times of course (I set 10). If the "true" value is not 0xFFFFFFFF, then the true value will be read at least once and so returned (I never got 10 consecutive times 0xFFFFFFFF, even not 2), and if the "true" value is 0xFFFFFFFF, then the only side effect of the patch is that the register will be read 10 times instead of 1. I don't think this can do any harm in any case. Anyway, there should be a competition for the most awful bugfix, I'm sure this one could win :-) So this hack fixes the software-visible part of the problem, but the punctual disparition of the IO space of the card is still there. For example, I don't know what happens as far as writing is concerned! Maybe some writes to registers of the card are ignored? I did not really investigate this, but I checked that there are no glitches in the sound (by checking that a sine is never distorted during a whole night), contrary to what happened without the fix. I join the patch corresponding to this hack, I'm not sure if it should be included in the main tree!... However, it may be usefull (it is for me anyway), because you can't use the AO4S card without it. I would be interested to know if other people had the same problem with this motherboard, or another one. Of course, I would also be interested if someone has got an idea about the reason for this strange behaviour. I will send an email to RME about this problem. Best regards, Remy Bruno

1 0

[alsa-devel] handling of dmix and non-dmix in the same configuration file
by Mike Frysinger 22 Feb '08

22 Feb '08

on the Blackfin processor, we have an "almost-but-not-quite" AC97 interface (called a "SPORT"). the trouble is that we can hook up AC97 codecs, but in order to properly communicate with codecs, we need to repacketize the communication between the higher alsa layers and the actual SPORT. great ... but now we want to support dmix/dsnoop. the overhead for dmix vs non-dmix is such that we make it optional. if the user does not need software mixing, they can disable this overhead. this leads to two (incompatible) alsa config files (in this case, for the AD1980 codec). we're wondering if it's possible to have the right settings selected on the fly some how during runtime rather than having to select one file or the other. the dmix config: 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 ] } } } the non-dmix config: AD1980.pcm.default { type plug slave { pcm { type mmap_emul slave { pcm { type hw card 0 format S16_LE rate 48000 channels 2 sync_ptr_ioctl 1 } } } } } -mike

2 1

Re: [alsa-devel] HDA: no sound in headphone-out caused by commit f889fa91ad47e (2.6.25-rc1 regression)
by Rene Dohmen 22 Feb '08

22 Feb '08

Thnx for you quick reply De sound of the line out is working but quality is poor and the volume is very low. I didn't have it working with other versions of ALSA. Tested with 1.0.14, then started reading, lots of suggestions stated that the problem would be solved in 1.0.15, so i compiled 1.0.16. De device i 'm using is supplied by tangent ( http://www.tangent.com/t_all_in_one_LCDPC/vita_8000s.htm) its an all in one solution with touchscreen and speakers integrated in one small cabinet. alsa-info.sh script output uploaded: http://pastebin.ca/913865 I wasn't aware of the fact that I communicated directly: I found the discussion in google groups: and just hitted reply :-) So sorry for that. I now subscribed myself to the ML Regards, Rene On Fri, Feb 22, 2008 at 8:09 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Thu, 21 Feb 2008 18:56:30 -0800 (PST), > acidjunk(a)gmail.com wrote: > > > > Hi, > > I have exactly the same problem with my sound device: > > hda_codec: Unknown model for ALC262: trying auto-probe > > lspci output: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 > > Family) High Definition Audio Controller (rev 01) from BIOS. > > > > I'm using alsa 1.0.16 compiled form source under ubuntu gutsy. I tried > > hardy but is has the same results > > > > The headphone output is working (not very loud, but usable), but the > > front speakers are not working. Did the patches solve the speaker > > issue? and how do I apply them (I am new to the patching technique). > > Please don't mix up. It is a totally different problem since your > machine has another codec (and even 1.0.16 doesn't work). The patch > cannot solve your problem. Save your time. > > Has any version of ALSA driver ever worked properly? If yes, we can > check the difference. > > If not, we need to track down the problem more deeply. First of all, > give more detailed hardware information: what is your laptop (or > desktop) vendor, model, etc. Try different model options. And the > output of alsa-info.sh -- it's vital information. > > Last but not least, don't forget to add alsa-devel ML to Cc in your > next reply. This isn't a kind of thing to be communicated privately > at all. > > > thanks, > > Takashi >

1 0

[alsa-devel] Alsa Driver Puzzlement
by Benson Dong 22 Feb '08

22 Feb '08

Hi all, What's the relationship between alsa driver and low-level audio hardware driver such as uda1341 driver?Is Alsa driver a individual module or cooperating with uda1341 dirver?If they co-work ,what's the interface between them? Anyone who knows please tell me? Thank in advance! Best, Benson _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[alsa-devel] Device name and hw configuration
by Emmanuel Milou 21 Feb '08

21 Feb '08

Hi all, I am developping a softphone based on alsa. Does the device 'default' take in consideration the hardware parameters I set, like buffer size and period size? If not, should I play with the software parameters to improve the sound quality? Thanks Manu

1 0

[alsa-devel] [0/4] Convert PXA AC97 to use clock API
by Mark Brown 21 Feb '08

21 Feb '08

This patch series adds support for AC97 clocks on the PXA27xx and PXA3xx and converts both SoC and non-SoC pxa2xx-ac97 drivers to use the clock API rather than pxa_set_cken(). Since the ALSA changes require the changes to the ARM platform code it probably makes sense for them to be merged through ARM rather than ALSA.

4 15

[alsa-devel] [PATCH] hda: STAC927x analog mic
by Matthew Ranostay 21 Feb '08

21 Feb '08

Some laptops have a internal analog microphone that is not setup by the BIOS. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 6b53fa114faa pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Feb 19 15:03:57 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 17:11:51 2008 -0500 @@ -3629,6 +3629,8 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + /* configure the analog microphone on some laptops */ + stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ stac92xx_set_config_reg(codec, 0x0f, 0x02270110); /* correct the front input jack as a mic */

2 2

Re: [alsa-devel] Ответы для Linux Format
by Serge A. Suchkov 21 Feb '08

21 Feb '08

В сообщении от Thursday 21 February 2008 12:33:26 Andrei Borovsky написал(а): > > Хмм... а можно урезать эту информацию до минимума ? > > Это ведь всего лишь хобби и я не склонен его афишировать настолько > > публично :) > > >Тем более мой вклад в GSPCA весьма скромен на общем фоне. > > Как скажете. В принципе, все это, конечно, необязательно. Даже фотографию > поместить не хотите? :) Нет, спасибо :) > > > С уважением, > Андрей Боровский. -- Regards, Serge A. Suchkov -- A baby is an alimentary canal with a loud voice at one end and no responsibility at the other.

1 1

Re: [alsa-devel] [PATCH] Fix a compile warning under gcc-4.2.3.
by Clemens Ladisch 21 Feb '08

21 Feb '08

Joshua Roys wrote: > sound/core/init.c: In function ‘snd_card_disconnect’: > sound/core/init.c:307: warning: the address of ‘snd_shutdown_f_ops’ will always evaluate as ‘true’ > > Signed-off-by: Joshua Roys <roysjosh(a)gmail.com> > --- > sound/core/init.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/core/init.c b/sound/core/init.c > index 2cb7099..1afc58b 100644 > --- a/sound/core/init.c > +++ b/sound/core/init.c > @@ -304,7 +304,6 @@ int snd_card_disconnect(struct snd_card *card) > list_add(&mfile->shutdown_list, &shutdown_files); > spin_unlock(&shutdown_lock); > > - fops_get(&snd_shutdown_f_ops); > mfile->file->f_op = &snd_shutdown_f_ops; > > mfile = mfile->next; This change may get rid of the warning, but is has the slight disadvantage that the kernel will blow up if you try to unload the sound modules before all device files have been closed. If you want to get rid of this warning, tell the compiler that fops_get knows what it does; or try making fops_get an inline function instead of a macro. Regards, Clemens

2 1

[alsa-devel] [PATCH] hda: 92HDxxxx PCI Quirks
by Matthew Ranostay 21 Feb '08

21 Feb '08

Added PCI_QUIRKS for laptop that have the 92HDxxx family of codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 6b53fa114faa pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Feb 19 15:03:57 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 17:18:15 2008 -0500 @@ -64,11 +64,14 @@ enum { enum { STAC_92HD73XX_REF, + STAC_DELL_M6_5ST, STAC_92HD73XX_MODELS }; enum { STAC_92HD71BXX_REF, + STAC_DELL_M4_5ST, + STAC_DELL_M4_6ST, STAC_92HD71BXX_MODELS }; @@ -334,10 +337,10 @@ static hda_nid_t stac922x_pin_nids[10] = 0x0f, 0x10, 0x11, 0x15, 0x1b, }; -static hda_nid_t stac92hd73xx_pin_nids[12] = { +static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x22 + 0x14, 0x1e, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -1220,24 +1223,48 @@ static struct snd_pci_quirk stac925x_cfg {} /* terminator */ }; -static unsigned int ref92hd73xx_pin_configs[12] = { +static unsigned int ref92hd73xx_pin_configs[13] = { 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, 0x0181302e, 0x01014010, 0x01014020, 0x01014030, 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, + 0x01452050, +}; + +static unsigned int dell_m6_5st_pin_configs[13] = { + 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, + 0x03a11020, 0x03011050, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, }; static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { - [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_DELL_M6_5ST] = dell_m6_5st_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = "ref", + [STAC_DELL_M6_5ST] = "dell-m6-5stack", }; static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD73XX_REF), + "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0256, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0257, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025e, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025f, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271, + "unknown Dell", STAC_DELL_M6_5ST), {} /* terminator */ }; @@ -1247,18 +1274,56 @@ static unsigned int ref92hd71bxx_pin_con 0x90a000f0, 0x01452050, }; +static unsigned int dell_m4_5st_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, +}; + +static unsigned int dell_m4_6st_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, + 0x40f000f0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, + [STAC_DELL_M4_5ST] = dell_m4_5st_pin_configs, + [STAC_DELL_M4_6ST] = dell_m4_6st_pin_configs, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", + [STAC_DELL_M4_5ST] = "dell-m4-5stack", + [STAC_DELL_M4_6ST] = "dell-m4-6stack", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0250, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024f, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024d, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0251, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0277, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0263, + "unknown Dell", STAC_DELL_M4_6ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0265, + "unknown Dell", STAC_DELL_M4_6ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0262, + "unknown Dell", STAC_DELL_M4_6ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, + "unknown Dell", STAC_DELL_M4_6ST), {} /* terminator */ }; @@ -3355,12 +3420,38 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); - spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; + + switch (spec->board_config) { + case STAC_DELL_M6_5ST: + switch (codec->subsystem_id) { + case 0x1028025e: /* Analog Mics */ + case 0x1028025f: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + spec->num_dmics = 0; + break; + case 0x10280254: /* Digital Mics */ + case 0x10280255: + case 0x10280271: + case 0x10280272: + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + case 0x10280256: /* Both */ + case 0x10280057: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + } + break; + default: + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + } spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids;

2 2

[alsa-devel] [PATCH] hda: STAC927x power management
by Matthew Ranostay 21 Feb '08

21 Feb '08

Fix issue on STAC927x codecs that first DAC was getting powered down even if was being used. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 30701aaf404c pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Feb 20 12:13:29 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 14:16:59 2008 -0700 @@ -2893,7 +2893,8 @@ static void stac92xx_power_down(struct h /* power down inactive DACs */ hda_nid_t *dac; for (dac = spec->dac_list; *dac; dac++) - if (!is_in_dac_nids(spec, *dac)) + if (!is_in_dac_nids(spec, *dac) && + spec->multiout.hp_nid != *dac) snd_hda_codec_write_cache(codec, *dac, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); }

2 1

Re: [alsa-devel] Alsa-devel Digest, Vol 12, Issue 70
by Anders Rønningen 21 Feb '08

21 Feb '08

I'll do the tests when I get home from work. Just wanted to check that I understood the instructions. The problem is like you said. The left HP and the internal speakers never give any sound, regardless of whether the right HP is connected. You want me to: 1. remove the mentioned line from alsa-kernel/pci/hda/patch_sigmatel.c. 2. configure and build the driver with full debug 3. reboot, run alsa-info.sh and upload the output Is that right? > No difference in the output but there must be a difference in the codec setting. You mean before and after I remove the line from patch_sigmatel.c? >What is the original configuration of pin 0xf? How can I know/find out this? - Anders On Feb 20, 2008 12:00 PM, <alsa-devel-request(a)alsa-project.org> wrote: > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: Problem with DELL XPS M1530 (Jiang zhe) > 2. Re: Problem with DELL XPS M1530 (Takashi Iwai) > 3. Re: Problem with DELL XPS M1530 (Takashi Iwai) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 20 Feb 2008 17:49:00 +0800 > From: Jiang zhe <zhe.jiang(a)intel.com> > Subject: Re: [alsa-devel] Problem with DELL XPS M1530 > To: Anders R?nningen <andersronningen(a)gmail.com> > Cc: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org > Message-ID: <1203500940.3006.4.camel(a)localhost.localdomain> > Content-Type: text/plain; charset=utf-8 > > On Wed, 2008-02-20 at 02:25 +0800, Anders R?nningen wrote: > > Hi Takashi. > > > > Some dmesg output: > > > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2994: > > autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2998: > > speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3002: > > hp_outs=1 (0xa/0xf/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3003: > > mono: mono_out=0x0 > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3011: > > inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 > > [ 14.196000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2224: > > dac_nids=2 (0x5/0x4/0x0/0x0/0x0) > > > > Unmuted and adjusted Surround volume, but it made no difference. > > > > - Anders > > > It's weird that 0xf is in both the line_outs and hp_outs. > And the hp_outs=1 when there are two items. > > What is the original configuration of pin 0xf? > Please try to get rid of this line from the > alsa-kernel/pci/hda/patch_sigmatel.c. > > SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), > > Rebuild and install the driver. > Please give the output of "cat /proc/asound/card0/codec#0". > > Thanks! > > > Takashi Iwai wrote: > > > At Tue, 19 Feb 2008 17:24:04 +0100, > > > Anders R?nningen wrote: > > > > > >> Hi. I am the mentioned user with this problem. Just thought I'd > > give you > > >> a little more info on this problem. > > >> > > >> What wishie described was correct, except it's the internal > > speakers and > > >> the /left/ hp-jack that stopped working, while the right one is > > still > > >> ok. Like he wrote, the reason we started this debugging was to get > > mic > > >> sound on my system. The thing I think is worth mentioning is that > > with > > >> alsa 1.0.14 on an out-of-the box Ubuntu Gutsy, both hp-jacks and > > the > > >> internal speakers worked for me. People in #alsa told me that the > > module > > >> was not loaded correctly, but whatever was loaded _almost_ worked > > well :-) > > >> > > > > > > OK, please build the driver with --with-debug=full option and show > > the > > > kernel messages such as "autoconfig: ..." and "dac_nids...", etc. > > > This contains the information that the pin configuration parser > > uses. > > > > > > Also, unmute and adjust "Surround" volume. > > > > > > > > > Takashi > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > > > ------------------------------ > > Message: 2 > Date: Wed, 20 Feb 2008 11:50:44 +0100 > From: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] Problem with DELL XPS M1530 > To: Jiang zhe <zhe.jiang(a)intel.com> > Cc: alsa-devel(a)alsa-project.org, Anders R?nningen > <andersronningen(a)gmail.com> > Message-ID: <s5hy79fsxyz.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=ISO-8859-1 > > At Wed, 20 Feb 2008 17:49:00 +0800, > Jiang zhe wrote: > > > > On Wed, 2008-02-20 at 02:25 +0800, Anders R?nningen wrote: > > > Hi Takashi. > > > > > > Some dmesg output: > > > > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2994: > > > autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2998: > > > speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3002: > > > hp_outs=1 (0xa/0xf/0x0/0x0/0x0) > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3003: > > > mono: mono_out=0x0 > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3011: > > > inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 > > > [ 14.196000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2224: > > > dac_nids=2 (0x5/0x4/0x0/0x0/0x0) > > > > > > Unmuted and adjusted Surround volume, but it made no difference. > > > > > > - Anders > > > > > It's weird that 0xf is in both the line_outs and hp_outs. > > And the hp_outs=1 when there are two items. > > It's due to a fix for an ALC268 laptop that has also two HPs but one > is actually a line-out. Note that hp_outs=1 in the print above. The > 0x0f in hp_outs is simply a garbage. Maybe that patch broke the > double HPs with STAC9228... > > > Takashi > > > ------------------------------ > > Message: 3 > Date: Wed, 20 Feb 2008 11:52:43 +0100 > From: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] Problem with DELL XPS M1530 > To: Anders R?nningen <andersronningen(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <s5hwsozsxvo.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=ISO-8859-1 > > At Tue, 19 Feb 2008 19:25:24 +0100, > Anders R?nningen wrote: > > > > Hi Takashi. > > > > Some dmesg output: > > > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2994: > > autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2998: > > speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3002: > > hp_outs=1 (0xa/0xf/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3003: > > mono: mono_out=0x0 > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3011: > > inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 > > [ 14.196000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2224: > > dac_nids=2 (0x5/0x4/0x0/0x0/0x0) > > > > Unmuted and adjusted Surround volume, but it made no difference. > > Please upload the alsa-info.sh output again. No difference in the > output but there must be a difference in the codec setting. > > Also, just to make the situation clear: > the problem is that the speaker and the rear HP don't give sound > output even if you *don't* plug in the front HP, right? > > > Takashi > > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 12, Issue 70 > ****************************************** > -- Mvh, Anders

3 5

Re: [alsa-devel] Problem with DELL XPS M1530
by Anders Rønningen 21 Feb '08

21 Feb '08

Hi. I am the mentioned user with this problem. Just thought I'd give you a little more info on this problem. What wishie described was correct, except it's the internal speakers and the /left/ hp-jack that stopped working, while the right one is still ok. Like he wrote, the reason we started this debugging was to get mic sound on my system. The thing I think is worth mentioning is that with alsa 1.0.14 on an out-of-the box Ubuntu Gutsy, both hp-jacks and the internal speakers worked for me. People in #alsa told me that the module was not loaded correctly, but whatever was loaded _almost_ worked well :-) - Anders

3 7

[alsa-devel] snd-hda-intel AD1981 - the microphone doesn't work
by Robert Gonciarz 20 Feb '08

20 Feb '08

Hi, I'm trying to activate a microphone (I have just external one) in my HP NX7300 (RU374ES) notebook (it has two jacks without internal mic) for six months. Alsamixer says that it's AD1981 chipset. The sound works great execept two things: - it's too silent when I set almost the maximum level (if you compare it with Windows XP driver) - but it's not so bad. It is said that a tothem can change some hidden settings to make it louder. - the microphone doesn't work even though I setted it in alsamiser (the settings: <capture tab> the "Mic" is red, "Mic boss" is setted with 67/67, "Capture" 80/80, Mix is blue and "-----", "Digital" 62/62, "Docking-" is blue and "----", "Internal" 67/67. The mic doesn't capture any voice at all. I was trying to load the module with different names: default, hp and toshiba works but the mic still doesn't work. lspci | grep Audio 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) lspci -nv 00:1b.0 0403: 8086:27d8 (rev 01) Subsystem: 103c:30a2 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at f4580000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel It was tested with snd-hda-intel module from the lastest sources from your website: alsa-driver-1.0.16rc2. I'm using 2.6.24.1 kernel. At the begining I tried to compile the kernel with a module which was included to it (it was working without a mic) and then I compiled the external module from your web. It was tested with many other kernels. Can you check it please? If you need more information just email me back. Robert

4 12

[alsa-devel] Regarding pointer callback
by Pharaoh . 20 Feb '08

20 Feb '08

Hi, In Writing an ALSA driver, it is written that pointer callback is called usually from the buffer-update routine in the pcm middle layer, which is invoked when snd_pcm_period_elapsed() is called in the interrupt routine. Then the pcm middle layer updates the position and calculates the available space, and wakes up the sleeping poll threads, etc. I know from a kernel space ALSA driver, the pointer callback is called after the driver calls snd_pcm_period_elapsed() from the interrupt handler or after a period is consumed. How pointer function of a user space plugin is called? I have a user space pcm external plugin which updates the buffer counter after every period is consumed, but when will it be called? How does the ALSA lib know when to call it since there is no snd_pcm_period_elapsed() in user space. -Phraoh.

1 0

[alsa-devel] Capture and playabck together
by Emmanuel Milou 20 Feb '08

20 Feb '08

Hi all, I am developping an open-source C++ softphone based on Alsa. We used to use Portaudio but we recently decided to switch to Alsa. I am quite unexperimented in Alsa dev and I got some difficulties to make it work fine. My capture and playback devices are plughw:0,0. I am using read/write transfers. When I disable the capture, playback works fine; when I disable the playback, capture works fine. The problems occurs when I used them together. If I set hardware parameters for both, the playback sound is awful and there is no capture, and finally core dumped. If I set hardware parameters for the playback stream only, the playback works fine, but no capture, I just get an echo with an input/ouput error. Where do you think the problem comes from? Isn't it possible to set hardware configuration for playback and capture? Is there a hardware ring buffer for each stream ? Thanks for your help Manu

1 0

[alsa-devel] [PATCH 0/2] TLV320AIC3X MCLK and rate updates
by Jarkko Nikula 20 Feb '08

20 Feb '08

Hi Two patches to TLV320AIC3X SoC codec driver. First one fixes typo (?) for 64 kHz sample rate and second one adds support to 12 MHz MCLK used e.g. in Nokia N810 Internet Tablet. Patches are generated from 2.6.25-rc2-omap1 tree but they apply to alsa git and alsa-kernel hg trees as well. -- Jarkko

2 3

[alsa-devel] Merging hg sources into kernel tree
by Andrew Boettcher 20 Feb '08

20 Feb '08

Is there a script to merge the hg sources into a kernel tree? If not do I need to worry about any changes other than the include/ directory? I have another module (cx88-alsa) that is in v4l but needs the newer alsa tree or I get "cx88_alsa: disagrees about version of symbol xxx", so I have to merge the sources for now. Andrew Boettcher

3 3

[alsa-devel] Moving X Window windows causes audio stuttering
by Timur Tabi 20 Feb '08

20 Feb '08

On our system, using mplayer to play a movie that has 44.1KHz audio (hardware only supports 48KHz) can cause audio stuttering if we're using OSS emulation, but not if we're using native ALSA. Is this a known problem? Does OSS emulation hog the CPU? -- Timur Tabi Linux kernel developer at Freescale

2 1

Re: [alsa-devel] [PATCH] Driver for Emagic Audiowerk 2
by Takashi Iwai 20 Feb '08

20 Feb '08

At Tue, 19 Feb 2008 00:38:54 +0100, Cédric Brégardis wrote: > > Signed-off-by: Cedric Bregardis <cedric.bregardis(a)free.fr> > > Signed-off-by: Jean-Christian Hassler <jhassler(a)free.fr> Could you avoid HTML mail? I cannot get the patch properly. If embedding text is difficult, rather use an attachment. thanks, Takashi

2 4

[alsa-devel] Ends up with write error
by Ning Zhang 20 Feb '08

20 Feb '08

Hi all, I am writing a ALSA driver for my ARM 926 platform. The ARM communicates with the DSP by accessing a 128-byte shared memory. I don't use IRQ mechanism.(In fact, I am not sure the platform has this mechanism) Here is some code slices: static struct snd_pcm_hardware snd_arch_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000), .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 128, .period_bytes_min = 2, .period_bytes_max = 128, .periods_min = 1, .periods_max = 64, }; static snd_pcm_uframes_t snd_arch_pcm_pointer(struct snd_pcm_substream *substream) { … /* * get the current hardware pointer */ current_ptr = arch_fw_read_dsp_reg(chip, AFE_RWADDR) & AFE_RD_ADDR; return bytes_to_frames(runtime, current_ptr); } static int snd_arch_pcm_playback_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) { …… count = frames_to_bytes(runtime, count); pos = frames_to_bytes(runtime, pos); chip->bufpos = pos; …. spin_lock(chip->reg_lock); if ( count > 0 && count <= MAX_BUF_SIZE ) { if ( copy_from_user(chip->bufptr, src, count) ) { snd_printk(KERN_ERR "can not copy user data!\n"); return -EFAULT; } if ( arch_fill_rx_buf(chip, (signed short*)(chip->bufptr), count) < 0 ) { snd_printk(KERN_ERR "fill rx buffer failed!\n"); return -1; } …………… } spin_unlock(chip->reg_lock); return 0; } The problem is my driver can make sounds, but ends up a write error. After much debugging, I think there might be some problem in the interworking between middle-level driver and the device driver. I found the runtime->control->avail_min=32, and the runtime->buffer_size and runtime->period_size are both 32. So at first middle-level driver call snd_pcm_lib_write1 transfers 32 frames(128 bytes) to the device driver's copy callback function and trigger function. Then when middle-level driver wants to transfer another 32 frames, it calls the pointer callback function and gets the position, because at this time the whole bytes has not been processed completely, so the pointer function's return value must less than 32, and the avail must less than runtime->control_->avail_min, and goes to this flow and ends up a write error. _______________________________________________________________________________ if (((avail < runtime->control->avail_min && size > avail) || (size >= runtime->xfer_align && avail < runtime->xfer_align))) { ……………... long tout; init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); while (1) { if (signal_pending(current)) { state = SIGNALED; break; } set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_unlock_irq(substream); tout = schedule_timeout(10 * HZ); snd_pcm_stream_lock_irq(substream); if (tout == 0) { if (runtime->status->state != SNDRV_PCM_STATE_PREPARED && runtime->status->state != SNDRV_PCM_STATE_PAUSED) { state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; break; } } switch (runtime->status->state) { ………….. : state = ERROR; goto _end_loop; default: break; } avail = snd_pcm_playback_avail(runtime); if (avail >= runtime->control->avail_min) { state = READY; break; } } _end_loop: remove_wait_queue(&runtime->sleep, &wait); switch (state) { ……. case EXPIRED: snd_printd("playback write error (DMA or IRQ trouble?)\n"); err = -EIO; goto _end_unlock; ……… } } This code is a part of snd_pcm_lib_write1() in sound/core/pcm_lib.c I think that I misunderstood the mechanism of middle-level driver. Could you please correct me? What should the pointer callback function return? What's it for? How comes the runtime->control->avail_min? Because I don't use the IRQ, so should I use the snd_pcm_period_elapsed function? And where? ALSA Version is 1.0.14rc1 and I use alsa-lib-1.0.15. Can someone comment on this and give me some hints to solve this problem? Thanks in advance. BRs, David

1 0

[alsa-devel] ALSA MIDI interface - plugin
by Pharaoh . 20 Feb '08

20 Feb '08

Hi I am writing an user space ALSA i/o type PCM plugin for my board. I chose this rather unconventional approach for supporting ALSA apps since the the the driver which communicates with DSP using shared memory is in userspace and for some constraints the driver communication driver will always be in userspace. My audio plugin uses calls of the coomunication driver and talks to DSP. I already have a working plugin in place. I use it like following, #aplay -D my_plugin test.wav and it works. Now, I have to support rawmidi interface also. Is it possible to write a userspace plugin for making MIDI work? Where can I get sample code etc? -pharaoh.

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Re: [alsa-devel] Problem with setting period time and buffer time.
by Clemens Ladisch 20 Feb '08

20 Feb '08

Ignacy Kasperowicz wrote: > But what about buffer_time, why can't I set this time ? The allowed range of values is hardware dependent, like the period time. Use snd_pcm_hw_params_set_buffer_time_near(). > Anyway is there a method to play my wave for a specified short time? Just write the data to the device. What is the problem? Regards, Clemens

2 4

[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 19 Feb '08

19 Feb '08

hi all i'm new to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't -) low volume -) inoperating headphone jack -) not working mic -) no subwoofer -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux alsa 1.0.15 drivers (compiled with debian-way: module-assistant a-i alsa-source) i've already tried all the option i could think about auto (default) toshiba 3stacks 3stacks-6ch 3stacks-6ch-dig neither one has better/worst behavior I've checked the module has toke those option by cat /sys/module/snd_hda_intel/parameters/model anyway I've read the ALSA-Configuration.txt shipped with kernel and for Codec ALC268 it only has 4 options ALC268 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops auto auto-config reading BIOS (default) I've googled a bit and i found out that the Toshiba A205 has completely different problem.. the sound doesn't work with it.. reading the changelog from 1.0.14 to 1.0.15 (alsa) it say: Added the proper model=toshiba for Toshiba A305 with ALC268 codec. so i think my audio card is an unknown model the laptop is brand new and i only found this model in Italy anyway i think that the same audio card is shipped with the Toshiba Satellite x205-S7483 witch is an american laptop some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659 please ask me if you need any more info or want me to do some test thanks you all for your work ps: i've subscribed to vger mailing list, what's the right mailing list for alsa-devel? -- Daniele Segato

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[alsa-devel] Maya44 revised patch
by Rainer Zimmermann 19 Feb '08

19 Feb '08

This is the revised version of the ESI Maya44 patch. changes are mainly coding style, some cleanups and using a patch file for ice1724.c . The added code now (mostly) passes checkpatch.pl, except for some "#if 0"'s which I'd leave in for the development phase. Also, I changed the rate setting logic, which (for now) allows all rates up to 192kHz even for capturing. As capturing actually only supports 96 kHz, while capturing & playback rates are linked, this is not really ok. I'd appreciate suggestions about how this could be handled properly, while still supporting playback up to 192 kHz... Again, feedback & testing is appreciated. For more information, see doc/README.maya44 . -Rainer -- Lightshed IT Services Löfflerstr. 27, 22765 Hamburg, Germany * mail(a)lightshed.de * www.lightshed.de

3 6

[alsa-devel] snd-hda-intel AD1981 - Sound volume low level
by Robert Gonciarz 19 Feb '08

19 Feb '08

Hi, I just want to ask you if you have heard about the problem with low sound volume level even though you set the alsamixer on the (almost) maximum volume level. The sound card: snd-hda-intel / AD1981 / ICH7 ) It's a big problem for me especially when I watch a movie with ac3 (DVD quality) decoded sound. The volume level is extrimly low in such situation (I mean the level you hear the sound). I have to connect my loudspeakers to a ampilifier and set it in very high level. I don't have such problem when I work with Windows XP. They are loud enough. That's alsa-info.sh result: http://pastebin.ca/908831 It is said that there are some preferences in the totem package (hidden in alsamixer) that can fix it and make alsa to remember higher sound level. I've noticed that problem with all kernels I tested (I've checked about 15-20 kernels from that range 2.6.13.x - 2.6.24.x) (I usually use vanillia kernel (from www.kernel.org) /with build alsa/). But I've tested the newest alsa-drivers: 1.0.16 with 2.6.24.2 kernel also and the result was the same. Could you chack that problem, please? Robert

2 1

Re: [alsa-devel] Problem with setting period time and buffer time.
by Ignacy Kasperowicz 19 Feb '08

19 Feb '08

Ok, so I misunderstood the idea of a period variable, many thanks for explanation Clemens. I've just read that buffer_time also isn't what I think it was ;) It's a size of the ring buffer of sound card and it's length depends on hardware, am I right? So the best way is to depend on default values of hardware? Anyway is there a method to play my wave for a specified short time? As I can see using timer interface is the only solution but still can't google anything useful :( Small example how to apply timers will be appreciated (I saw example timer.c from alsa-lib, but I'd prefer to read some explanation how the code works). Thanks from advance for any replays. -- ################################################ gry świetlicowe drogą do wyższej fazy przemian. gg:681548 jid: fik(a)jabbim.pl usecide.net ################################################

1 0

[alsa-devel] Problem with setting period time and buffer time.
by Ignacy Kasperowicz 19 Feb '08

19 Feb '08

Hello. I need to set proper period time for my application. What I need my code to do is to play sine wave with set frequency and for exact time of 1/1200 s (something about 834 us). When I try to set period time for 834 us I've got this error: Unable to set period time 834 for playback: Invalid argument . Honestly speaking I've got this error message no matter what value I try to set). The same situation take place when I try to set buffer time for something about 1000us (so one period could fit in my buffer). The purpose of this program is to generate sine wave for AFSK (audio frequency-shift keying). I want to play wave for period_time (834us) then if needed change frequency and play changed wave again for period_time with phase shifting. Wave generation with phase shifting for variable frequencies works, but not with period_time I need. Could anyone look at source and tell me why I can't set this values? I placed code with syntax highlighting at: http://pygments.org/demo/606/ . (Don't look at prepare_message function, it isn't used right now). Thanks from advice. Ignacy. -- ################################################ gry świetlicowe drogą do wyższej fazy przemian. gg:681548 jid: fik(a)jabbim.pl usecide.net ################################################

2 1

Re: [alsa-devel] installation steps for us-122l tascam driver
by belibem 18 Feb '08

18 Feb '08

Are there any news on the subject? What does it take to apply the patch? A packet for any major distribution would be greatly appreciated.

1 0

[alsa-devel] Cirrus Logic CS42L52 Low power codec
by Brian Austin 18 Feb '08

18 Feb '08

Hello, I have a codec I would like to submit. The Cirrus Logic CS42L52 Low Power codec. I have the codec code as well as the machine code for the Cirrus Logic EP9407A, but cannot submit that yet. Is it OK to just submit the codec code for now? This does not use the new API, so a hint as to which tree to make the diff from would help. Thanks, Brian Austin

4 10

[alsa-devel] Problem with DELL XPS M1530
by Travis Place 18 Feb '08

18 Feb '08

Hi all, Been helping a user on #alsa for a day or two now, with issues regarding recording from Mic on his Dell XPS M1530 laptop. He originally had alsa-driver 1.0.14, so i had him upgrade to 1.0.16. Now, his Mic works for recording, but he has lost playback on one of his HP jacks (the right one), and Internal laptop speakers. I had a brief look at the code in patch_sigmatel.c, to see what was going on.. But since im no code guru, i wasnt much help.. The output of alsa-info.sh is located at http://pastebin.ca/908579 I did notice some comments in the code about correcting HP and Mic jacks for certain models (atleast thats what I read it as). Any ideas would be appreciated. Travis (wishie). -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 2

[alsa-devel] unable to get alsa configured on Acer Aspire 1640Z
by Richard Lyons 18 Feb '08

18 Feb '08

Hi, I hope this is the right place to report this. Computer: laptop Acer Aspire 1640Z distro: Debian etch kernel: 2.6.17 lspci 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) lspci -nv 00:1b.0 0403: 8086:2668 (rev 04) Subsystem: 1025:009e Flags: bus master, fast devsel, latency 0, IRQ 177 Memory at d000c000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) I've tried model=acer and acer-aspire to no avail. I notice in dmesg after trying 'modprobe snd-hda-intel model=acer' this: hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... but alsamixer shows: Card: HDA Intel Chip: Realtek ALC883 at the top. There seems to be uncertainty about which model number, 882 or 883. I wonder if this is the source of the difficulty. HTH -- richard Lyons

3 4

[alsa-devel] Intel8x0 8 channel sound
by Andrew Boettcher 18 Feb '08

18 Feb '08

For some intel 8x0 cards there is 8 channel sound (such as the ALC850 chip on NFORCE 4 boards), and this has not been patched yet though the patch has been on the bug tracker. Here it is again updated for 1.0.16. There is also a patch for the configuration in alsa-lib for cards/NFORCE.conf. I have been patching my sources every time I update my kernel, I figured it was time for someone to actually submit this somewhere else. The patch was written by Martin Ellis.

2 6

[alsa-devel] alsa-info.sh script
by Travis Place 18 Feb '08

18 Feb '08

Is there any chance that i will be able to push updates for my script back to ALSA? Or is this to be a no-go ? Travis Place. (wishie) -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

3 3

[alsa-devel] [PATCH] Driver for Emagic Audiowerk 2
by Cédric Brégardis 18 Feb '08

18 Feb '08

Hi, We managed to write ALSA driver for Emagic Audiowerk 2 PCI sound card (based on Philips SAA7146 chip). You can find more information on https://gna.org/projects/aw2-alsa/. We think it's time to integrate the sources into the alsa/kernel tree. So, here is the patch for this driver. We hope this patch is usable. Regards, Jean-Christian Hassler and Cedric Bregardis Signed-off-by: Cedric Bregardis <cedric.bregardis(a)free.fr> Signed-off-by: Jean-Christian Hassler <jhassler(a)free.fr> --- sound/pci/Kconfig | 15 + sound/pci/Makefile | 1 + sound/pci/aw2/Makefile | 3 + sound/pci/aw2/SAA7146.h | 168 ++++++++++ sound/pci/aw2/aw2-alsa.c | 753 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/aw2/aw2-saa7146.c | 613 +++++++++++++++++++++++++++++++++++ sound/pci/aw2/aw2-saa7146.h | 125 +++++++ sound/pci/aw2/aw2_tsl.h | 116 +++++++ 8 files changed, 1794 insertions(+), 0 deletions(-) create mode 100644 sound/pci/aw2/Makefile create mode 100644 sound/pci/aw2/SAA7146.h create mode 100644 sound/pci/aw2/aw2-alsa.c create mode 100644 sound/pci/aw2/aw2-saa7146.c create mode 100644 sound/pci/aw2/aw2-saa7146.h create mode 100644 sound/pci/aw2/aw2_tsl.h diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 812085d..194394b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -122,6 +122,21 @@ config SND_AU8830 To compile this driver as a module, choose M here: the module will be called snd-au8830. +config SND_AW2 + tristate "Emagic Audiowerk 2" + depends on SND + help + Say Y here to include support for Emagic Audiowerk 2 soundcards. + + Supported features: Analog and SPDIF output. Analog or SPDIF input. + Note: Switch between analog and digital input does not always work. + It can produce continuous noise. The workaround is to switch again + (and again) between digital and analog input until it works. + + To compile this driver as a module, choose M here: the module + will be called snd-aw2. + + config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" depends on SND && EXPERIMENTAL diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 2d42fd2..85ef14b 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -58,6 +58,7 @@ obj-$(CONFIG_SND) += \ ac97/ \ ali5451/ \ au88x0/ \ + aw2/ \ ca0106/ \ cs46xx/ \ cs5535audio/ \ diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile new file mode 100644 index 0000000..842335d --- /dev/null +++ b/sound/pci/aw2/Makefile @@ -0,0 +1,3 @@ +snd-aw2-objs := aw2-alsa.o aw2-saa7146.o + +obj-$(CONFIG_SND_AW2) += snd-aw2.o diff --git a/sound/pci/aw2/SAA7146.h b/sound/pci/aw2/SAA7146.h new file mode 100644 index 0000000..1fbc66e --- /dev/null +++ b/sound/pci/aw2/SAA7146.h @@ -0,0 +1,168 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver 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 the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +/* SAA7146 registers */ +#define PCI_BT_A 0x4C +#define IICTFR 0x8C +#define IICSTA 0x90 +#define BaseA1_in 0x94 +#define ProtA1_in 0x98 +#define PageA1_in 0x9C +#define BaseA1_out 0xA0 +#define ProtA1_out 0xA4 +#define PageA1_out 0xA8 +#define BaseA2_in 0xAC +#define ProtA2_in 0xB0 +#define PageA2_in 0xB4 +#define BaseA2_out 0xB8 +#define ProtA2_out 0xBC +#define PageA2_out 0xC0 +#define IER 0xDC +#define GPIO_CTRL 0xE0 +#define ACON1 0xF4 +#define ACON2 0xF8 +#define MC1 0xFC +#define MC2 0x100 +#define ISR 0x10C +#define PSR 0x110 +#define SSR 0x114 +#define PCI_ADP1 0x12C +#define PCI_ADP2 0x130 +#define PCI_ADP3 0x134 +#define PCI_ADP4 0x138 +#define LEVEL_REP 0x140 +#define FB_BUFFER1 0x144 +#define FB_BUFFER2 0x148 +#define TSL1 0x180 +#define TSL2 0x1C0 + +#define ME (1UL << 11) +#define LIMIT (1UL << 4) +#define PV (1UL << 3) + +/* PSR/ISR/IER */ +#define PPEF (1UL << 31) +#define PABO (1UL << 30) +#define IIC_S (1UL << 17) +#define IIC_E (1UL << 16) +#define A2_in (1UL << 15) +#define A2_out (1UL << 14) +#define A1_in (1UL << 13) +#define A1_out (1UL << 12) +#define AFOU (1UL << 11) +#define PIN3 (1UL << 6) +#define PIN2 (1UL << 5) +#define PIN1 (1UL << 4) +#define PIN0 (1UL << 3) +#define ECS (1UL << 2) +#define EC3S (1UL << 1) +#define EC0S (1UL << 0) + +/* SSR */ +#define PRQ (1UL << 31) +#define PMA (1UL << 30) +#define IIC_EA (1UL << 21) +#define IIC_EW (1UL << 20) +#define IIC_ER (1UL << 19) +#define IIC_EL (1UL << 18) +#define IIC_EF (1UL << 17) +#define AF2_in (1UL << 10) +#define AF2_out (1UL << 9) +#define AF1_in (1UL << 8) +#define AF1_out (1UL << 7) +#define EC5S (1UL << 3) +#define EC4S (1UL << 2) +#define EC2S (1UL << 1) +#define EC1S (1UL << 0) + +/* PCI_BT_A */ +#define BurstA1_in (1UL << 26) +#define ThreshA1_in (1UL << 24) +#define BurstA1_out (1UL << 18) +#define ThreshA1_out (1UL << 16) +#define BurstA2_in (1UL << 10) +#define ThreshA2_in (1UL << 8) +#define BurstA2_out (1UL << 2) +#define ThreshA2_out (1UL << 0) + +/* MC1 */ +#define MRST_N (1UL << 15) +#define EAP (1UL << 9) +#define EI2C (1UL << 8) +#define TR_E_A2_OUT (1UL << 3) +#define TR_E_A2_IN (1UL << 2) +#define TR_E_A1_OUT (1UL << 1) +#define TR_E_A1_IN (1UL << 0) + +/* MC2 */ +#define UPLD_IIC (1UL << 0) + +/* ACON1 */ +#define AUDIO_MODE (1UL << 29) +#define MAXLEVEL (1UL << 22) +#define A1_SWAP (1UL << 21) +#define A2_SWAP (1UL << 20) +#define WS0_CTRL (1UL << 18) +#define WS0_SYNC (1UL << 16) +#define WS1_CTRL (1UL << 14) +#define WS1_SYNC (1UL << 12) +#define WS2_CTRL (1UL << 10) +#define WS2_SYNC (1UL << 8) +#define WS3_CTRL (1UL << 6) +#define WS3_SYNC (1UL << 4) +#define WS4_CTRL (1UL << 2) +#define WS4_SYNC (1UL << 0) + +/* ACON2 */ +#define A1_CLKSRC (1UL << 27) +#define A2_CLKSRC (1UL << 22) +#define INVERT_BCLK1 (1UL << 21) +#define INVERT_BCLK2 (1UL << 20) +#define BCLK1_OEN (1UL << 19) +#define BCLK2_OEN (1UL << 18) + +/* IICSTA */ +#define IICCC (1UL << 8) +#define ABORT (1UL << 7) +#define SPERR (1UL << 6) +#define APERR (1UL << 5) +#define DTERR (1UL << 4) +#define DRERR (1UL << 3) +#define AL (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +/* IICTFR */ +#define BYTE2 (1UL << 24) +#define BYTE1 (1UL << 16) +#define BYTE0 (1UL << 8) +#define ATRR2 (1UL << 6) +#define ATRR1 (1UL << 4) +#define ATRR0 (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +#define START 3 +#define CONT 2 +#define STOP 1 +#define NOP 0 diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c new file mode 100644 index 0000000..f954b0b --- /dev/null +++ b/sound/pci/aw2/aw2-alsa.c @@ -0,0 +1,753 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver 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 the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/control.h> +#include <linux/interrupt.h> +#include <asm/io.h> +#include "SAA7146.h" + +#include <linux/delay.h> + +#include "aw2-saa7146.h" + +MODULE_LICENSE("GPL"); +/********************************* + * DEFINES + ********************************/ +#define PCI_VENDOR_ID_SAA7146 0x1131 +#define PCI_DEVICE_ID_SAA7146 0x7146 + +#define CTL_ROUTE_ANALOG 0 +#define CTL_ROUTE_DIGITAL 1 + +/********************************* + * TYPEDEFS + ********************************/ + /* hardware definition */ +static struct snd_pcm_hardware snd_aw2_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 4, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_aw2_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +struct aw2_pcm_device_t { + struct snd_pcm *pcm; + unsigned int stream_number; + struct aw2_t *chip; +}; + +struct aw2_t { + struct snd_aw2_saa7146_t saa7146; + + struct snd_card *card; + + struct aw2_pcm_device_t device_playback[NB_STREAM_PLAYBACK]; + struct aw2_pcm_device_t device_capture[NB_STREAM_CAPTURE]; +}; + +/********************************* + * FUNCTION DECLARATIONS + ********************************/ +static int __init alsa_card_aw2_init(void); +static void __exit alsa_card_aw2_exit(void); +static int snd_aw2_dev_free(struct snd_device *device); +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2_t **rchip); +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id); +static void __devexit snd_aw2_remove(struct pci_dev *pci); +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd); +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream); +static int __devinit snd_aw2_new_pcm(struct aw2_t *chip); + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); + +/********************************* + * VARIABLES + ********************************/ +/* module parameters (see "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; + +static struct pci_device_id snd_aw2_ids[] = { + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + 0, 0, 0}, + {0} +}; + +MODULE_DEVICE_TABLE(pci, snd_aw2_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = "My Own Chip", + .id_table = snd_aw2_ids, + .probe = snd_aw2_probe, + .remove = __devexit_p(snd_aw2_remove), +}; + +/* operators for playback PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_playback_ops = { + .open = snd_aw2_pcm_playback_open, + .close = snd_aw2_pcm_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_playback, + .trigger = snd_aw2_pcm_trigger_playback, + .pointer = snd_aw2_pcm_pointer_playback, +}; + +/* operators for capture PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_capture_ops = { + .open = snd_aw2_pcm_capture_open, + .close = snd_aw2_pcm_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_capture, + .trigger = snd_aw2_pcm_trigger_capture, + .pointer = snd_aw2_pcm_pointer_capture, +}; + +static struct snd_kcontrol_new aw2_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Capture Route", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = snd_aw2_control_switch_capture_info, + .get = snd_aw2_control_switch_capture_get, + .put = snd_aw2_control_switch_capture_put +}; + +/********************************* + * FUNCTION IMPLEMENTATIONS + ********************************/ + +/* initialization of the module */ +static int __init alsa_card_aw2_init(void) +{ + printk(KERN_DEBUG "Load aw2 module\n"); + return pci_register_driver(&driver); +} + +/* clean up the module */ +static void __exit alsa_card_aw2_exit(void) +{ + printk(KERN_DEBUG "Unload aw2 module\n"); + pci_unregister_driver(&driver); +} + +module_init(alsa_card_aw2_init); +module_exit(alsa_card_aw2_exit); + +/* component-destructor + * (see "Management of Cards and Components") + */ +static int snd_aw2_dev_free(struct snd_device *device) +{ + struct aw2_t *chip = device->device_data; + return snd_aw2_saa7146_free(&chip->saa7146); +} + +/* chip-specific constructor + * (see "Management of Cards and Components") + */ +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2_t **rchip) +{ + struct aw2_t *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_aw2_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + pci_set_master(pci); + + /* check PCI availability (32bit DMA) */ + if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) || + (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) { + printk(KERN_ERR "Impossible to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + chip = kcalloc(1, sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->saa7146.pci = pci; + chip->saa7146.irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "Audiowerk2"); + if (err < 0) { + kfree(chip); + return err; + } + chip->saa7146.iobase_phys = pci_resource_start(pci, 0); + chip->saa7146.iobase_virt = + ioremap_nocache(chip->saa7146.iobase_phys, + pci_resource_len(pci, 0)); + + if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, + IRQF_SHARED, "Audiowerk2", (void *)chip)) { + printk(KERN_ERR "Cannot grab irq %d\n", pci->irq); + snd_aw2_saa7146_free(&chip->saa7146); + return -EBUSY; + } + chip->saa7146.irq = pci->irq; + + /* (2) initialization of the chip hardware */ + snd_aw2_saa7146_setup(&chip->saa7146); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_aw2_saa7146_free(&chip->saa7146); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + + /* Activate this portion to dump all the saa7146 registers : */ +#if 0 + snd_aw2_saa7146_dump_registers(&chip->saa7146); +#endif + + printk(KERN_INFO + "Audiowerk 2 sound card (saa7146 chipset) detected and " + "managed\n"); + return 0; +} + +/* constructor -- see "Constructor" sub-section */ +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct aw2_t *chip; + int err; + + /* (1) Continue if device is not enabled, else inc dev */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) Create card instance */ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + /* (3) Create main component */ + err = snd_aw2_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* initialize semaphore */ + init_MUTEX(&(chip->saa7146.sem)); + /* init spinlock */ + spin_lock_init(&chip->saa7146.reg_lock); + /* initialize wait queue */ + init_waitqueue_head(&chip->saa7146.wait_iic); + /* (4) Define driver ID and name string */ + strcpy(card->driver, "aw2"); + strcpy(card->shortname, "Audiowerk2"); + + sprintf(card->longname, "%s with SAA7146 at 0x%lx irq %i", + card->shortname, chip->saa7146.port, chip->saa7146.irq); + + /* (5) Create other components */ + snd_aw2_new_pcm(chip); + + /* (6) Register card instance */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) Set PCI driver data */ + pci_set_drvdata(pci, card); + + dev++; + return 0; +} + +/* destructor -- see "Destructor" sub-section */ +static void __devexit snd_aw2_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* open callback */ +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + printk(KERN_DEBUG "Playback_open \n"); + runtime->hw = snd_aw2_playback_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream) +{ + return 0; + +} + +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + printk(KERN_DEBUG "Capture_open \n"); + runtime->hw = snd_aw2_capture_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream) +{ + /* TODO: something to do ? */ + return 0; +} + + /* hw_params callback */ +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare callback for playback */ +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + down(&chip->saa7146.sem); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_playback_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + up(&chip->saa7146.sem); + + return 0; +} + +/* prepare callback for capture */ +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + down(&chip->saa7146.sem); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_capture_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + up(&chip->saa7146.sem); + + return 0; +} + +/* playback trigger callback */ +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + spin_lock(&chip->saa7146.reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->saa7146.reg_lock); + return status; +} + +/* capture trigger callback */ +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + spin_lock(&chip->saa7146.reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->saa7146.reg_lock); + return status; +} + +/* playback pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_pointer_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* capture pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_pointer_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* create a pcm device */ +static int __devinit snd_aw2_new_pcm(struct aw2_t *chip) +{ + struct snd_pcm *pcm_playback_ana; + struct snd_pcm *pcm_playback_num; + struct snd_pcm *pcm_capture; + int err = 0; + + /* Create new Alsa PCM device */ + + err = + snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0, + &pcm_playback_ana); + if (err >= 0) { + /* Creation ok */ + struct aw2_pcm_device_t *pcm_device = + &chip->device_playback[NUM_STREAM_PLAYBACK_ANA]; + + /* Set PCM device name */ + strcpy(pcm_playback_ana->name, "Analog playback"); + /* Associate private data to PCM device */ + pcm_playback_ana->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_ana, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_ana; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->saa7146.pci), + 64 * 1024, 64 * 1024); + if (err) { + printk(KERN_ERR + "snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + } + } + + err = + snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0, + &pcm_playback_num); + + if (err >= 0) { + /* Creation ok */ + struct aw2_pcm_device_t *pcm_device = + &chip->device_playback[NUM_STREAM_PLAYBACK_DIG]; + + /* Set PCM device name */ + strcpy(pcm_playback_num->name, "Digital playback"); + /* Associate private data to PCM device */ + pcm_playback_num->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_num, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_num; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_DIG; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->saa7146.pci), + 64 * 1024, 64 * 1024); + if (err) { + printk(KERN_ERR + "snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + } + } + + err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1, + &pcm_capture); + + if (err >= 0) { + /* Creation ok */ + struct aw2_pcm_device_t *pcm_device = + &chip->device_capture[NUM_STREAM_CAPTURE_ANA]; + + /* Set PCM device name */ + strcpy(pcm_capture->name, "Capture"); + /* Associate private data to PCM device */ + pcm_capture->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_capture, SNDRV_PCM_STREAM_CAPTURE, + &snd_aw2_capture_ops); + /* store PCM device */ + pcm_device->pcm = pcm_capture; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_CAPTURE_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm_capture, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->saa7146.pci), + 64 * 1024, 64 * 1024); + if (err) { + printk(KERN_ERR + "snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + } + } + + /* Create control */ + err = snd_ctl_add(chip->card, snd_ctl_new1(&aw2_control, chip)); + if (err < 0) { + printk(KERN_ERR "snd_ctl_add error (0x%X)\n", err); + } + return 0; +} + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { + "Analog", "Digital" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) { + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + } + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2_t *chip = snd_kcontrol_chip(kcontrol); + if (snd_aw2_saa7146_is_using_digital_input(&chip->saa7146)) { + ucontrol->value.enumerated.item[0] = CTL_ROUTE_DIGITAL; + } else { + ucontrol->value.enumerated.item[0] = CTL_ROUTE_ANALOG; + } + return 0; +} + +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2_t *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int is_disgital = + snd_aw2_saa7146_is_using_digital_input(&chip->saa7146); + + if (((ucontrol->value.integer.value[0] == CTL_ROUTE_DIGITAL) + && !is_disgital) + || ((ucontrol->value.integer.value[0] == CTL_ROUTE_ANALOG) + && is_disgital)) { + snd_aw2_saa7146_use_digital_input(&chip->saa7146, !is_disgital); + changed = 1; + } + return changed; +} diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c new file mode 100644 index 0000000..f3e180f --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.c @@ -0,0 +1,613 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver 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 the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define AW2_SAA7146_M + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include <linux/interrupt.h> +#include <asm/io.h> +#include "aw2_tsl.h" +#include "SAA7146.h" + +#include <linux/delay.h> + +#include <asm/system.h> + +#include "aw2-saa7146.h" + +#define WRITEREG(value, addr) do { writel((value), chip->iobase_virt + (addr));\ + mb(); } while (0) +#define READREG(addr) readl(chip->iobase_virt + (addr)) + +static struct snd_aw2_saa7146_cb_param + arr_substream_it_playback_cb[NB_STREAM_PLAYBACK]; +static struct snd_aw2_saa7146_cb_param + arr_substream_it_capture_cb[NB_STREAM_CAPTURE]; + +static int snd_aw2_saa7146_i2c_write(struct snd_aw2_saa7146_t *chip, + unsigned char address, const void *buf, + size_t count); +static int snd_aw2_saa7146_get_limit(int size); + +void snd_aw2_saa7146_dump_registers(struct snd_aw2_saa7146_t *chip) +{ + int i = 0; + for (i = 0; i <= 0x148; i += 4) { + printk(KERN_DEBUG "###DUMP 0x%03x: 0x%08x\n", i, READREG(i)); + } +} + +/* chip-specific destructor + * (see "PCI Resource Managements") + */ +int snd_aw2_saa7146_free(struct snd_aw2_saa7146_t *chip) +{ + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* release the irq */ + if (chip->irq >= 0) { + free_irq(chip->irq, (void *)chip); + } + /* release the i/o ports & memory */ + if (chip->iobase_virt) { + iounmap(chip->iobase_virt); + } + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; +} + +void snd_aw2_saa7146_setup(struct snd_aw2_saa7146_t *chip) +{ + /* set PCI burst/threshold + + Burst length definition + VALUE BURST LENGTH + 000 1 Dword + 001 2 Dwords + 010 4 Dwords + 011 8 Dwords + 100 16 Dwords + 101 32 Dwords + 110 64 Dwords + 111 128 Dwords + + Threshold definition + VALUE WRITE MODE READ MODE + 00 1 Dword of valid data 1 empty Dword + 01 4 Dwords of valid data 4 empty Dwords + 10 8 Dwords of valid data 8 empty Dwords + 11 16 Dwords of valid data 16 empty Dwords */ + + unsigned int acon2; + unsigned int acon1 = 0; + int i; + + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* enable audio interface */ +#ifdef __BIG_ENDIAN + acon1 |= A1_SWAP; + acon1 |= A2_SWAP; +#endif + /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */ + + /* At initialization WS1 and WS2 are disbaled (configured as input */ + acon1 |= 0 * WS1_CTRL; + acon1 |= 0 * WS2_CTRL; + + /* WS4 is not used. So it must not restart A2. + This is why it is configured as output (force to low) */ + acon1 |= 3 * WS4_CTRL; + + /* WS3_CTRL, WS3_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS3_CTRL; + + /* A1 and A2 are active and asynchronous */ + acon1 |= 3 * AUDIO_MODE; + WRITEREG(acon1, ACON1); + + /* The following comes from original windows driver. + It is needed to have a correct behavior of input and output + simultenously, but I don't know why ! */ + WRITEREG(3 * (BurstA1_in) + 3 * (ThreshA1_in) + + 3 * (BurstA1_out) + 3 * (ThreshA1_out) + + 3 * (BurstA2_out) + 3 * (ThreshA2_out), PCI_BT_A); + + /* enable audio port pins */ + WRITEREG((EAP << 16) | EAP, MC1); + + /* enable I2C */ + WRITEREG((EI2C << 16) | EI2C, MC1); + /* enable interrupts */ + WRITEREG(A1_out | A2_out | A1_in | IIC_S | IIC_E, IER); + + /* audio configuration */ + acon2 = A2_CLKSRC | BCLK1_OEN; + WRITEREG(acon2, ACON2); + + /* By default use analog input */ + snd_aw2_saa7146_use_digital_input(chip, 0); + + /* TSL setup */ + for (i = 0; i < 8; ++i) { + WRITEREG(tsl1[i], TSL1 + (i * 4)); + WRITEREG(tsl2[i], TSL2 + (i * 4)); + } + +} + +void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146_t *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA2_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA2_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA2_out); + + } else if (stream_number == 1) { + WRITEREG(dw_page, PageA1_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_out); + } else { + printk(KERN_ERR + "snd_aw2_saa7146_pcm_init_playback: Substream number is " + "not 0 or 1 -> not managed\n"); + } +} + +void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146_t *chip, + int stream_number, unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA1_in); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_in); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_in); + } else { + printk(KERN_ERR + "snd_aw2_saa7146_pcm_init_capture: Substream number is " + "not 0 -> not managed\n"); + } +} + +void snd_aw2_saa7146_define_it_playback_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_PLAYBACK) { + arr_substream_it_playback_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_playback_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_define_it_capture_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_CAPTURE) { + arr_substream_it_capture_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_capture_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + unsigned int acon1 = 0; + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + acon1 = READREG(ACON1); + if (stream_number == 0) { + WRITEREG((TR_E_A2_OUT << 16) | TR_E_A2_OUT, MC1); + + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS2_CTRL; + WRITEREG(acon1, ACON1); + + } else if (stream_number == 1) { + WRITEREG((TR_E_A1_OUT << 16) | TR_E_A1_OUT, MC1); + + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 |= 1 * WS1_CTRL; + WRITEREG(acon1, ACON1); + } else { + } + +} + +void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + unsigned int acon1 = 0; + acon1 = READREG(ACON1); + if (stream_number == 0) { + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 &= ~(3 * WS2_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A2_OUT << 16), MC1); + } else if (stream_number == 1) { + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 &= ~(3 * WS1_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A1_OUT << 16), MC1); + } else { + } +} + +void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + if (stream_number == 0) { + WRITEREG((TR_E_A1_IN << 16) | TR_E_A1_IN, MC1); + } else { + } +} + +void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + if (stream_number == 0) { + WRITEREG((TR_E_A1_IN << 16), MC1); + } else { + } + +} + +irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id) +{ + unsigned int isr; + unsigned int iicsta; + struct snd_aw2_saa7146_t *chip = dev_id; + + isr = READREG(ISR); + if (!isr) + return IRQ_NONE; + + WRITEREG(isr, ISR); + + if (isr & (IIC_S | IIC_E)) { + iicsta = READREG(IICSTA); + WRITEREG(0x100, IICSTA); + wake_up(&chip->wait_iic); + } else if (isr) { + } + + if (isr & A1_out) { + if (arr_substream_it_playback_cb[1].p_it_callback != NULL) { + arr_substream_it_playback_cb[1]. + p_it_callback(arr_substream_it_playback_cb[1]. + p_callback_param); + } + } + if (isr & A2_out) { + if (arr_substream_it_playback_cb[0].p_it_callback != NULL) { + arr_substream_it_playback_cb[0]. + p_it_callback(arr_substream_it_playback_cb[0]. + p_callback_param); + } + + } + if (isr & A1_in) { + if (arr_substream_it_capture_cb[0].p_it_callback != NULL) { + arr_substream_it_capture_cb[0]. + p_it_callback(arr_substream_it_capture_cb[0]. + p_callback_param); + } + } + return IRQ_HANDLED; +} + +unsigned int snd_aw2_saa7146_get_hw_pointer_playback(struct snd_aw2_saa7146_t + *chip, int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + long pci_adp = 0; + size_t ptr = 0; + + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP3); + ptr = pci_adp - (long)start_addr; + + if (ptr == buffer_size) { + ptr = 0; + } + } + if (stream_number == 1) { + pci_adp = READREG(PCI_ADP1); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) { + ptr = 0; + } + } else { + } + return ptr; +} + +unsigned int snd_aw2_saa7146_get_hw_pointer_capture(struct snd_aw2_saa7146_t + *chip, int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + size_t pci_adp = 0; + size_t ptr = 0; + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP2); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) { + ptr = 0; + } + } else { + } + return ptr; +} + +int snd_aw2_saa7146_set_sample_rate(struct snd_aw2_saa7146_t *chip, int srate) +{ + unsigned int S; + unsigned char DB[4]; + unsigned int iicsta = 0; + int err = 0; + /* check srate range */ + if ((srate < 30000) || (srate > 50000)) + return -EINVAL; + + /* PLL output is 256*fs, reference 1kHz */ + S = srate * 256 / 1000; + + /* LSB is at S2 */ + S <<= 2; + + DB[0] = (S << 1) & 0xff; + DB[1] = (S >> 7) & 0xff; + DB[2] = ((S >> 15) & 0xff) | 0x18; + DB[3] = 0; + + if (chip->iic) + return -EBUSY; + chip->iic = 1; + iicsta = READREG(IICSTA); + up(&chip->sem); + err = snd_aw2_saa7146_i2c_write(chip, 0xc4, DB, sizeof(DB)); + down(&chip->sem); + chip->iic = 0; + return err; + +} + +void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146_t *chip, + int use_digital) +{ + /* FIXME: switch between analog and digital input does not always work. + It can produce a kind of white noise. It seams that received data + are inverted sometime (endian inversion). Why ? I don't know, maybe + a problem of synchronization... However for the time being I have + not found the problem. Workaround: switch again (and again) between + digital and analog input until it works. */ + if (use_digital) { + WRITEREG(0x40, GPIO_CTRL); + } else { + WRITEREG(0x50, GPIO_CTRL); + } +} + +int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146_t *chip) +{ + unsigned int reg_val = READREG(GPIO_CTRL); + if ((reg_val & 0xFF) == 0x40) { + return 1; + } else { + return 0; + } +} + +static int snd_aw2_saa7146_i2c_write(struct snd_aw2_saa7146_t *chip, + unsigned char address, const void *buf, + size_t count) +{ + unsigned int iicsta = 0; + unsigned int iictfr = 0; + int i, b; + + /* We only keep low significant bits i.e. without clock setting */ + iicsta = (READREG(IICSTA) & 0xFF); + + /* Clock setting */ + iicsta = 1 * IICCC; + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + if ((iicsta & BUSY) || (iicsta & ERR)) { /* error or busy */ + WRITEREG(iicsta | ABORT, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + /* clear error bits (2x) */ + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + } + + iictfr = address * BYTE2; + iictfr |= START * ATRR2; + + for (i = 0; i <= count; ++i) { + b = i % 3; + if ((b == 0) || (i == count)) { + wait_queue_t wait; + init_waitqueue_entry(&wait, current); + + add_wait_queue(&chip->wait_iic, &wait); + set_current_state(TASK_INTERRUPTIBLE); + + WRITEREG(iictfr, IICTFR); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + + schedule(); + set_current_state(TASK_RUNNING); + remove_wait_queue(&chip->wait_iic, &wait); + + if (signal_pending(current)) { + return -ERESTARTSYS; + } + + iicsta = READREG(IICSTA); + if (iicsta & ERR) { + return -EIO; + } + if (i == count) + break; + iictfr = 0; + } + switch (b) { + case 0: + iictfr |= BYTE2 * ((char *)buf)[i]; + iictfr |= ATRR2 * ((count - i == 1) ? STOP : CONT); + break; + case 1: + iictfr |= BYTE1 * ((char *)buf)[i]; + iictfr |= ATRR1 * ((count - i == 1) ? STOP : CONT); + break; + case 2: + iictfr |= BYTE0 * ((char *)buf)[i]; + iictfr |= ATRR0 * ((count - i == 1) ? STOP : CONT); + break; + } + } + return 0; +} + +static int snd_aw2_saa7146_get_limit(int size) +{ + int limitsize = 32; + int limit = 0; + while (limitsize < size) { + limitsize *= 2; + limit++; + } + return limit; +} diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h new file mode 100644 index 0000000..93e4883 --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.h @@ -0,0 +1,125 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver 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 the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#ifndef AW2_SAA7146 +#define AW2_SAA7146 + +#ifdef AW2_SAA7146_M +#define PUBLIC +#else +#define PUBLIC extern +#endif + +#define NB_STREAM_PLAYBACK 2 +#define NB_STREAM_CAPTURE 1 + +#define NUM_STREAM_PLAYBACK_ANA 0 +#define NUM_STREAM_PLAYBACK_DIG 1 + +#define NUM_STREAM_CAPTURE_ANA 0 + +typedef void (*snd_aw2_saa7146_it_cb) (void *); + +struct snd_aw2_saa7146_cb_param { + snd_aw2_saa7146_it_cb p_it_callback; + void *p_callback_param; +}; + +/* definition of the chip-specific record */ + +struct snd_aw2_saa7146_t { + struct pci_dev *pci; + + unsigned long iobase_phys; + void __iomem *iobase_virt; + unsigned long port; + int irq; + + struct semaphore sem; + spinlock_t reg_lock; + int iic; + wait_queue_head_t wait_iic; +}; + +PUBLIC void snd_aw2_saa7146_dump_registers(struct snd_aw2_saa7146_t *chip); + +PUBLIC int snd_aw2_saa7146_free(struct snd_aw2_saa7146_t *chip); +PUBLIC void snd_aw2_saa7146_setup(struct snd_aw2_saa7146_t *chip); +PUBLIC void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146_t *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +PUBLIC void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146_t *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +PUBLIC void snd_aw2_saa7146_define_it_playback_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +PUBLIC void snd_aw2_saa7146_define_it_capture_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +PUBLIC void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146_t + *chip, int stream_number); +PUBLIC void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146_t + *chip, int stream_number); + +PUBLIC void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146_t + *chip, + int stream_number); +PUBLIC void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146_t + *chip, int stream_number); + +PUBLIC irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id); +PUBLIC unsigned int snd_aw2_saa7146_get_hw_pointer_playback(struct + snd_aw2_saa7146_t + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); +PUBLIC unsigned int snd_aw2_saa7146_get_hw_pointer_capture(struct + snd_aw2_saa7146_t + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); +PUBLIC int snd_aw2_saa7146_set_sample_rate(struct snd_aw2_saa7146_t *chip, + int srate); + +PUBLIC void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146_t *chip, + int use_digital); + +PUBLIC int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146_t + *chip); + +#endif diff --git a/sound/pci/aw2/aw2_tsl.h b/sound/pci/aw2/aw2_tsl.h new file mode 100644 index 0000000..af2edbf --- /dev/null +++ b/sound/pci/aw2/aw2_tsl.h @@ -0,0 +1,116 @@ +/***************************************************************************** + * + * Copyright 1998 Emagic Soft- und Hardware GmbH + * Copyright 2002 Martijn Sipkema + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver 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 the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define TSL_WS0 (1UL << 31) +#define TSL_WS1 (1UL << 30) +#define TSL_WS2 (1UL << 29) +#define TSL_WS3 (1UL << 28) +#define TSL_WS4 (1UL << 27) +#define TSL_DIS_A1 (1UL << 24) +#define TSL_SDW_A1 (1UL << 23) +#define TSL_SIB_A1 (1UL << 22) +#define TSL_SF_A1 (1UL << 21) +#define TSL_LF_A1 (1UL << 20) +#define TSL_BSEL_A1 (1UL << 17) +#define TSL_DOD_A1 (1UL << 15) +#define TSL_LOW_A1 (1UL << 14) +#define TSL_DIS_A2 (1UL << 11) +#define TSL_SDW_A2 (1UL << 10) +#define TSL_SIB_A2 (1UL << 9) +#define TSL_SF_A2 (1UL << 8) +#define TSL_LF_A2 (1UL << 7) +#define TSL_BSEL_A2 (1UL << 4) +#define TSL_DOD_A2 (1UL << 2) +#define TSL_LOW_A2 (1UL << 1) +#define TSL_EOS (1UL << 0) + + /* Audiowerk8 hardware setup: */ + /* WS0, SD4, TSL1 - Analog/ digital in */ + /* WS1, SD0, TSL1 - Analog out #1, digital out */ + /* WS2, SD2, TSL1 - Analog out #2 */ + /* WS3, SD1, TSL2 - Analog out #3 */ + /* WS4, SD3, TSL2 - Analog out #4 */ + + /* Audiowerk8 timing: */ + /* Timeslot: | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ... */ + + /* A1_INPUT: */ + /* SD4: <_ADC-L_>-------<_ADC-R_>-------< */ + /* WS0: _______________/---------------\_ */ + + /* A1_OUTPUT: */ + /* SD0: <_1-L___>-------<_1-R___>-------< */ + /* WS1: _______________/---------------\_ */ + /* SD2: >-------<_2-L___>-------<_2-R___> */ + /* WS2: -------\_______________/--------- */ + + /* A2_OUTPUT: */ + /* SD1: <_3-L___>-------<_3-R___>-------< */ + /* WS3: _______________/---------------\_ */ + /* SD3: >-------<_4-L___>-------<_4-R___> */ + /* WS4: -------\_______________/--------- */ + +#ifdef __BIG_ENDIAN + /* TODO: not yet implemented */ +#else /* */ + +static int tsl1[8] = { + 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1, + + 1 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 1 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 1 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 | + 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0 | TSL_SF_A1 | TSL_EOS, +}; + +static int tsl2[8] = { + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_LF_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS +}; + +#endif /* */ -- 1.5.2.5

2 1

[alsa-devel] [PATCH 1/1] Add __devinit macro to at73c213 sound driver probe functions
by Hans-Christian Egtvedt 18 Feb '08

18 Feb '08

This patch adds __devinit to the functions used when probing. Will also reduce the memory footprint a bit if CONFIG_HOTPLUG is not enabled. Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- sound/spi/at73c213.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index fee869b..83fa475 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -744,7 +744,7 @@ cleanup: /* * Device functions */ -static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) { /* * Continuous clock output. @@ -774,7 +774,7 @@ static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) return 0; } -static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) { int retval; unsigned char dac_ctrl = 0; @@ -939,7 +939,7 @@ out: return retval; } -static int snd_at73c213_probe(struct spi_device *spi) +static int __devinit snd_at73c213_probe(struct spi_device *spi) { struct snd_card *card; struct snd_at73c213 *chip; -- 1.5.2.5

2 1

[alsa-devel] [PATCH] Davinci ASoC support
by Vladimir Barinov 18 Feb '08

18 Feb '08

Add ASoC support for the TI Davinci SoC and the Davicni-EVM reference board. It includes: - ASoC Davinci DMA driver - ASoC Davinci I2S (Davinci McBSP module based) driver - ASoC Davinci-EVM reference board Signed-off-by: Vladimir Barinov <vbarinov(a)ru.mvista.com> Index: linux-2.6.25-rc1/sound/soc/davinci/Makefile =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/Makefile @@ -0,0 +1,11 @@ +# DAVINCI Platform Support +snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-i2s-objs := davinci-i2s.o + +obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o + +# DAVINCI Machine Support +snd-soc-evm-objs := davinci-evm.o + +obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-evm.c =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-evm.c @@ -0,0 +1,208 @@ +/* + * ASoC driver for TI DAVINCI EVM platform + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/dma.h> +#include <asm/arch/hardware.h> + +#include "../codecs/tlv320aic3x.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +#define EVM_CODEC_CLOCK 22579200 + +static int evm_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_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops evm_ops = { + .hw_params = evm_hw_params, +}; + +/* davinci-evm machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +/* davinci-evm machine audio_mapnections to the codec pins */ +static const char *audio_map[][3] = { + /* Headphone connected to HPLOUT, HPROUT */ + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LLOUT"}, + {"Line Out", NULL, "RLOUT"}, + + /* Mic connected to (MIC3L | MIC3R) */ + {"MIC3L", NULL, "Mic Bias 2V"}, + {"MIC3R", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "Mic Jack"}, + + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ + {"LINE1L", NULL, "Line In"}, + {"LINE2L", NULL, "Line In"}, + {"LINE1R", NULL, "Line In"}, + {"LINE2R", NULL, "Line In"}, + + {NULL, NULL, NULL}, +}; + +/* Logic for a aic3x as connected on a davinci-evm */ +static int evm_aic3x_init(struct snd_soc_codec *codec) +{ + int i; + + /* Add davinci-evm specific widgets */ + for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + + /* Set up davinci-evm specific audio path audio_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]); + + /* not connected */ + snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); + snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); + snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + + /* always connected */ + snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line In", 1); + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +/* davinci-evm digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +/* davinci-evm audio machine driver */ +static struct snd_soc_machine snd_soc_machine_evm = { + .name = "DaVinci EVM", + .dai_link = &evm_dai, + .num_links = 1, +}; + +/* evm audio private data */ +static struct aic3x_setup_data evm_aic3x_setup = { + .i2c_address = 0x1b, +}; + +/* evm audio subsystem */ +static struct snd_soc_device evm_snd_devdata = { + .machine = &snd_soc_machine_evm, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &evm_aic3x_setup, +}; + +static struct resource evm_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data evm_snd_data = { + .tx_dma_ch = DM644X_DMACH_MCBSP_TX, + .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +}; + +static struct platform_device *evm_snd_device; + +static int __init evm_init(void) +{ + int ret; + + evm_snd_device = platform_device_alloc("soc-audio", 0); + if (!evm_snd_device) + return -ENOMEM; + + platform_set_drvdata(evm_snd_device, &evm_snd_devdata); + evm_snd_devdata.dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = &evm_snd_data; + + ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, + ARRAY_SIZE(evm_snd_resources)); + if (ret) { + platform_device_put(evm_snd_device); + return ret; + } + + ret = platform_device_add(evm_snd_device); + if (ret) + platform_device_put(evm_snd_device); + + return ret; +} + +static void __exit evm_exit(void) +{ + platform_device_unregister(evm_snd_device); +} + +module_init(evm_init); +module_exit(evm_exit); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); +MODULE_LICENSE("GPL"); Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.c =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.c @@ -0,0 +1,407 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" + +#define DAVINCI_MCBSP_DRR_REG 0x00 +#define DAVINCI_MCBSP_DXR_REG 0x04 +#define DAVINCI_MCBSP_SPCR_REG 0x08 +#define DAVINCI_MCBSP_RCR_REG 0x0c +#define DAVINCI_MCBSP_XCR_REG 0x10 +#define DAVINCI_MCBSP_SRGR_REG 0x14 +#define DAVINCI_MCBSP_PCR_REG 0x24 + +#define DAVINCI_MCBSP_SPCR_RRST (1 << 0) +#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4) +#define DAVINCI_MCBSP_SPCR_XRST (1 << 16) +#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20) +#define DAVINCI_MCBSP_SPCR_GRST (1 << 22) +#define DAVINCI_MCBSP_SPCR_FRST (1 << 23) +#define DAVINCI_MCBSP_SPCR_FREE (1 << 25) + +#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_XCR_XFIG (1 << 18) +#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) +#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) +#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) + +#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) +#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) +#define DAVINCI_MCBSP_PCR_FSRP (1 << 2) +#define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) +#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) +#define DAVINCI_MCBSP_PCR_FSRM (1 << 10) +#define DAVINCI_MCBSP_PCR_FSXM (1 << 11) + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +enum { + DAVINCI_MCBSP_WORD_8 = 0, + DAVINCI_MCBSP_WORD_12, + DAVINCI_MCBSP_WORD_16, + DAVINCI_MCBSP_WORD_20, + DAVINCI_MCBSP_WORD_24, + DAVINCI_MCBSP_WORD_32, +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { + .name = "I2S PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { + .name = "I2S PCM Stereo in", +}; + +struct davinci_mcbsp_dev { + void __iomem *base; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_mcbsp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Start frame sync */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST | + DAVINCI_MCBSP_SPCR_FRST, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} + +static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; + case SND_SOC_DAIFMT_CBM_CFM: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_IB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP | + DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_interval *i = NULL; + int mcbsp_word_length; + u32 w; + + /* general line settings */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + DAVINCI_MCBSP_SPCR_RINTM(3) | + DAVINCI_MCBSP_SPCR_XINTM(3) | + DAVINCI_MCBSP_SPCR_FREE); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + mcbsp_word_length = DAVINCI_MCBSP_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + mcbsp_word_length = DAVINCI_MCBSP_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + mcbsp_word_length = DAVINCI_MCBSP_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + + return 0; +} + +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcbsp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcbsp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int davinci_i2s_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *pdata; + int ret; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "McBSP region already claimed\n"); + return -EBUSY; + } + + dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_region; + } + + cpu_dai->private_data = dev; + + dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_free_mem; + } + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + pdata = pdev->dev.platform_data; + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + + return 0; + +err_free_mem: + kfree(dev); +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + return ret; +} + +static void davinci_i2s_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct resource *mem; + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); +} + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +struct snd_soc_cpu_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .type = SND_SOC_DAI_I2S, + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params,}, + .dai_ops = { + .set_fmt = davinci_i2s_set_dai_fmt, + }, +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); +MODULE_LICENSE("GPL"); Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.h =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#ifndef _DAVINCI_I2S_H +#define _DAVINCI_I2S_H + +extern struct snd_soc_cpu_dai davinci_i2s_dai; + +#endif Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.c =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.c @@ -0,0 +1,389 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> + +#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 | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +struct davinci_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ + int master_lch; /* Master DMA channel */ + int slave_lch; /* Slave DMA channel */ + struct davinci_pcm_dma_params *params; /* DMA params */ +}; + +static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int lch = prtd->slave_lch; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + unsigned short src_bidx, dst_bidx; + unsigned int data_type; + unsigned int count; + + period_size = snd_pcm_lib_period_bytes(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); + + data_type = prtd->params->data_type; + count = period_size / data_type; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = prtd->params->dma_addr; + src_bidx = data_type; + dst_bidx = 0; + } else { + src = prtd->params->dma_addr; + dst = dma_pos; + src_bidx = 0; + dst_bidx = data_type; + } + + davinci_set_dma_src_params(lch, src, INCR, W8BIT); + davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); + davinci_set_dma_src_index(lch, src_bidx, 0); + davinci_set_dma_dest_index(lch, dst_bidx, 0); + davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +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); + + if (unlikely(ch_status != DMA_COMPLETE)) + return; + + if (snd_pcm_running(substream)) { + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } +} + +static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int tcc = TCC_ANY; + int ret; + + if (!dma_data) + return -ENODEV; + + prtd->params = dma_data; + + /* Request master DMA channel */ + ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + davinci_pcm_dma_irq, substream, + &prtd->master_lch, &tcc, EVENTQ_0); + if (ret) + return ret; + + /* Request slave DMA channel */ + ret = davinci_request_dma(PARAM_ANY, "Link", + NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); + if (ret) { + davinci_free_dma(prtd->master_lch); + return ret; + } + + /* Link slave DMA channel in loopback */ + davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + + return 0; +} + +static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_start_dma(prtd->master_lch); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_stop_dma(prtd->master_lch); + break; + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static int davinci_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct paramentry_descriptor temp; + + prtd->period = 0; + davinci_pcm_enqueue_dma(substream); + + /* Get slave channel dma params for master channel startup */ + davinci_get_dma_params(prtd->slave_lch, &temp); + davinci_set_dma_params(prtd->master_lch, &temp); + + return 0; +} + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + unsigned int offset; + dma_addr_t count; + dma_addr_t src, dst; + + spin_lock(&prtd->lock); + + davinci_dma_getposition(prtd->master_lch, &src, &dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = src - runtime->dma_addr; + else + count = dst - runtime->dma_addr;; + + spin_unlock(&prtd->lock); + + offset = bytes_to_frames(runtime, count); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + + prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + ret = davinci_pcm_dma_request(substream); + if (ret) { + printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); + kfree(prtd); + } + + return ret; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + + davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + + davinci_free_dma(prtd->slave_lch); + davinci_free_dma(prtd->master_lch); + + kfree(prtd); + + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .prepare = davinci_pcm_prepare, + .trigger = davinci_pcm_trigger, + .pointer = davinci_pcm_pointer, + .mmap = davinci_pcm_mmap, +}; + +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = davinci_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + 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); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void davinci_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->capture.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform davinci_soc_platform = { + .name = "davinci-audio", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, + .pcm_free = davinci_pcm_free, +}; +EXPORT_SYMBOL_GPL(davinci_soc_platform); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); +MODULE_LICENSE("GPL"); Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.h =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.h @@ -0,0 +1,29 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#ifndef _DAVINCI_PCM_H +#define _DAVINCI_PCM_H + +struct davinci_pcm_dma_params { + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ +}; + +struct evm_snd_platform_data { + int tx_dma_ch; + int rx_dma_ch; +}; + +extern struct snd_soc_platform davinci_soc_platform; + +#endif Index: linux-2.6.25-rc1/sound/soc/Kconfig =================================================================== --- linux-2.6.25-rc1.orig/sound/soc/Kconfig +++ linux-2.6.25-rc1/sound/soc/Kconfig @@ -29,6 +29,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/fsl/Kconfig" +source "sound/soc/davinci/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" Index: linux-2.6.25-rc1/sound/soc/Makefile =================================================================== --- linux-2.6.25-rc1.orig/sound/soc/Makefile +++ linux-2.6.25-rc1/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ Index: linux-2.6.25-rc1/sound/soc/davinci/Kconfig =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/Kconfig @@ -0,0 +1,19 @@ +config SND_DAVINCI_SOC + tristate "SoC Audio for the TI DAVINCI chip" + depends on ARCH_DAVINCI && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the DAVINCI AC97 or I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_DAVINCI_SOC_I2S + tristate + +config SND_DAVINCI_SOC_EVM + tristate "SoC Audio support for DaVinci EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci EVM platform.

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[alsa-devel] alsa-info.sh output with ISA cards (cs4236)
by Nigel Henry 18 Feb '08

18 Feb '08

I've been trying to help someone out on the Fedora list, who has had problems with snd-cs4236 since FC6, where he had to add "modprobe snd-cs4236" to /etc/rc.local to get the module to load. With Fedora 8 the problem got worse, and sound would only work as root. Using a tip from someone else on the list, added lines for sound, and cdrom in /etc/security/console.perms.d/50-default.perms, and the sound now works as user. I asked him to run alsa-info.sh, having got the sound working, but there is very limited output, as below. http://pastebin.ca/903074 The question is, should the script work as well with ISA cards, as it does with PCI ones? Nigel.

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Re: [alsa-devel] Random inverted audio channels
by Mark Brown 18 Feb '08

18 Feb '08

On Mon, Feb 18, 2008 at 09:38:12AM +0100, Alexandre BOUIN wrote: > We are porting a WM8900 driver from a 8731 one. It fit quite well our > requirements and should not be the same as your dev branch. > Sincerely, It didn't know you're working on it. No problem at all - I was just asking from curiosity. > > The most likely issue here is a DMA underrun but from what you say you > > may already have ruled that out? It would be helpful if you could run > > through the debugging you've already done here. > I had traced the addresses sent into PDC (soc/at91-pcm.c), and are always > the same. > What other traces are you looking for ? Have you checked to make sure that the PDC is not reporting any errors?

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Re: [alsa-devel] Random inverted audio channels
by Mark Brown 18 Feb '08

18 Feb '08

On Fri, Feb 15, 2008 at 05:39:49PM +0100, Alexandre BOUIN wrote: > This embedded project includes a AT91SAM9260 CPU and a WM8900 codec. JOOI, is this the WM8900 driver in our dev branch or another one? It shouldn't make any difference either way what driver is used for the codec, though, so long as it is generating the relevant clock signals. > We have an issue : when we play a right channel stereo track, sound is > played on right channel, but sometimes on left. > This issue could happen right after starting kernel, or later after 10-20 > sounds played. > With Frank Mandarino, we've checked any issue from SSC(audio bus) or PDC > (dma) controlers, but no much success. > Any idea to help us debugging this issue would be welcome ! The most likely issue here is a DMA underrun but from what you say you may already have ruled that out? It would be helpful if you could run through the debugging you've already done here. One thing to try if your design allows it would be to try making the CPU the master. That should reduce the number of variables, at least.

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[alsa-devel] 2.6.25-rc2 compile error: 'OPL3_PATCH_HASH_SIZE' undeclared
by George Spelvin 18 Feb '08

18 Feb '08

sound/drivers/opl3/opl3_synth.c:308 sound/drivers/opl3/opl3_synth.c:335 I think it's caused by the fact that I don't have the sequencer enabled. The definition inside include/sound/opl3.h is insdie an #ifdef, but the use is not. CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_HWDEP=y CONFIG_SND_RAWMIDI=y # CONFIG_SND_SEQUENCER is not set # CONFIG_SND_MIXER_OSS is not set # CONFIG_SND_PCM_OSS is not set # CONFIG_SND_DYNAMIC_MINORS is not set # CONFIG_SND_SUPPORT_OLD_API is not set CONFIG_SND_VERBOSE_PROCFS=y # CONFIG_SND_VERBOSE_PRINTK is not set # CONFIG_SND_DEBUG is not set

2 2

Re: [alsa-devel] Random inverted audio channels
by Timur Tabi 15 Feb '08

15 Feb '08

Alexandre BOUIN wrote: > We have an issue : when we play a right channel stereo track, sound is > played on right channel, but sometimes on left. > This issue could happen right after starting kernel, or later after 10-20 > sounds played. I've had a similar problem. In my case, the clock from the codec to the chip was unstable. When I attached a 5pF capacitor to it, the clock stabilized and the problem went away. Try using a probe to validate the clock signal. -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] Maya44 revised patch
by Takashi Iwai 15 Feb '08

15 Feb '08

At Thu, 14 Feb 2008 17:59:05 +0100, Rainer Zimmermann wrote: > > > > Hi Takashi and Pavel, > > > Takashi Iwai wrote: > >> The constraint comes from the codec, the chip itself can handle 192kHz, > >> e.g. the 192kHz SPDIF input works fine. Actually more of the ice1724 cards > >> have this limit. If this issue is solved in a general way, it could be > >> implemented to other cards with 192kHz DACs/96kHz ADCs too. > > > > Thanks for a quick information. Then, yes, we should fix this issue in > > general. The separate lists of rates for playback and capture would be > > required in the end... > > > > ok. Actually, as Pavel mentioned SPDIF, I guess there should be a third set of > rates for SPDIF, to be used when the appropriate channel is switched to SPDIF. That's a good point. > >>> Anyway, we can limit this in a scenario like below: - disallow the rate > >>> over 96kHz if the capture stream is being opened - if the rate is set > >>> above 96kHz, the capture stream cannot be opened, returns -EBUSY or so. > >>> > > Yes, I was thinking of something along those lines. But 2 possible issues: > > - maybe a stupid question, but couldn't there be some "backdoor"? E.g., could a > playback stream change its rate to above 96kHz while a capture stream is open? > > - In the latter case, I think it should read: "if a playback stream is running > at above 96kHz, the capture stream cannot be opened". Otherwise, e.g. arecord > couldn't open the device if another app left the rate at 192kHz. Yes, the defnial of opening a capture stream over 96kHz is a drawback in this method. > Are there other examples (non-ice17xx) how this is handled? Either setting a fixed rate via a control or disallow-streams-that- don't-support way. > >>> BTW, is it supposed to be still highly experimental? If the driver works > >>> more or less stably with your device (and on the recent kernel), it'd be > >>> also good to put this to the upstream, i.e. in alsa-kernel tree instead > >>> of alsa-driver tree. > > Yes, so far the driver is stable on my system (2.6.22) and on Piotr's, but I > would still consider it experimental. It still contains experimental code and > some experimental controls, and I guess there should be more testing, in > particular the MI/ODI/O and SPDIF part is still untested. > Well, your decision... It's rather your decision at this stage since I have no hardware for testing. I'd just like to mention that merging to the upstream gives more users a chance to try. > As for the vu-meters: > > > Since the meaning of these read-only controls is clear only upon detailed > > study of the driver and Envy24 datasheet plus I encounter a lot of > > questins/complaints about them, I suggest to change their type from MIXER to > > PCM. They would still be readily accessible from amixer or API. > > No objection from my part, but could it break any app software? any other comments? Well, I think it's fine at this moment since we have (still) no envy24control for vt172x. Actually, some other controls are also candidates to hide from the mixer interface. We need a more detailed review in this regard. Takashi

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[alsa-devel] Salsa-lib pcm_link broken
by Matthieu CASTET 15 Feb '08

15 Feb '08

Hi, snd_pcm_link use SNDRV_PCM_IOCTL_UNLINK insted of SNDRV_PCM_IOCTL_LINK. Matthieu

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[alsa-devel] Random inverted audio channels
by Alexandre BOUIN 15 Feb '08

15 Feb '08

Hi alsa-devel list, I'm currently working on an internal project. Here is some details : This embedded project includes a AT91SAM9260 CPU and a WM8900 codec. Audio driver is using ASoC API. Audio data could be either switch from I2S or DSP A, but I prefer the second one. Audio Codec is master. We have an issue : when we play a right channel stereo track, sound is played on right channel, but sometimes on left. This issue could happen right after starting kernel, or later after 10-20 sounds played. With Frank Mandarino, we've checked any issue from SSC(audio bus) or PDC (dma) controlers, but no much success. Any idea to help us debugging this issue would be welcome ! Regards, Alex.

1 0

[alsa-devel] [PATCH] ASoC: WM9713 driver
by Mark Brown 15 Feb '08

15 Feb '08

This patch adds an ASoC driver for the WM9713 AC97 codec. Signed-off-by: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9713.c | 1289 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9713.h | 53 ++ 4 files changed, 1348 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm9713.c create mode 100644 sound/soc/codecs/wm9713.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 898a7d3..3903ab7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,10 @@ config SND_SOC_WM9712 tristate depends on SND_SOC +config SND_SOC_WM9713 + tristate + depends on SND_SOC + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c6e5338..4e1314c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o +snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -11,5 +12,6 @@ obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o +obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c new file mode 100644 index 0000000..c3d0afd --- /dev/null +++ b/sound/soc/codecs/wm9713.c @@ -0,0 +1,1289 @@ +/* + * wm9713.c -- ALSA Soc WM9713 codec support + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 4th Feb 2006 Initial version. + * + * Features:- + * + * o Support for AC97 Codec, Voice DAC and Aux DAC + * o Support for DAPM + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "wm9713.h" + +#define WM9713_VERSION "0.15" + +struct wm9713_priv { + u32 pll_in; /* PLL input frequency */ + u32 pll_out; /* PLL output frequency */ +}; + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg); +static int ac97_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val); + +/* + * WM9713 register cache + * Reg 0x3c bit 15 is used by touch driver. + */ +static const u16 wm9713_reg[] = { + 0x6174, 0x8080, 0x8080, 0x8080, + 0xc880, 0xe808, 0xe808, 0x0808, + 0x00da, 0x8000, 0xd600, 0xaaa0, + 0xaaa0, 0xaaa0, 0x0000, 0x0000, + 0x0f0f, 0x0040, 0x0000, 0x7f00, + 0x0405, 0x0410, 0xbb80, 0xbb80, + 0x0000, 0xbb80, 0x0000, 0x4523, + 0x0000, 0x2000, 0x7eff, 0xffff, + 0x0000, 0x0000, 0x0080, 0x0000, + 0x0000, 0x0000, 0xfffe, 0xffff, + 0x0000, 0x0000, 0x0000, 0xfffe, + 0x4000, 0x0000, 0x0000, 0x0000, + 0xb032, 0x3e00, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0006, + 0x0001, 0x0000, 0x574d, 0x4c13, + 0x0000, 0x0000, 0x0000 +}; + +/* virtual HP mixers regs */ +#define HPL_MIXER 0x80 +#define HPR_MIXER 0x82 +#define MICB_MUX 0x82 + +static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; +static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; +static const char *wm9713_rec_src[] = + {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker", + "Mono Out", "Zh"}; +static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; +static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", + "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_spk_pga[] = + {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", + "Speaker Vmid", "Inv Vmid"}; +static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone", + "Headphone Vmid"}; +static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"}; +static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"}; +static const char *wm9713_dac_inv[] = + {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone", + "Headphone Mono", "NC", "Vmid"}; +static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"}; +static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"}; +static const char *wm9713_micb_select[] = {"MPB", "MPA"}; + +static const struct soc_enum wm9713_enum[] = { +SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/ +SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ +SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ +SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ +SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ +}; + +static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { +SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), +SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1), +SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), +SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), +SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), +SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), + +SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), +SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), + +SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), +SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), +SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0), +SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), + +SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), +SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), + +SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), +SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), +SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), +SOC_ENUM("ALC Function", wm9713_enum[6]), +SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), +SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), +SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), +SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), +SOC_ENUM("ALC NG Type", wm9713_enum[17]), +SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), + +SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0), +SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), + +SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), +SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), +SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), + +SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), +SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), +SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), + +SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), +SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), + +SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), + +SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), +SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), +SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), + +SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), +SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), +SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), + +SOC_ENUM("Bass Control", wm9713_enum[16]), +SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), +SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), +SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), +SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), +SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), + +SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), +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 + * register map, thus we add a new (virtual) register to help determine the + * audio route within the device. + */ +static int mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 l, r, beep, tone, phone, rec, pcm, aux; + + l = ac97_read(w->codec, HPL_MIXER); + r = ac97_read(w->codec, HPR_MIXER); + beep = ac97_read(w->codec, AC97_PC_BEEP); + tone = ac97_read(w->codec, AC97_MASTER_TONE); + phone = ac97_read(w->codec, AC97_PHONE); + rec = ac97_read(w->codec, AC97_REC_SEL); + pcm = ac97_read(w->codec, AC97_PCM); + aux = ac97_read(w->codec, AC97_AUX); + + if (event & SND_SOC_DAPM_PRE_REG) + return 0; + if ((l & 0x1) || (r & 0x1)) + ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); + else + ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); + + if ((l & 0x2) || (r & 0x2)) + ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); + else + ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); + + if ((l & 0x4) || (r & 0x4)) + ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); + else + ac97_write(w->codec, AC97_PHONE, phone | 0x8000); + + if ((l & 0x8) || (r & 0x8)) + ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); + else + ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); + + if ((l & 0x10) || (r & 0x10)) + ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); + else + ac97_write(w->codec, AC97_PCM, pcm | 0x8000); + + if ((l & 0x20) || (r & 0x20)) + ac97_write(w->codec, AC97_AUX, aux & 0x7fff); + else + ac97_write(w->codec, AC97_AUX, aux | 0x8000); + + return 0; +} + +/* Left Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), +}; + +/* Right Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), +}; + +/* headphone capture mux */ +static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[1]); + +/* headphone mic mux */ +static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[0]); + +/* Speaker Mixer */ +static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1), +SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1), +SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1), +}; + +/* mono mic mux */ +static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[2]); + +/* mono output mux */ +static const struct snd_kcontrol_new wm9713_mono_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[7]); + +/* speaker left output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[8]); + +/* speaker right output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[9]); + +/* headphone left output mux */ +static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[10]); + +/* headphone right output mux */ +static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[11]); + +/* Out3 mux */ +static const struct snd_kcontrol_new wm9713_out3_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[12]); + +/* Out4 mux */ +static const struct snd_kcontrol_new wm9713_out4_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[13]); + +/* DAC inv mux 1 */ +static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[14]); + +/* DAC inv mux 2 */ +static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[15]); + +/* Capture source left */ +static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[3]); + +/* Capture source right */ +static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[4]); + +/* mic source */ +static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[18]); + +/* mic source B virtual control */ +static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[19]); + +static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = { +SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_rec_mux_controls), +SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_mic_mux_controls), +SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mic_mux_controls), +SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mux_controls), +SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkl_mux_controls), +SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkr_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out3_mux_controls), +SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out4_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv1_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv2_mux_controls), +SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcl_mux_controls), +SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcr_mux_controls), +SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, + &wm9713_mic_sel_mux_controls), +SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, + &wm9713_micb_sel_mux_controls), +SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, + &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, + &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, + &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, + &wm9713_speaker_mixer_controls[0], + ARRAY_SIZE(wm9713_speaker_mixer_controls)), +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1), +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1), +SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1), +SND_SOC_DAPM_OUTPUT("MONO"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKL"), +SND_SOC_DAPM_OUTPUT("SPKR"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_INPUT("LINEL"), +SND_SOC_DAPM_INPUT("LINER"), +SND_SOC_DAPM_INPUT("MONOIN"), +SND_SOC_DAPM_INPUT("PCBEEP"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2A"), +SND_SOC_DAPM_INPUT("MIC2B"), +SND_SOC_DAPM_VMID("VMID"), +}; + +static const char *audio_map[][3] = { + /* left HP mixer */ + {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, + {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Left HP Mixer", NULL, "Capture Headphone Mux"}, + + /* right HP mixer */ + {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, + {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Right HP Mixer", NULL, "Capture Headphone Mux"}, + + /* virtual mixer - mixes left & right channels for spk and mono */ + {"AC97 Mixer", NULL, "Left DAC"}, + {"AC97 Mixer", NULL, "Right DAC"}, + {"Line Mixer", NULL, "Right Line In"}, + {"Line Mixer", NULL, "Left Line In"}, + {"HP Mixer", NULL, "Left HP Mixer"}, + {"HP Mixer", NULL, "Right HP Mixer"}, + {"Capture Mixer", NULL, "Left Capture Source"}, + {"Capture Mixer", NULL, "Right Capture Source"}, + + /* speaker mixer */ + {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, + + /* mono mixer */ + {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", NULL, "Capture Mono Mux"}, + + /* DAC inv mux 1 */ + {"DAC Inv Mux 1", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"}, + + /* DAC inv mux 2 */ + {"DAC Inv Mux 2", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"}, + + /* headphone left mux */ + {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"}, + + /* headphone right mux */ + {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"}, + + /* speaker left mux */ + {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"}, + {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* speaker right mux */ + {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"}, + {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"}, + + /* mono mux */ + {"Mono Out Mux", "Mono", "Mono Mixer"}, + {"Mono Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* out 3 mux */ + {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"}, + + /* out 4 mux */ + {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Out Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Out Mux"}, + {"OUT3", NULL, "Out 3"}, + {"Out 3", NULL, "Out 3 Mux"}, + {"OUT4", NULL, "Out 4"}, + {"Out 4", NULL, "Out 4 Mux"}, + {"SPKL", NULL, "Left Speaker"}, + {"Left Speaker", NULL, "Left Speaker Out Mux"}, + {"SPKR", NULL, "Right Speaker"}, + {"Right Speaker", NULL, "Right Speaker Out Mux"}, + {"MONO", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono Out Mux"}, + + /* input pga */ + {"Left Line In", NULL, "LINEL"}, + {"Right Line In", NULL, "LINER"}, + {"Mono In", NULL, "MONOIN"}, + {"Mic A PGA", NULL, "Mic A Pre Amp"}, + {"Mic B PGA", NULL, "Mic B Pre Amp"}, + + /* left capture select */ + {"Left Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Left Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Left Capture Source", "Line", "LINEL"}, + {"Left Capture Source", "Mono In", "MONOIN"}, + {"Left Capture Source", "Headphone", "Left HP Mixer"}, + {"Left Capture Source", "Speaker", "Speaker Mixer"}, + {"Left Capture Source", "Mono Out", "Mono Mixer"}, + + /* right capture select */ + {"Right Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Right Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Right Capture Source", "Line", "LINER"}, + {"Right Capture Source", "Mono In", "MONOIN"}, + {"Right Capture Source", "Headphone", "Right HP Mixer"}, + {"Right Capture Source", "Speaker", "Speaker Mixer"}, + {"Right Capture Source", "Mono Out", "Mono Mixer"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Source"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Source"}, + + /* mic */ + {"Mic A Pre Amp", NULL, "Mic A Source"}, + {"Mic A Source", "Mic 1", "MIC1"}, + {"Mic A Source", "Mic 2 A", "MIC2A"}, + {"Mic A Source", "Mic 2 B", "Mic B Source"}, + {"Mic B Pre Amp", "MPB", "Mic B Source"}, + {"Mic B Source", NULL, "MIC2B"}, + + /* headphone capture */ + {"Capture Headphone Mux", "Stereo", "Capture Mixer"}, + {"Capture Headphone Mux", "Left", "Left Capture Source"}, + {"Capture Headphone Mux", "Right", "Right Capture Source"}, + + /* mono capture */ + {"Capture Mono Mux", "Stereo", "Capture Mixer"}, + {"Capture Mono Mux", "Left", "Left Capture Source"}, + {"Capture Mono Mux", "Right", "Right Capture Source"}, + + {NULL, NULL, NULL}, +}; + +static int wm9713_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); + + /* set up audio path audio_mapnects */ + 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; +} + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || + reg == AC97_CD) + return soc_ac97_ops.read(codec->ac97, reg); + else { + reg = reg >> 1; + + if (reg > (ARRAY_SIZE(wm9713_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg <= (ARRAY_SIZE(wm9713_reg))) + cache[reg] = val; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 divsel:1; + u32 divctl:1; + u32 lf:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the PLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 22) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + + /* The the PLL output is always 98.304MHz. */ + target = 98304000; + + /* If the input frequency is over 14.4MHz then scale it down. */ + if (source > 14400000) { + source >>= 1; + pll_div->divsel = 1; + + if (source > 14400000) { + source >>= 1; + pll_div->divctl = 1; + } else + pll_div->divctl = 0; + + } else { + pll_div->divsel = 0; + pll_div->divctl = 0; + } + + /* Low frequency sources require an additional divide in the + * loop. + */ + if (source < 8192000) { + pll_div->lf = 1; + target >>= 2; + } else + pll_div->lf = 0; + + Ndiv = target / source; + if ((Ndiv < 5) || (Ndiv > 12)) + printk(KERN_WARNING + "WM9713 PLL N value %d out of recommended range!\n", + Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +/** + * Please note that changing the PLL input frequency may require + * resynchronisation with the AC97 controller. + */ +static int wm9713_set_pll(struct snd_soc_codec *codec, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct wm9713_priv *wm9713 = codec->private_data; + u16 reg, reg2; + struct _pll_div pll_div; + + /* turn PLL off ? */ + if (freq_in == 0 || freq_out == 0) { + /* disable PLL power and select ext source */ + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + wm9713->pll_out = 0; + return 0; + } + + pll_factors(&pll_div, freq_in); + + if (pll_div.k == 0) { + reg = (pll_div.n << 12) | (pll_div.lf << 11) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } else { + /* write the fractional k to the reg 0x46 pages */ + reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + + /* K [21:20] */ + reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [19:16] */ + reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [15:12] */ + reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [11:8] */ + reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [7:4] */ + reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } + + /* turn PLL on and select as source */ + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + wm9713->pll_out = freq_out; + wm9713->pll_in = freq_in; + + /* wait 10ms AC97 link frames for the link to stabilise */ + schedule_timeout_interruptible(msecs_to_jiffies(10)); + return 0; +} + +static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + return wm9713_set_pll(codec, pll_id, freq_in, freq_out); +} + +/* + * Tristate the PCM DAI lines, tristate can be disabled by calling + * wm9713_set_dai_fmt() + */ +static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, + int tristate) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; + + if (tristate) + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + + return 0; +} + +/* + * Configure WM9713 clock dividers. + * Voice DAC needs 256 FS + */ +static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM9713_PCMCLK_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKA_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKB_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_HIFI_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_PCMBCLK_DIV: + reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + break; + case WM9713_PCMCLK_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + break; + case WM9713_HIFI_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 reg = 0x8000; + + /* clock masters */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= 0x4000; + gpio |= 0x0010; + break; + case SND_SOC_DAIFMT_CBM_CFS: + reg |= 0x6000; + gpio |= 0x0018; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= 0x0200; + gpio |= 0x001a; + break; + case SND_SOC_DAIFMT_CBS_CFM: + gpio |= 0x0012; + break; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + reg |= 0x00c0; + break; + case SND_SOC_DAIFMT_IB_NF: + reg |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= 0x0040; + break; + } + + /* DAI format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + reg |= 0x0043; + break; + } + + ac97_write(codec, AC97_GPIO_CFG, gpio); + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 0x000c; + break; + } + + /* enable PCM interface in master mode */ + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 status; + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; + ac97_write(codec, AC97_HANDSET_RATE, 0x0280); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, 0x0F80); + ac97_write(codec, AC97_EXTENDED_MID, status); +} + +static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +static int ac97_aux_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 vra, xsle; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + xsle = ac97_read(codec, AC97_PCI_SID); + ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + + return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); +} + +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_FORMATS \ + (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_codec_dai wm9713_dai[] = { +{ + .name = "AC97 HiFi", + .type = SND_SOC_DAI_AC97_BUS, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_hifi_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_aux_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "WM9713 Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .ops = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, + }, + }, +}; +EXPORT_SYMBOL_GPL(wm9713_dai); + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (!(ac97_read(codec, 0) & 0x8000)) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) & 0x8000) + return -EIO; + return 0; +} +EXPORT_SYMBOL_GPL(wm9713_reset); + +static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +{ + u16 reg; + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* enable thermal shutdown */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* enable master bias and vmid */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* disable everything including AC link */ + ac97_write(codec, AC97_EXTENDED_MID, 0xffff); + ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->dapm_state = event; + return 0; +} + +static int wm9713_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; + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int wm9713_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm9713_priv *wm9713 = codec->private_data; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* do we need to re-start the PLL ? */ + if (wm9713->pll_out) + wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + + /* only synchronise the codec if warm reset failed */ + if (ret == 0) { + for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { + if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || + i == AC97_EXTENDED_MSTATUS || i > 0x66) + continue; + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + } + + if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + + return ret; +} + +static int wm9713_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0, reg; + + printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9713_reg); + codec->reg_cache_step = 2; + + codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL); + if (codec->private_data == NULL) { + ret = -ENOMEM; + goto priv_err; + } + + codec->name = "WM9713"; + codec->owner = THIS_MODULE; + codec->dai = wm9713_dai; + codec->num_dai = ARRAY_SIZE(wm9713_dai); + codec->write = ac97_write; + codec->read = ac97_read; + codec->dapm_event = wm9713_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + wm9713_reset(codec, 0); + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "AC97 link error\n"); + goto reset_err; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* unmute the adc - move to kcontrol */ + reg = ac97_read(codec, AC97_CD) & 0x7fff; + ac97_write(codec, AC97_CD, reg); + + wm9713_add_controls(codec); + wm9713_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); + +pcm_err: + snd_soc_free_ac97_codec(codec); + +codec_err: + kfree(codec->private_data); + +priv_err: + kfree(codec->reg_cache); + +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9713_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec->dai); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9713 = { + .probe = wm9713_soc_probe, + .remove = wm9713_soc_remove, + .suspend = wm9713_soc_suspend, + .resume = wm9713_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713); + +MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h new file mode 100644 index 0000000..d357b6c --- /dev/null +++ b/sound/soc/codecs/wm9713.h @@ -0,0 +1,53 @@ +/* + * wm9713.h -- WM9713 Soc Audio driver + */ + +#ifndef _WM9713_H +#define _WM9713_H + +/* clock inputs */ +#define WM9713_CLKA_PIN 0 +#define WM9713_CLKB_PIN 1 + +/* clock divider ID's */ +#define WM9713_PCMCLK_DIV 0 +#define WM9713_CLKA_MULT 1 +#define WM9713_CLKB_MULT 2 +#define WM9713_HIFI_DIV 3 +#define WM9713_PCMBCLK_DIV 4 +#define WM9713_PCMCLK_PLL_DIV 5 +#define WM9713_HIFI_PLL_DIV 6 + +/* Calculate the appropriate bit mask for the external PCM clock divider */ +#define WM9713_PCMDIV(x) ((x - 1) << 8) + +/* Calculate the appropriate bit mask for the external HiFi clock divider */ +#define WM9713_HIFIDIV(x) ((x - 1) << 12) + +/* MCLK clock mulitipliers */ +#define WM9713_CLKA_X1 (0 << 1) +#define WM9713_CLKA_X2 (1 << 1) +#define WM9713_CLKB_X1 (0 << 2) +#define WM9713_CLKB_X2 (1 << 2) + +/* MCLK clock MUX */ +#define WM9713_CLK_MUX_A (0 << 0) +#define WM9713_CLK_MUX_B (1 << 0) + +/* Voice DAI BCLK divider */ +#define WM9713_PCMBCLK_DIV_1 (0 << 9) +#define WM9713_PCMBCLK_DIV_2 (1 << 9) +#define WM9713_PCMBCLK_DIV_4 (2 << 9) +#define WM9713_PCMBCLK_DIV_8 (3 << 9) +#define WM9713_PCMBCLK_DIV_16 (4 << 9) + +#define WM9713_DAI_AC97_HIFI 0 +#define WM9713_DAI_AC97_AUX 1 +#define WM9713_DAI_PCM_VOICE 2 + +extern struct snd_soc_codec_device soc_codec_dev_wm9713; +extern struct snd_soc_codec_dai wm9713_dai[3]; + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm); + +#endif -- 1.5.4.1

2 1

Re: [alsa-devel] midi problem
by rmgls＠free.fr 14 Feb '08

14 Feb '08

Hi Clemens, Thanks for your reply. you rwote: >> rtc: lost some interrupt at 1024hz >This means that something else prevented the RTC interrupt from being >handled. Are you running some closed-source driver (graphics or WiFi)? No. I confirm: with a generic kernel 2.6.22.3-686 all works fine. since i upgraded to 2.6.24-rc8/2.6.24/2.6.24.1 even with the 2.6.22-3 config with very few changes as you can see later,, something breaks. its a debian testing running on asustek p4p800 smp motherboard with 2 xoen 3ghz. 2g ram and radeon 9800. in case this help, here is the conf file: ============================== # # Automatically generated make config: don't edit # Linux kernel version: 2.6.24.1-custom # Thu Feb 14 20:21:06 2008 # # CONFIG_64BIT is not set CONFIG_X86_32=y # CONFIG_X86_64 is not set CONFIG_X86=y CONFIG_GENERIC_TIME=y CONFIG_GENERIC_CMOS_UPDATE=y CONFIG_CLOCKSOURCE_WATCHDOG=y CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y CONFIG_LOCKDEP_SUPPORT=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_SEMAPHORE_SLEEPERS=y CONFIG_MMU=y CONFIG_ZONE_DMA=y CONFIG_QUICKLIST=y CONFIG_GENERIC_ISA_DMA=y CONFIG_GENERIC_IOMAP=y CONFIG_GENERIC_BUG=y CONFIG_GENERIC_HWEIGHT=y CONFIG_ARCH_MAY_HAVE_PC_FDC=y CONFIG_DMI=y # CONFIG_RWSEM_GENERIC_SPINLOCK is not set CONFIG_RWSEM_XCHGADD_ALGORITHM=y # CONFIG_ARCH_HAS_ILOG2_U32 is not set # CONFIG_ARCH_HAS_ILOG2_U64 is not set CONFIG_GENERIC_CALIBRATE_DELAY=y # CONFIG_GENERIC_TIME_VSYSCALL is not set CONFIG_ARCH_SUPPORTS_OPROFILE=y # CONFIG_ZONE_DMA32 is not set CONFIG_ARCH_POPULATES_NODE_MAP=y # CONFIG_AUDIT_ARCH is not set CONFIG_GENERIC_HARDIRQS=y CONFIG_GENERIC_IRQ_PROBE=y CONFIG_GENERIC_PENDING_IRQ=y CONFIG_X86_SMP=y CONFIG_X86_HT=y CONFIG_X86_BIOS_REBOOT=y CONFIG_X86_TRAMPOLINE=y CONFIG_KTIME_SCALAR=y CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" # # General setup # CONFIG_EXPERIMENTAL=y CONFIG_LOCK_KERNEL=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_LOCALVERSION="" # CONFIG_LOCALVERSION_AUTO is not set CONFIG_SWAP=y CONFIG_SYSVIPC=y CONFIG_SYSVIPC_SYSCTL=y CONFIG_POSIX_MQUEUE=y CONFIG_BSD_PROCESS_ACCT=y CONFIG_BSD_PROCESS_ACCT_V3=y # CONFIG_TASKSTATS is not set # CONFIG_USER_NS is not set # CONFIG_PID_NS is not set CONFIG_AUDIT=y # CONFIG_AUDITSYSCALL is not set # CONFIG_IKCONFIG is not set CONFIG_LOG_BUF_SHIFT=15 # CONFIG_CGROUPS is not set CONFIG_FAIR_GROUP_SCHED=y CONFIG_FAIR_USER_SCHED=y # CONFIG_FAIR_CGROUP_SCHED is not set CONFIG_SYSFS_DEPRECATED=y # CONFIG_RELAY is not set CONFIG_BLK_DEV_INITRD=y CONFIG_INITRAMFS_SOURCE="" CONFIG_CC_OPTIMIZE_FOR_SIZE=y CONFIG_SYSCTL=y # CONFIG_EMBEDDED is not set CONFIG_UID16=y CONFIG_SYSCTL_SYSCALL=y CONFIG_KALLSYMS=y # CONFIG_KALLSYMS_ALL is not set # CONFIG_KALLSYMS_EXTRA_PASS is not set CONFIG_HOTPLUG=y CONFIG_PRINTK=y CONFIG_BUG=y CONFIG_ELF_CORE=y CONFIG_BASE_FULL=y CONFIG_FUTEX=y CONFIG_ANON_INODES=y CONFIG_EPOLL=y CONFIG_SIGNALFD=y CONFIG_EVENTFD=y CONFIG_SHMEM=y CONFIG_VM_EVENT_COUNTERS=y CONFIG_SLAB=y # CONFIG_SLUB is not set # CONFIG_SLOB is not set CONFIG_SLABINFO=y CONFIG_RT_MUTEXES=y # CONFIG_TINY_SHMEM is not set CONFIG_BASE_SMALL=0 CONFIG_MODULES=y CONFIG_MODULE_UNLOAD=y CONFIG_MODULE_FORCE_UNLOAD=y CONFIG_MODVERSIONS=y # CONFIG_MODULE_SRCVERSION_ALL is not set CONFIG_KMOD=y CONFIG_STOP_MACHINE=y CONFIG_BLOCK=y CONFIG_LBD=y # CONFIG_BLK_DEV_IO_TRACE is not set CONFIG_LSF=y # CONFIG_BLK_DEV_BSG is not set # # IO Schedulers # CONFIG_IOSCHED_NOOP=y CONFIG_IOSCHED_AS=y CONFIG_IOSCHED_DEADLINE=y CONFIG_IOSCHED_CFQ=y # CONFIG_DEFAULT_AS is not set # CONFIG_DEFAULT_DEADLINE is not set CONFIG_DEFAULT_CFQ=y # CONFIG_DEFAULT_NOOP is not set CONFIG_DEFAULT_IOSCHED="cfq" # # Processor type and features # CONFIG_TICK_ONESHOT=y CONFIG_NO_HZ=y CONFIG_HIGH_RES_TIMERS=y CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_SMP=y CONFIG_X86_PC=y # CONFIG_X86_ELAN is not set # CONFIG_X86_VOYAGER is not set # CONFIG_X86_NUMAQ is not set # CONFIG_X86_SUMMIT is not set # CONFIG_X86_BIGSMP is not set # CONFIG_X86_VISWS is not set # CONFIG_X86_GENERICARCH is not set # CONFIG_X86_ES7000 is not set # CONFIG_X86_VSMP is not set CONFIG_SCHED_NO_NO_OMIT_FRAME_POINTER=y # CONFIG_PARAVIRT_GUEST is not set # CONFIG_M386 is not set # CONFIG_M486 is not set # CONFIG_M586 is not set # CONFIG_M586TSC is not set # CONFIG_M586MMX is not set CONFIG_M686=y # CONFIG_MPENTIUMII is not set # CONFIG_MPENTIUMIII is not set # CONFIG_MPENTIUMM is not set # CONFIG_MPENTIUM4 is not set # CONFIG_MK6 is not set # CONFIG_MK7 is not set # CONFIG_MK8 is not set # CONFIG_MCRUSOE is not set # CONFIG_MEFFICEON is not set # CONFIG_MWINCHIPC6 is not set # CONFIG_MWINCHIP2 is not set # CONFIG_MWINCHIP3D is not set # CONFIG_MGEODEGX1 is not set # CONFIG_MGEODE_LX is not set # CONFIG_MCYRIXIII is not set # CONFIG_MVIAC3_2 is not set # CONFIG_MVIAC7 is not set # CONFIG_MPSC is not set # CONFIG_MCORE2 is not set # CONFIG_GENERIC_CPU is not set # CONFIG_X86_GENERIC is not set CONFIG_X86_CMPXCHG=y CONFIG_X86_L1_CACHE_SHIFT=5 CONFIG_X86_XADD=y CONFIG_X86_PPRO_FENCE=y CONFIG_X86_WP_WORKS_OK=y CONFIG_X86_INVLPG=y CONFIG_X86_BSWAP=y CONFIG_X86_POPAD_OK=y CONFIG_X86_GOOD_APIC=y CONFIG_X86_USE_PPRO_CHECKSUM=y CONFIG_X86_TSC=y CONFIG_X86_CMOV=y CONFIG_X86_MINIMUM_CPU_FAMILY=4 CONFIG_HPET_TIMER=y CONFIG_NR_CPUS=8 CONFIG_SCHED_SMT=y CONFIG_SCHED_MC=y CONFIG_PREEMPT_NONE=y # CONFIG_PREEMPT_VOLUNTARY is not set # CONFIG_PREEMPT is not set # CONFIG_PREEMPT_BKL is not set CONFIG_X86_LOCAL_APIC=y CONFIG_X86_IO_APIC=y CONFIG_X86_MCE=y CONFIG_X86_MCE_NONFATAL=m CONFIG_X86_MCE_P4THERMAL=y CONFIG_VM86=y CONFIG_TOSHIBA=m CONFIG_I8K=m # CONFIG_X86_REBOOTFIXUPS is not set CONFIG_MICROCODE=m CONFIG_MICROCODE_OLD_INTERFACE=y CONFIG_X86_MSR=m CONFIG_X86_CPUID=m # CONFIG_NOHIGHMEM is not set CONFIG_HIGHMEM4G=y # CONFIG_HIGHMEM64G is not set CONFIG_PAGE_OFFSET=0xC0000000 CONFIG_HIGHMEM=y CONFIG_ARCH_FLATMEM_ENABLE=y CONFIG_ARCH_SPARSEMEM_ENABLE=y CONFIG_ARCH_SELECT_MEMORY_MODEL=y CONFIG_SELECT_MEMORY_MODEL=y CONFIG_FLATMEM_MANUAL=y # CONFIG_DISCONTIGMEM_MANUAL is not set # CONFIG_SPARSEMEM_MANUAL is not set CONFIG_FLATMEM=y CONFIG_FLAT_NODE_MEM_MAP=y CONFIG_SPARSEMEM_STATIC=y # CONFIG_SPARSEMEM_VMEMMAP_ENABLE is not set CONFIG_SPLIT_PTLOCK_CPUS=4 # CONFIG_RESOURCES_64BIT is not set CONFIG_ZONE_DMA_FLAG=1 CONFIG_BOUNCE=y CONFIG_NR_QUICK=1 CONFIG_VIRT_TO_BUS=y # CONFIG_HIGHPTE is not set # CONFIG_MATH_EMULATION is not set CONFIG_MTRR=y CONFIG_EFI=y # CONFIG_IRQBALANCE is not set CONFIG_BOOT_IOREMAP=y # CONFIG_SECCOMP is not set # CONFIG_HZ_100 is not set CONFIG_HZ_250=y # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set CONFIG_HZ=250 CONFIG_KEXEC=y # CONFIG_CRASH_DUMP is not set CONFIG_PHYSICAL_START=0x100000 # CONFIG_RELOCATABLE is not set CONFIG_PHYSICAL_ALIGN=0x100000 CONFIG_HOTPLUG_CPU=y CONFIG_COMPAT_VDSO=y CONFIG_ARCH_ENABLE_MEMORY_HOTPLUG=y # # Power management options # CONFIG_PM=y CONFIG_PM_LEGACY=y # CONFIG_PM_DEBUG is not set CONFIG_PM_SLEEP_SMP=y CONFIG_PM_SLEEP=y CONFIG_SUSPEND_SMP_POSSIBLE=y CONFIG_SUSPEND=y CONFIG_HIBERNATION_SMP_POSSIBLE=y # CONFIG_HIBERNATION is not set CONFIG_ACPI=y CONFIG_ACPI_SLEEP=y CONFIG_ACPI_PROCFS=y CONFIG_ACPI_PROCFS_POWER=y CONFIG_ACPI_SYSFS_POWER=y CONFIG_ACPI_PROC_EVENT=y CONFIG_ACPI_AC=m CONFIG_ACPI_BATTERY=m CONFIG_ACPI_BUTTON=m CONFIG_ACPI_FAN=m CONFIG_ACPI_DOCK=m CONFIG_ACPI_BAY=m CONFIG_ACPI_PROCESSOR=m CONFIG_ACPI_HOTPLUG_CPU=y CONFIG_ACPI_THERMAL=m CONFIG_ACPI_ASUS=m CONFIG_ACPI_TOSHIBA=m CONFIG_ACPI_BLACKLIST_YEAR=0 # CONFIG_ACPI_DEBUG is not set CONFIG_ACPI_EC=y CONFIG_ACPI_POWER=y CONFIG_ACPI_SYSTEM=y CONFIG_X86_PM_TIMER=y CONFIG_ACPI_CONTAINER=m CONFIG_ACPI_SBS=m CONFIG_APM=m # CONFIG_APM_IGNORE_USER_SUSPEND is not set # CONFIG_APM_DO_ENABLE is not set # CONFIG_APM_CPU_IDLE is not set # CONFIG_APM_DISPLAY_BLANK is not set # CONFIG_APM_ALLOW_INTS is not set # CONFIG_APM_REAL_MODE_POWER_OFF is not set # # CPU Frequency scaling # CONFIG_CPU_FREQ=y CONFIG_CPU_FREQ_TABLE=m # CONFIG_CPU_FREQ_DEBUG is not set CONFIG_CPU_FREQ_STAT=m # CONFIG_CPU_FREQ_STAT_DETAILS is not set CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE=y # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set CONFIG_CPU_FREQ_GOV_PERFORMANCE=y CONFIG_CPU_FREQ_GOV_POWERSAVE=m CONFIG_CPU_FREQ_GOV_USERSPACE=m CONFIG_CPU_FREQ_GOV_ONDEMAND=m CONFIG_CPU_FREQ_GOV_CONSERVATIVE=m # # CPUFreq processor drivers # CONFIG_X86_ACPI_CPUFREQ=m CONFIG_X86_POWERNOW_K6=m CONFIG_X86_POWERNOW_K7=m CONFIG_X86_POWERNOW_K7_ACPI=y CONFIG_X86_POWERNOW_K8=m CONFIG_X86_POWERNOW_K8_ACPI=y CONFIG_X86_GX_SUSPMOD=m CONFIG_X86_SPEEDSTEP_CENTRINO=m CONFIG_X86_SPEEDSTEP_CENTRINO_TABLE=y CONFIG_X86_SPEEDSTEP_ICH=m CONFIG_X86_SPEEDSTEP_SMI=m CONFIG_X86_P4_CLOCKMOD=m CONFIG_X86_CPUFREQ_NFORCE2=m CONFIG_X86_LONGRUN=m CONFIG_X86_LONGHAUL=m CONFIG_X86_E_POWERSAVER=m # # shared options # # CONFIG_X86_ACPI_CPUFREQ_PROC_INTF is not set CONFIG_X86_SPEEDSTEP_LIB=m CONFIG_X86_SPEEDSTEP_RELAXED_CAP_CHECK=y # CONFIG_CPU_IDLE is not set # # Bus options (PCI etc.) # CONFIG_PCI=y # CONFIG_PCI_GOBIOS is not set # CONFIG_PCI_GOMMCONFIG is not set # CONFIG_PCI_GODIRECT is not set CONFIG_PCI_GOANY=y CONFIG_PCI_BIOS=y CONFIG_PCI_DIRECT=y CONFIG_PCI_MMCONFIG=y CONFIG_PCI_DOMAINS=y CONFIG_PCIEPORTBUS=y CONFIG_HOTPLUG_PCI_PCIE=m CONFIG_PCIEAER=y CONFIG_ARCH_SUPPORTS_MSI=y CONFIG_PCI_MSI=y CONFIG_PCI_LEGACY=y # CONFIG_PCI_DEBUG is not set CONFIG_HT_IRQ=y CONFIG_ISA_DMA_API=y CONFIG_ISA=y # CONFIG_EISA is not set # CONFIG_MCA is not set CONFIG_SCx200=m CONFIG_SCx200HR_TIMER=m CONFIG_K8_NB=y CONFIG_PCCARD=m # CONFIG_PCMCIA_DEBUG is not set CONFIG_PCMCIA=m CONFIG_PCMCIA_LOAD_CIS=y CONFIG_PCMCIA_IOCTL=y CONFIG_CARDBUS=y # # PC-card bridges # CONFIG_YENTA=m CONFIG_YENTA_O2=y CONFIG_YENTA_RICOH=y CONFIG_YENTA_TI=y CONFIG_YENTA_ENE_TUNE=y CONFIG_YENTA_TOSHIBA=y CONFIG_PD6729=m CONFIG_I82092=m CONFIG_I82365=m CONFIG_TCIC=m CONFIG_PCMCIA_PROBE=y CONFIG_PCCARD_NONSTATIC=m CONFIG_HOTPLUG_PCI=m CONFIG_HOTPLUG_PCI_FAKE=m CONFIG_HOTPLUG_PCI_COMPAQ=m # CONFIG_HOTPLUG_PCI_COMPAQ_NVRAM is not set CONFIG_HOTPLUG_PCI_IBM=m CONFIG_HOTPLUG_PCI_ACPI=m CONFIG_HOTPLUG_PCI_ACPI_IBM=m CONFIG_HOTPLUG_PCI_CPCI=y CONFIG_HOTPLUG_PCI_CPCI_ZT5550=m CONFIG_HOTPLUG_PCI_CPCI_GENERIC=m CONFIG_HOTPLUG_PCI_SHPC=m # # Executable file formats / Emulations # CONFIG_BINFMT_ELF=y CONFIG_BINFMT_AOUT=m CONFIG_BINFMT_MISC=m # # Networking # CONFIG_NET=y # # Networking options # CONFIG_PACKET=y CONFIG_PACKET_MMAP=y CONFIG_UNIX=y CONFIG_XFRM=y CONFIG_XFRM_USER=m # CONFIG_XFRM_SUB_POLICY is not set # CONFIG_XFRM_MIGRATE is not set CONFIG_NET_KEY=m # CONFIG_NET_KEY_MIGRATE is not set CONFIG_INET=y CONFIG_IP_MULTICAST=y CONFIG_IP_ADVANCED_ROUTER=y CONFIG_ASK_IP_FIB_HASH=y # CONFIG_IP_FIB_TRIE is not set CONFIG_IP_FIB_HASH=y CONFIG_IP_MULTIPLE_TABLES=y CONFIG_IP_ROUTE_MULTIPATH=y CONFIG_IP_ROUTE_VERBOSE=y # CONFIG_IP_PNP is not set CONFIG_NET_IPIP=m CONFIG_NET_IPGRE=m CONFIG_NET_IPGRE_BROADCAST=y CONFIG_IP_MROUTE=y CONFIG_IP_PIMSM_V1=y CONFIG_IP_PIMSM_V2=y # CONFIG_ARPD is not set CONFIG_SYN_COOKIES=y CONFIG_INET_AH=m CONFIG_INET_ESP=m CONFIG_INET_IPCOMP=m CONFIG_INET_XFRM_TUNNEL=m CONFIG_INET_TUNNEL=m CONFIG_INET_XFRM_MODE_TRANSPORT=m CONFIG_INET_XFRM_MODE_TUNNEL=m CONFIG_INET_XFRM_MODE_BEET=m CONFIG_INET_LRO=m CONFIG_INET_DIAG=m CONFIG_INET_TCP_DIAG=m CONFIG_TCP_CONG_ADVANCED=y CONFIG_TCP_CONG_BIC=y CONFIG_TCP_CONG_CUBIC=m CONFIG_TCP_CONG_WESTWOOD=m CONFIG_TCP_CONG_HTCP=m CONFIG_TCP_CONG_HSTCP=m CONFIG_TCP_CONG_HYBLA=m CONFIG_TCP_CONG_VEGAS=m CONFIG_TCP_CONG_SCALABLE=m CONFIG_TCP_CONG_LP=m CONFIG_TCP_CONG_VENO=m CONFIG_TCP_CONG_YEAH=m CONFIG_TCP_CONG_ILLINOIS=m CONFIG_DEFAULT_BIC=y # CONFIG_DEFAULT_CUBIC is not set # CONFIG_DEFAULT_HTCP is not set # CONFIG_DEFAULT_VEGAS is not set # CONFIG_DEFAULT_WESTWOOD is not set # CONFIG_DEFAULT_RENO is not set CONFIG_DEFAULT_TCP_CONG="bic" # CONFIG_TCP_MD5SIG is not set CONFIG_IP_VS=m # CONFIG_IP_VS_DEBUG is not set CONFIG_IP_VS_TAB_BITS=12 # # IPVS transport protocol load balancing support # CONFIG_IP_VS_PROTO_TCP=y CONFIG_IP_VS_PROTO_UDP=y CONFIG_IP_VS_PROTO_ESP=y CONFIG_IP_VS_PROTO_AH=y # # IPVS scheduler # CONFIG_IP_VS_RR=m CONFIG_IP_VS_WRR=m CONFIG_IP_VS_LC=m CONFIG_IP_VS_WLC=m CONFIG_IP_VS_LBLC=m CONFIG_IP_VS_LBLCR=m CONFIG_IP_VS_DH=m CONFIG_IP_VS_SH=m CONFIG_IP_VS_SED=m CONFIG_IP_VS_NQ=m # # IPVS application helper # CONFIG_IP_VS_FTP=m CONFIG_IPV6=m CONFIG_IPV6_PRIVACY=y # CONFIG_IPV6_ROUTER_PREF is not set # CONFIG_IPV6_OPTIMISTIC_DAD is not set CONFIG_INET6_AH=m CONFIG_INET6_ESP=m CONFIG_INET6_IPCOMP=m # CONFIG_IPV6_MIP6 is not set CONFIG_INET6_XFRM_TUNNEL=m CONFIG_INET6_TUNNEL=m CONFIG_INET6_XFRM_MODE_TRANSPORT=m CONFIG_INET6_XFRM_MODE_TUNNEL=m CONFIG_INET6_XFRM_MODE_BEET=m CONFIG_INET6_XFRM_MODE_ROUTEOPTIMIZATION=m CONFIG_IPV6_SIT=m CONFIG_IPV6_TUNNEL=m # CONFIG_IPV6_MULTIPLE_TABLES is not set # CONFIG_NETLABEL is not set CONFIG_NETWORK_SECMARK=y CONFIG_NETFILTER=y # CONFIG_NETFILTER_DEBUG is not set CONFIG_BRIDGE_NETFILTER=y # # Core Netfilter Configuration # CONFIG_NETFILTER_NETLINK=m CONFIG_NETFILTER_NETLINK_QUEUE=m CONFIG_NETFILTER_NETLINK_LOG=m CONFIG_NF_CONNTRACK_ENABLED=m CONFIG_NF_CONNTRACK=m CONFIG_NF_CT_ACCT=y CONFIG_NF_CONNTRACK_MARK=y CONFIG_NF_CONNTRACK_SECMARK=y CONFIG_NF_CONNTRACK_EVENTS=y CONFIG_NF_CT_PROTO_GRE=m CONFIG_NF_CT_PROTO_SCTP=m # CONFIG_NF_CT_PROTO_UDPLITE is not set CONFIG_NF_CONNTRACK_AMANDA=m CONFIG_NF_CONNTRACK_FTP=m CONFIG_NF_CONNTRACK_H323=m CONFIG_NF_CONNTRACK_IRC=m CONFIG_NF_CONNTRACK_NETBIOS_NS=m CONFIG_NF_CONNTRACK_PPTP=m CONFIG_NF_CONNTRACK_SANE=m CONFIG_NF_CONNTRACK_SIP=m CONFIG_NF_CONNTRACK_TFTP=m CONFIG_NF_CT_NETLINK=m CONFIG_NETFILTER_XTABLES=m CONFIG_NETFILTER_XT_TARGET_CLASSIFY=m CONFIG_NETFILTER_XT_TARGET_CONNMARK=m CONFIG_NETFILTER_XT_TARGET_DSCP=m CONFIG_NETFILTER_XT_TARGET_MARK=m CONFIG_NETFILTER_XT_TARGET_NFQUEUE=m CONFIG_NETFILTER_XT_TARGET_NFLOG=m CONFIG_NETFILTER_XT_TARGET_NOTRACK=m # CONFIG_NETFILTER_XT_TARGET_TRACE is not set CONFIG_NETFILTER_XT_TARGET_SECMARK=m CONFIG_NETFILTER_XT_TARGET_CONNSECMARK=m CONFIG_NETFILTER_XT_TARGET_TCPMSS=m CONFIG_NETFILTER_XT_MATCH_COMMENT=m CONFIG_NETFILTER_XT_MATCH_CONNBYTES=m # CONFIG_NETFILTER_XT_MATCH_CONNLIMIT is not set CONFIG_NETFILTER_XT_MATCH_CONNMARK=m CONFIG_NETFILTER_XT_MATCH_CONNTRACK=m CONFIG_NETFILTER_XT_MATCH_DCCP=m CONFIG_NETFILTER_XT_MATCH_DSCP=m CONFIG_NETFILTER_XT_MATCH_ESP=m CONFIG_NETFILTER_XT_MATCH_HELPER=m CONFIG_NETFILTER_XT_MATCH_LENGTH=m CONFIG_NETFILTER_XT_MATCH_LIMIT=m CONFIG_NETFILTER_XT_MATCH_MAC=m CONFIG_NETFILTER_XT_MATCH_MARK=m CONFIG_NETFILTER_XT_MATCH_POLICY=m CONFIG_NETFILTER_XT_MATCH_MULTIPORT=m CONFIG_NETFILTER_XT_MATCH_PHYSDEV=m CONFIG_NETFILTER_XT_MATCH_PKTTYPE=m CONFIG_NETFILTER_XT_MATCH_QUOTA=m CONFIG_NETFILTER_XT_MATCH_REALM=m CONFIG_NETFILTER_XT_MATCH_SCTP=m CONFIG_NETFILTER_XT_MATCH_STATE=m CONFIG_NETFILTER_XT_MATCH_STATISTIC=m CONFIG_NETFILTER_XT_MATCH_STRING=m CONFIG_NETFILTER_XT_MATCH_TCPMSS=m # CONFIG_NETFILTER_XT_MATCH_TIME is not set # CONFIG_NETFILTER_XT_MATCH_U32 is not set CONFIG_NETFILTER_XT_MATCH_HASHLIMIT=m # # IP: Netfilter Configuration # CONFIG_NF_CONNTRACK_IPV4=m CONFIG_NF_CONNTRACK_PROC_COMPAT=y CONFIG_IP_NF_QUEUE=m CONFIG_IP_NF_IPTABLES=m CONFIG_IP_NF_MATCH_IPRANGE=m CONFIG_IP_NF_MATCH_TOS=m CONFIG_IP_NF_MATCH_RECENT=m CONFIG_IP_NF_MATCH_ECN=m CONFIG_IP_NF_MATCH_AH=m CONFIG_IP_NF_MATCH_TTL=m CONFIG_IP_NF_MATCH_OWNER=m CONFIG_IP_NF_MATCH_ADDRTYPE=m CONFIG_IP_NF_FILTER=m CONFIG_IP_NF_TARGET_REJECT=m CONFIG_IP_NF_TARGET_LOG=m CONFIG_IP_NF_TARGET_ULOG=m CONFIG_NF_NAT=m CONFIG_NF_NAT_NEEDED=y CONFIG_IP_NF_TARGET_MASQUERADE=m CONFIG_IP_NF_TARGET_REDIRECT=m CONFIG_IP_NF_TARGET_NETMAP=m CONFIG_IP_NF_TARGET_SAME=m CONFIG_NF_NAT_SNMP_BASIC=m CONFIG_NF_NAT_PROTO_GRE=m CONFIG_NF_NAT_FTP=m CONFIG_NF_NAT_IRC=m CONFIG_NF_NAT_TFTP=m CONFIG_NF_NAT_AMANDA=m CONFIG_NF_NAT_PPTP=m CONFIG_NF_NAT_H323=m CONFIG_NF_NAT_SIP=m CONFIG_IP_NF_MANGLE=m CONFIG_IP_NF_TARGET_TOS=m CONFIG_IP_NF_TARGET_ECN=m CONFIG_IP_NF_TARGET_TTL=m CONFIG_IP_NF_TARGET_CLUSTERIP=m CONFIG_IP_NF_RAW=m CONFIG_IP_NF_ARPTABLES=m CONFIG_IP_NF_ARPFILTER=m CONFIG_IP_NF_ARP_MANGLE=m # # IPv6: Netfilter Configuration (EXPERIMENTAL) # CONFIG_NF_CONNTRACK_IPV6=m CONFIG_IP6_NF_QUEUE=m CONFIG_IP6_NF_IPTABLES=m CONFIG_IP6_NF_MATCH_RT=m CONFIG_IP6_NF_MATCH_OPTS=m CONFIG_IP6_NF_MATCH_FRAG=m CONFIG_IP6_NF_MATCH_HL=m CONFIG_IP6_NF_MATCH_OWNER=m CONFIG_IP6_NF_MATCH_IPV6HEADER=m CONFIG_IP6_NF_MATCH_AH=m CONFIG_IP6_NF_MATCH_MH=m CONFIG_IP6_NF_MATCH_EUI64=m CONFIG_IP6_NF_FILTER=m CONFIG_IP6_NF_TARGET_LOG=m CONFIG_IP6_NF_TARGET_REJECT=m CONFIG_IP6_NF_MANGLE=m CONFIG_IP6_NF_TARGET_HL=m CONFIG_IP6_NF_RAW=m # # DECnet: Netfilter Configuration # CONFIG_DECNET_NF_GRABULATOR=m # # Bridge: Netfilter Configuration # CONFIG_BRIDGE_NF_EBTABLES=m CONFIG_BRIDGE_EBT_BROUTE=m CONFIG_BRIDGE_EBT_T_FILTER=m CONFIG_BRIDGE_EBT_T_NAT=m CONFIG_BRIDGE_EBT_802_3=m CONFIG_BRIDGE_EBT_AMONG=m CONFIG_BRIDGE_EBT_ARP=m CONFIG_BRIDGE_EBT_IP=m CONFIG_BRIDGE_EBT_LIMIT=m CONFIG_BRIDGE_EBT_MARK=m CONFIG_BRIDGE_EBT_PKTTYPE=m CONFIG_BRIDGE_EBT_STP=m CONFIG_BRIDGE_EBT_VLAN=m CONFIG_BRIDGE_EBT_ARPREPLY=m CONFIG_BRIDGE_EBT_DNAT=m CONFIG_BRIDGE_EBT_MARK_T=m CONFIG_BRIDGE_EBT_REDIRECT=m CONFIG_BRIDGE_EBT_SNAT=m CONFIG_BRIDGE_EBT_LOG=m CONFIG_BRIDGE_EBT_ULOG=m CONFIG_IP_DCCP=m CONFIG_INET_DCCP_DIAG=m CONFIG_IP_DCCP_ACKVEC=y # # DCCP CCIDs Configuration (EXPERIMENTAL) # CONFIG_IP_DCCP_CCID2=m # CONFIG_IP_DCCP_CCID2_DEBUG is not set CONFIG_IP_DCCP_CCID3=m CONFIG_IP_DCCP_TFRC_LIB=m # CONFIG_IP_DCCP_CCID3_DEBUG is not set CONFIG_IP_DCCP_CCID3_RTO=100 # # DCCP Kernel Hacking # # CONFIG_IP_DCCP_DEBUG is not set CONFIG_IP_SCTP=m # CONFIG_SCTP_DBG_MSG is not set # CONFIG_SCTP_DBG_OBJCNT is not set # CONFIG_SCTP_HMAC_NONE is not set # CONFIG_SCTP_HMAC_SHA1 is not set CONFIG_SCTP_HMAC_MD5=y CONFIG_TIPC=m CONFIG_TIPC_ADVANCED=y CONFIG_TIPC_ZONES=3 CONFIG_TIPC_CLUSTERS=1 CONFIG_TIPC_NODES=255 CONFIG_TIPC_SLAVE_NODES=0 CONFIG_TIPC_PORTS=8191 CONFIG_TIPC_LOG=0 # CONFIG_TIPC_DEBUG is not set CONFIG_ATM=y CONFIG_ATM_CLIP=y # CONFIG_ATM_CLIP_NO_ICMP is not set CONFIG_ATM_LANE=m CONFIG_ATM_MPOA=m CONFIG_ATM_BR2684=m # CONFIG_ATM_BR2684_IPFILTER is not set CONFIG_BRIDGE=m CONFIG_VLAN_8021Q=m CONFIG_DECNET=m # CONFIG_DECNET_ROUTER is not set CONFIG_LLC=y CONFIG_LLC2=m CONFIG_IPX=m # CONFIG_IPX_INTERN is not set CONFIG_ATALK=m CONFIG_DEV_APPLETALK=m CONFIG_LTPC=m # CONFIG_COPS is not set CONFIG_IPDDP=m CONFIG_IPDDP_ENCAP=y CONFIG_IPDDP_DECAP=y CONFIG_X25=m CONFIG_LAPB=m CONFIG_ECONET=m CONFIG_ECONET_AUNUDP=y CONFIG_ECONET_NATIVE=y CONFIG_WAN_ROUTER=m CONFIG_NET_SCHED=y # # Queueing/Scheduling # CONFIG_NET_SCH_CBQ=m CONFIG_NET_SCH_HTB=m CONFIG_NET_SCH_HFSC=m CONFIG_NET_SCH_ATM=m CONFIG_NET_SCH_PRIO=m # CONFIG_NET_SCH_RR is not set CONFIG_NET_SCH_RED=m CONFIG_NET_SCH_SFQ=m CONFIG_NET_SCH_TEQL=m CONFIG_NET_SCH_TBF=m CONFIG_NET_SCH_GRED=m CONFIG_NET_SCH_DSMARK=m CONFIG_NET_SCH_NETEM=m CONFIG_NET_SCH_INGRESS=m # # Classification # CONFIG_NET_CLS=y CONFIG_NET_CLS_BASIC=m CONFIG_NET_CLS_TCINDEX=m CONFIG_NET_CLS_ROUTE4=m CONFIG_NET_CLS_ROUTE=y CONFIG_NET_CLS_FW=m CONFIG_NET_CLS_U32=m CONFIG_CLS_U32_PERF=y CONFIG_CLS_U32_MARK=y CONFIG_NET_CLS_RSVP=m CONFIG_NET_CLS_RSVP6=m CONFIG_NET_EMATCH=y CONFIG_NET_EMATCH_STACK=32 CONFIG_NET_EMATCH_CMP=m CONFIG_NET_EMATCH_NBYTE=m CONFIG_NET_EMATCH_U32=m CONFIG_NET_EMATCH_META=m CONFIG_NET_EMATCH_TEXT=m CONFIG_NET_CLS_ACT=y CONFIG_NET_ACT_POLICE=m CONFIG_NET_ACT_GACT=m CONFIG_GACT_PROB=y CONFIG_NET_ACT_MIRRED=m CONFIG_NET_ACT_IPT=m # CONFIG_NET_ACT_NAT is not set CONFIG_NET_ACT_PEDIT=m CONFIG_NET_ACT_SIMP=m # CONFIG_NET_CLS_POLICE is not set CONFIG_NET_CLS_IND=y CONFIG_NET_SCH_FIFO=y # # Network testing # CONFIG_NET_PKTGEN=m CONFIG_HAMRADIO=y # # Packet Radio protocols # CONFIG_AX25=m # CONFIG_AX25_DAMA_SLAVE is not set CONFIG_NETROM=m CONFIG_ROSE=m # # AX.25 network device drivers # CONFIG_MKISS=m CONFIG_6PACK=m CONFIG_BPQETHER=m CONFIG_SCC=m # CONFIG_SCC_DELAY is not set # CONFIG_SCC_TRXECHO is not set CONFIG_BAYCOM_SER_FDX=m CONFIG_BAYCOM_SER_HDX=m CONFIG_BAYCOM_PAR=m CONFIG_BAYCOM_EPP=m CONFIG_YAM=m CONFIG_IRDA=m # # IrDA protocols # CONFIG_IRLAN=m CONFIG_IRNET=m CONFIG_IRCOMM=m # CONFIG_IRDA_ULTRA is not set # # IrDA options # CONFIG_IRDA_CACHE_LAST_LSAP=y CONFIG_IRDA_FAST_RR=y CONFIG_IRDA_DEBUG=y # # Infrared-port device drivers # # # SIR device drivers # CONFIG_IRTTY_SIR=m # # Dongle support # CONFIG_DONGLE=y CONFIG_ESI_DONGLE=m CONFIG_ACTISYS_DONGLE=m CONFIG_TEKRAM_DONGLE=m CONFIG_TOIM3232_DONGLE=m CONFIG_LITELINK_DONGLE=m CONFIG_MA600_DONGLE=m CONFIG_GIRBIL_DONGLE=m CONFIG_MCP2120_DONGLE=m CONFIG_OLD_BELKIN_DONGLE=m CONFIG_ACT200L_DONGLE=m CONFIG_KINGSUN_DONGLE=m # CONFIG_KSDAZZLE_DONGLE is not set # CONFIG_KS959_DONGLE is not set # # Old SIR device drivers # # # Old Serial dongle support # # # FIR device drivers # CONFIG_USB_IRDA=m CONFIG_SIGMATEL_FIR=m CONFIG_NSC_FIR=m CONFIG_WINBOND_FIR=m CONFIG_TOSHIBA_FIR=m CONFIG_SMC_IRCC_FIR=m CONFIG_ALI_FIR=m CONFIG_VLSI_FIR=m CONFIG_VIA_FIR=m CONFIG_MCS_FIR=m CONFIG_BT=m CONFIG_BT_L2CAP=m CONFIG_BT_SCO=m CONFIG_BT_RFCOMM=m CONFIG_BT_RFCOMM_TTY=y CONFIG_BT_BNEP=m CONFIG_BT_BNEP_MC_FILTER=y CONFIG_BT_BNEP_PROTO_FILTER=y CONFIG_BT_CMTP=m CONFIG_BT_HIDP=m # # Bluetooth device drivers # CONFIG_BT_HCIUSB=m CONFIG_BT_HCIUSB_SCO=y # CONFIG_BT_HCIBTSDIO is not set CONFIG_BT_HCIUART=m CONFIG_BT_HCIUART_H4=y CONFIG_BT_HCIUART_BCSP=y # CONFIG_BT_HCIUART_LL is not set CONFIG_BT_HCIBCM203X=m CONFIG_BT_HCIBPA10X=m CONFIG_BT_HCIBFUSB=m CONFIG_BT_HCIDTL1=m CONFIG_BT_HCIBT3C=m CONFIG_BT_HCIBLUECARD=m CONFIG_BT_HCIBTUART=m CONFIG_BT_HCIVHCI=m CONFIG_AF_RXRPC=m # CONFIG_AF_RXRPC_DEBUG is not set CONFIG_RXKAD=m CONFIG_FIB_RULES=y # # Wireless # CONFIG_CFG80211=m CONFIG_NL80211=y CONFIG_WIRELESS_EXT=y CONFIG_MAC80211=m CONFIG_MAC80211_RCSIMPLE=y CONFIG_MAC80211_LEDS=y # CONFIG_MAC80211_DEBUGFS is not set # CONFIG_MAC80211_DEBUG is not set CONFIG_IEEE80211=m # CONFIG_IEEE80211_DEBUG is not set CONFIG_IEEE80211_CRYPT_WEP=m CONFIG_IEEE80211_CRYPT_CCMP=m CONFIG_IEEE80211_CRYPT_TKIP=m CONFIG_IEEE80211_SOFTMAC=m # CONFIG_IEEE80211_SOFTMAC_DEBUG is not set CONFIG_RFKILL=m CONFIG_RFKILL_INPUT=m CONFIG_RFKILL_LEDS=y # CONFIG_NET_9P is not set # # Device Drivers # # # Generic Driver Options # CONFIG_UEVENT_HELPER_PATH="/sbin/hotplug" CONFIG_STANDALONE=y CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=m # CONFIG_DEBUG_DRIVER is not set # CONFIG_DEBUG_DEVRES is not set # CONFIG_SYS_HYPERVISOR is not set CONFIG_CONNECTOR=m CONFIG_MTD=m # CONFIG_MTD_DEBUG is not set CONFIG_MTD_CONCAT=m CONFIG_MTD_PARTITIONS=y CONFIG_MTD_REDBOOT_PARTS=m CONFIG_MTD_REDBOOT_DIRECTORY_BLOCK=-1 # CONFIG_MTD_REDBOOT_PARTS_UNALLOCATED is not set # CONFIG_MTD_REDBOOT_PARTS_READONLY is not set # # User Modules And Translation Layers # CONFIG_MTD_CHAR=m CONFIG_MTD_BLKDEVS=m CONFIG_MTD_BLOCK=m CONFIG_MTD_BLOCK_RO=m CONFIG_FTL=m CONFIG_NFTL=m CONFIG_NFTL_RW=y CONFIG_INFTL=m CONFIG_RFD_FTL=m CONFIG_SSFDC=m # CONFIG_MTD_OOPS is not set # # RAM/ROM/Flash chip drivers # CONFIG_MTD_CFI=m CONFIG_MTD_JEDECPROBE=m CONFIG_MTD_GEN_PROBE=m # CONFIG_MTD_CFI_ADV_OPTIONS is not set CONFIG_MTD_MAP_BANK_WIDTH_1=y CONFIG_MTD_MAP_BANK_WIDTH_2=y CONFIG_MTD_MAP_BANK_WIDTH_4=y # CONFIG_MTD_MAP_BANK_WIDTH_8 is not set # CONFIG_MTD_MAP_BANK_WIDTH_16 is not set # CONFIG_MTD_MAP_BANK_WIDTH_32 is not set CONFIG_MTD_CFI_I1=y CONFIG_MTD_CFI_I2=y # CONFIG_MTD_CFI_I4 is not set # CONFIG_MTD_CFI_I8 is not set CONFIG_MTD_CFI_INTELEXT=m CONFIG_MTD_CFI_AMDSTD=m CONFIG_MTD_CFI_STAA=m CONFIG_MTD_CFI_UTIL=m CONFIG_MTD_RAM=m CONFIG_MTD_ROM=m CONFIG_MTD_ABSENT=m # # Mapping drivers for chip access # CONFIG_MTD_COMPLEX_MAPPINGS=y CONFIG_MTD_PHYSMAP=m CONFIG_MTD_PHYSMAP_START=0x8000000 CONFIG_MTD_PHYSMAP_LEN=0x4000000 CONFIG_MTD_PHYSMAP_BANKWIDTH=2 CONFIG_MTD_PNC2000=m CONFIG_MTD_SC520CDP=m CONFIG_MTD_NETSC520=m CONFIG_MTD_TS5500=m CONFIG_MTD_SBC_GXX=m CONFIG_MTD_SCx200_DOCFLASH=m # CONFIG_MTD_AMD76XROM is not set # CONFIG_MTD_ICHXROM is not set # CONFIG_MTD_ESB2ROM is not set # CONFIG_MTD_CK804XROM is not set # CONFIG_MTD_SCB2_FLASH is not set CONFIG_MTD_NETtel=m CONFIG_MTD_DILNETPC=m CONFIG_MTD_DILNETPC_BOOTSIZE=0x80000 # CONFIG_MTD_L440GX is not set CONFIG_MTD_PCI=m # CONFIG_MTD_INTEL_VR_NOR is not set CONFIG_MTD_PLATRAM=m # # Self-contained MTD device drivers # CONFIG_MTD_PMC551=m # CONFIG_MTD_PMC551_BUGFIX is not set # CONFIG_MTD_PMC551_DEBUG is not set CONFIG_MTD_DATAFLASH=m CONFIG_MTD_M25P80=m CONFIG_MTD_SLRAM=m CONFIG_MTD_PHRAM=m CONFIG_MTD_MTDRAM=m CONFIG_MTDRAM_TOTAL_SIZE=4096 CONFIG_MTDRAM_ERASE_SIZE=128 CONFIG_MTD_BLOCK2MTD=m # # Disk-On-Chip Device Drivers # CONFIG_MTD_DOC2000=m CONFIG_MTD_DOC2001=m CONFIG_MTD_DOC2001PLUS=m CONFIG_MTD_DOCPROBE=m CONFIG_MTD_DOCECC=m # CONFIG_MTD_DOCPROBE_ADVANCED is not set CONFIG_MTD_DOCPROBE_ADDRESS=0 CONFIG_MTD_NAND=m # CONFIG_MTD_NAND_VERIFY_WRITE is not set # CONFIG_MTD_NAND_ECC_SMC is not set # CONFIG_MTD_NAND_MUSEUM_IDS is not set CONFIG_MTD_NAND_IDS=m CONFIG_MTD_NAND_DISKONCHIP=m # CONFIG_MTD_NAND_DISKONCHIP_PROBE_ADVANCED is not set CONFIG_MTD_NAND_DISKONCHIP_PROBE_ADDRESS=0 # CONFIG_MTD_NAND_DISKONCHIP_BBTWRITE is not set CONFIG_MTD_NAND_CAFE=m CONFIG_MTD_NAND_CS553X=m # CONFIG_MTD_NAND_NANDSIM is not set CONFIG_MTD_NAND_PLATFORM=m # CONFIG_MTD_ALAUDA is not set CONFIG_MTD_ONENAND=m CONFIG_MTD_ONENAND_VERIFY_WRITE=y # CONFIG_MTD_ONENAND_OTP is not set # CONFIG_MTD_ONENAND_2X_PROGRAM is not set # CONFIG_MTD_ONENAND_SIM is not set # # UBI - Unsorted block images # CONFIG_MTD_UBI=m CONFIG_MTD_UBI_WL_THRESHOLD=4096 CONFIG_MTD_UBI_BEB_RESERVE=1 # CONFIG_MTD_UBI_GLUEBI is not set # # UBI debugging options # # CONFIG_MTD_UBI_DEBUG is not set CONFIG_PARPORT=m CONFIG_PARPORT_PC=m CONFIG_PARPORT_SERIAL=m CONFIG_PARPORT_PC_FIFO=y # CONFIG_PARPORT_PC_SUPERIO is not set CONFIG_PARPORT_PC_PCMCIA=m # CONFIG_PARPORT_GSC is not set CONFIG_PARPORT_AX88796=m CONFIG_PARPORT_1284=y CONFIG_PARPORT_NOT_PC=y CONFIG_PNP=y # CONFIG_PNP_DEBUG is not set # # Protocols # CONFIG_ISAPNP=y CONFIG_PNPBIOS=y CONFIG_PNPBIOS_PROC_FS=y CONFIG_PNPACPI=y CONFIG_BLK_DEV=y CONFIG_BLK_DEV_FD=m CONFIG_BLK_DEV_XD=m CONFIG_PARIDE=m # # Parallel IDE high-level drivers # CONFIG_PARIDE_PD=m CONFIG_PARIDE_PCD=m CONFIG_PARIDE_PF=m CONFIG_PARIDE_PT=m CONFIG_PARIDE_PG=m # # Parallel IDE protocol modules # CONFIG_PARIDE_ATEN=m CONFIG_PARIDE_BPCK=m CONFIG_PARIDE_BPCK6=m CONFIG_PARIDE_COMM=m CONFIG_PARIDE_DSTR=m CONFIG_PARIDE_FIT2=m CONFIG_PARIDE_FIT3=m CONFIG_PARIDE_EPAT=m # CONFIG_PARIDE_EPATC8 is not set CONFIG_PARIDE_EPIA=m CONFIG_PARIDE_FRIQ=m CONFIG_PARIDE_FRPW=m CONFIG_PARIDE_KBIC=m CONFIG_PARIDE_KTTI=m CONFIG_PARIDE_ON20=m CONFIG_PARIDE_ON26=m CONFIG_BLK_CPQ_DA=m CONFIG_BLK_CPQ_CISS_DA=m CONFIG_CISS_SCSI_TAPE=y CONFIG_BLK_DEV_DAC960=m CONFIG_BLK_DEV_UMEM=m # CONFIG_BLK_DEV_COW_COMMON is not set CONFIG_BLK_DEV_LOOP=m CONFIG_BLK_DEV_CRYPTOLOOP=m CONFIG_BLK_DEV_NBD=m CONFIG_BLK_DEV_SX8=m # CONFIG_BLK_DEV_UB is not set CONFIG_BLK_DEV_RAM=y CONFIG_BLK_DEV_RAM_COUNT=16 CONFIG_BLK_DEV_RAM_SIZE=8192 CONFIG_BLK_DEV_RAM_BLOCKSIZE=1024 CONFIG_CDROM_PKTCDVD=m CONFIG_CDROM_PKTCDVD_BUFFERS=8 # CONFIG_CDROM_PKTCDVD_WCACHE is not set CONFIG_ATA_OVER_ETH=m CONFIG_MISC_DEVICES=y CONFIG_IBM_ASM=m CONFIG_PHANTOM=m # CONFIG_EEPROM_93CX6 is not set CONFIG_SGI_IOC4=m CONFIG_TIFM_CORE=m CONFIG_TIFM_7XX1=m CONFIG_ASUS_LAPTOP=m # CONFIG_FUJITSU_LAPTOP is not set CONFIG_MSI_LAPTOP=m CONFIG_SONY_LAPTOP=m # CONFIG_SONYPI_COMPAT is not set CONFIG_THINKPAD_ACPI=m # CONFIG_THINKPAD_ACPI_DEBUG is not set CONFIG_THINKPAD_ACPI_BAY=y CONFIG_IDE=m CONFIG_BLK_DEV_IDE=m # # Please see Documentation/ide.txt for help/info on IDE drives # # CONFIG_BLK_DEV_IDE_SATA is not set # CONFIG_BLK_DEV_HD_IDE is not set CONFIG_BLK_DEV_IDEDISK=m # CONFIG_IDEDISK_MULTI_MODE is not set CONFIG_BLK_DEV_IDECS=m CONFIG_BLK_DEV_DELKIN=m CONFIG_BLK_DEV_IDECD=m CONFIG_BLK_DEV_IDETAPE=m CONFIG_BLK_DEV_IDEFLOPPY=m # CONFIG_BLK_DEV_IDESCSI is not set # CONFIG_BLK_DEV_IDEACPI is not set # CONFIG_IDE_TASK_IOCTL is not set CONFIG_IDE_PROC_FS=y # # IDE chipset support/bugfixes # CONFIG_IDE_GENERIC=m # CONFIG_BLK_DEV_PLATFORM is not set CONFIG_BLK_DEV_CMD640=y # CONFIG_BLK_DEV_CMD640_ENHANCED is not set CONFIG_BLK_DEV_IDEPNP=y # # PCI IDE chipsets support # CONFIG_BLK_DEV_IDEPCI=y CONFIG_IDEPCI_SHARE_IRQ=y CONFIG_IDEPCI_PCIBUS_ORDER=y # CONFIG_BLK_DEV_OFFBOARD is not set CONFIG_BLK_DEV_GENERIC=m CONFIG_BLK_DEV_OPTI621=m CONFIG_BLK_DEV_RZ1000=m CONFIG_BLK_DEV_IDEDMA_PCI=y CONFIG_BLK_DEV_AEC62XX=m CONFIG_BLK_DEV_ALI15X3=m # CONFIG_WDC_ALI15X3 is not set CONFIG_BLK_DEV_AMD74XX=m CONFIG_BLK_DEV_ATIIXP=m CONFIG_BLK_DEV_CMD64X=m CONFIG_BLK_DEV_TRIFLEX=m CONFIG_BLK_DEV_CY82C693=m CONFIG_BLK_DEV_CS5520=m CONFIG_BLK_DEV_CS5530=m CONFIG_BLK_DEV_CS5535=m CONFIG_BLK_DEV_HPT34X=m # CONFIG_HPT34X_AUTODMA is not set CONFIG_BLK_DEV_HPT366=m CONFIG_BLK_DEV_JMICRON=m CONFIG_BLK_DEV_SC1200=m CONFIG_BLK_DEV_PIIX=m CONFIG_BLK_DEV_IT8213=m CONFIG_BLK_DEV_IT821X=m CONFIG_BLK_DEV_NS87415=m CONFIG_BLK_DEV_PDC202XX_OLD=m CONFIG_PDC202XX_BURST=y CONFIG_BLK_DEV_PDC202XX_NEW=m CONFIG_BLK_DEV_SVWKS=m CONFIG_BLK_DEV_SIIMAGE=m CONFIG_BLK_DEV_SIS5513=m CONFIG_BLK_DEV_SLC90E66=m CONFIG_BLK_DEV_TRM290=m CONFIG_BLK_DEV_VIA82CXXX=m CONFIG_BLK_DEV_TC86C001=m # CONFIG_IDE_ARM is not set # # Other IDE chipsets support # # # Note: most of these also require special kernel boot parameters # # CONFIG_BLK_DEV_4DRIVES is not set # CONFIG_BLK_DEV_ALI14XX is not set # CONFIG_BLK_DEV_DTC2278 is not set # CONFIG_BLK_DEV_HT6560B is not set # CONFIG_BLK_DEV_QD65XX is not set # CONFIG_BLK_DEV_UMC8672 is not set CONFIG_BLK_DEV_IDEDMA=y CONFIG_IDE_ARCH_OBSOLETE_INIT=y # CONFIG_BLK_DEV_HD is not set # # SCSI device support # CONFIG_RAID_ATTRS=m CONFIG_SCSI=m CONFIG_SCSI_DMA=y CONFIG_SCSI_TGT=m CONFIG_SCSI_NETLINK=y CONFIG_SCSI_PROC_FS=y # # SCSI support type (disk, tape, CD-ROM) # CONFIG_BLK_DEV_SD=m CONFIG_CHR_DEV_ST=m CONFIG_CHR_DEV_OSST=m CONFIG_BLK_DEV_SR=m CONFIG_BLK_DEV_SR_VENDOR=y CONFIG_CHR_DEV_SG=m CONFIG_CHR_DEV_SCH=m # # Some SCSI devices (e.g. CD jukebox) support multiple LUNs # CONFIG_SCSI_MULTI_LUN=y CONFIG_SCSI_CONSTANTS=y CONFIG_SCSI_LOGGING=y CONFIG_SCSI_SCAN_ASYNC=y CONFIG_SCSI_WAIT_SCAN=m # # SCSI Transports # CONFIG_SCSI_SPI_ATTRS=m CONFIG_SCSI_FC_ATTRS=m # CONFIG_SCSI_FC_TGT_ATTRS is not set CONFIG_SCSI_ISCSI_ATTRS=m CONFIG_SCSI_SAS_ATTRS=m CONFIG_SCSI_SAS_LIBSAS=m # CONFIG_SCSI_SAS_ATA is not set # CONFIG_SCSI_SAS_LIBSAS_DEBUG is not set CONFIG_SCSI_SRP_ATTRS=m # CONFIG_SCSI_SRP_TGT_ATTRS is not set CONFIG_SCSI_LOWLEVEL=y CONFIG_ISCSI_TCP=m CONFIG_BLK_DEV_3W_XXXX_RAID=m CONFIG_SCSI_3W_9XXX=m CONFIG_SCSI_7000FASST=m CONFIG_SCSI_ACARD=m CONFIG_SCSI_AHA152X=m CONFIG_SCSI_AHA1542=m CONFIG_SCSI_AACRAID=m CONFIG_SCSI_AIC7XXX=m CONFIG_AIC7XXX_CMDS_PER_DEVICE=8 CONFIG_AIC7XXX_RESET_DELAY_MS=15000 CONFIG_AIC7XXX_DEBUG_ENABLE=y CONFIG_AIC7XXX_DEBUG_MASK=0 CONFIG_AIC7XXX_REG_PRETTY_PRINT=y CONFIG_SCSI_AIC7XXX_OLD=m CONFIG_SCSI_AIC79XX=m CONFIG_AIC79XX_CMDS_PER_DEVICE=32 CONFIG_AIC79XX_RESET_DELAY_MS=15000 CONFIG_AIC79XX_DEBUG_ENABLE=y CONFIG_AIC79XX_DEBUG_MASK=0 CONFIG_AIC79XX_REG_PRETTY_PRINT=y CONFIG_SCSI_AIC94XX=m # CONFIG_AIC94XX_DEBUG is not set CONFIG_SCSI_DPT_I2O=m CONFIG_SCSI_ADVANSYS=m CONFIG_SCSI_IN2000=m CONFIG_SCSI_ARCMSR=m # CONFIG_SCSI_ARCMSR_AER is not set CONFIG_MEGARAID_NEWGEN=y CONFIG_MEGARAID_MM=m CONFIG_MEGARAID_MAILBOX=m CONFIG_MEGARAID_LEGACY=m CONFIG_MEGARAID_SAS=m CONFIG_SCSI_HPTIOP=m CONFIG_SCSI_BUSLOGIC=m # CONFIG_SCSI_OMIT_FLASHPOINT is not set CONFIG_SCSI_DMX3191D=m CONFIG_SCSI_DTC3280=m CONFIG_SCSI_EATA=m CONFIG_SCSI_EATA_TAGGED_QUEUE=y CONFIG_SCSI_EATA_LINKED_COMMANDS=y CONFIG_SCSI_EATA_MAX_TAGS=16 CONFIG_SCSI_FUTURE_DOMAIN=m CONFIG_SCSI_GDTH=m CONFIG_SCSI_GENERIC_NCR5380=m CONFIG_SCSI_GENERIC_NCR5380_MMIO=m CONFIG_SCSI_GENERIC_NCR53C400=y CONFIG_SCSI_IPS=m CONFIG_SCSI_INITIO=m # CONFIG_SCSI_INIA100 is not set CONFIG_SCSI_PPA=m CONFIG_SCSI_IMM=m # CONFIG_SCSI_IZIP_EPP16 is not set # CONFIG_SCSI_IZIP_SLOW_CTR is not set CONFIG_SCSI_NCR53C406A=m CONFIG_SCSI_STEX=m CONFIG_SCSI_SYM53C8XX_2=m CONFIG_SCSI_SYM53C8XX_DMA_ADDRESSING_MODE=1 CONFIG_SCSI_SYM53C8XX_DEFAULT_TAGS=16 CONFIG_SCSI_SYM53C8XX_MAX_TAGS=64 CONFIG_SCSI_SYM53C8XX_MMIO=y CONFIG_SCSI_IPR=m # CONFIG_SCSI_IPR_TRACE is not set # CONFIG_SCSI_IPR_DUMP is not set CONFIG_SCSI_PAS16=m CONFIG_SCSI_PSI240I=m CONFIG_SCSI_QLOGIC_FAS=m CONFIG_SCSI_QLOGIC_1280=m CONFIG_SCSI_QLA_FC=m CONFIG_SCSI_QLA_ISCSI=m CONFIG_SCSI_LPFC=m # CONFIG_SCSI_SEAGATE is not set CONFIG_SCSI_SYM53C416=m CONFIG_SCSI_DC395x=m CONFIG_SCSI_DC390T=m CONFIG_SCSI_T128=m CONFIG_SCSI_U14_34F=m CONFIG_SCSI_U14_34F_TAGGED_QUEUE=y CONFIG_SCSI_U14_34F_LINKED_COMMANDS=y CONFIG_SCSI_U14_34F_MAX_TAGS=8 CONFIG_SCSI_ULTRASTOR=m CONFIG_SCSI_NSP32=m CONFIG_SCSI_DEBUG=m CONFIG_SCSI_SRP=m # CONFIG_SCSI_LOWLEVEL_PCMCIA is not set CONFIG_ATA=m # CONFIG_ATA_NONSTANDARD is not set CONFIG_ATA_ACPI=y CONFIG_SATA_AHCI=m CONFIG_SATA_SVW=m CONFIG_ATA_PIIX=m CONFIG_SATA_MV=m CONFIG_SATA_NV=m CONFIG_PDC_ADMA=m CONFIG_SATA_QSTOR=m CONFIG_SATA_PROMISE=m CONFIG_SATA_SX4=m CONFIG_SATA_SIL=m CONFIG_SATA_SIL24=m CONFIG_SATA_SIS=m CONFIG_SATA_ULI=m CONFIG_SATA_VIA=m CONFIG_SATA_VITESSE=m CONFIG_SATA_INIC162X=m # CONFIG_PATA_ACPI is not set # CONFIG_PATA_ALI is not set # CONFIG_PATA_AMD is not set # CONFIG_PATA_ARTOP is not set # CONFIG_PATA_ATIIXP is not set # CONFIG_PATA_CMD640_PCI is not set # CONFIG_PATA_CMD64X is not set # CONFIG_PATA_CS5520 is not set # CONFIG_PATA_CS5530 is not set # CONFIG_PATA_CS5535 is not set # CONFIG_PATA_CS5536 is not set # CONFIG_PATA_CYPRESS is not set CONFIG_PATA_EFAR=m CONFIG_ATA_GENERIC=m # CONFIG_PATA_HPT366 is not set # CONFIG_PATA_HPT37X is not set # CONFIG_PATA_HPT3X2N is not set # CONFIG_PATA_HPT3X3 is not set CONFIG_PATA_ISAPNP=m # CONFIG_PATA_IT821X is not set # CONFIG_PATA_IT8213 is not set # CONFIG_PATA_JMICRON is not set CONFIG_PATA_LEGACY=m # CONFIG_PATA_TRIFLEX is not set CONFIG_PATA_MARVELL=m # CONFIG_PATA_MPIIX is not set # CONFIG_PATA_OLDPIIX is not set # CONFIG_PATA_NETCELL is not set # CONFIG_PATA_NS87410 is not set # CONFIG_PATA_NS87415 is not set # CONFIG_PATA_OPTI is not set # CONFIG_PATA_OPTIDMA is not set CONFIG_PATA_PCMCIA=m CONFIG_PATA_PDC_OLD=m CONFIG_PATA_QDI=m CONFIG_PATA_RADISYS=m # CONFIG_PATA_RZ1000 is not set # CONFIG_PATA_SC1200 is not set # CONFIG_PATA_SERVERWORKS is not set # CONFIG_PATA_PDC2027X is not set # CONFIG_PATA_SIL680 is not set CONFIG_PATA_SIS=m # CONFIG_PATA_VIA is not set CONFIG_PATA_WINBOND=m CONFIG_PATA_WINBOND_VLB=m CONFIG_MD=y CONFIG_BLK_DEV_MD=m CONFIG_MD_LINEAR=m CONFIG_MD_RAID0=m CONFIG_MD_RAID1=m CONFIG_MD_RAID10=m CONFIG_MD_RAID456=m CONFIG_MD_RAID5_RESHAPE=y CONFIG_MD_MULTIPATH=m CONFIG_MD_FAULTY=m CONFIG_BLK_DEV_DM=m # CONFIG_DM_DEBUG is not set CONFIG_DM_CRYPT=m CONFIG_DM_SNAPSHOT=m CONFIG_DM_MIRROR=m CONFIG_DM_ZERO=m CONFIG_DM_MULTIPATH=m CONFIG_DM_MULTIPATH_EMC=m # CONFIG_DM_MULTIPATH_RDAC is not set # CONFIG_DM_MULTIPATH_HP is not set CONFIG_DM_DELAY=m # CONFIG_DM_UEVENT is not set CONFIG_FUSION=y CONFIG_FUSION_SPI=m CONFIG_FUSION_FC=m CONFIG_FUSION_SAS=m CONFIG_FUSION_MAX_SGE=40 CONFIG_FUSION_CTL=m CONFIG_FUSION_LAN=m # CONFIG_FUSION_LOGGING is not set # # IEEE 1394 (FireWire) support # CONFIG_FIREWIRE=m CONFIG_FIREWIRE_OHCI=m CONFIG_FIREWIRE_SBP2=m # CONFIG_IEEE1394 is not set CONFIG_I2O=m CONFIG_I2O_LCT_NOTIFY_ON_CHANGES=y CONFIG_I2O_EXT_ADAPTEC=y CONFIG_I2O_CONFIG=m CONFIG_I2O_CONFIG_OLD_IOCTL=y CONFIG_I2O_BUS=m CONFIG_I2O_BLOCK=m CONFIG_I2O_SCSI=m CONFIG_I2O_PROC=m # CONFIG_MACINTOSH_DRIVERS is not set CONFIG_NETDEVICES=y # CONFIG_NETDEVICES_MULTIQUEUE is not set CONFIG_IFB=m CONFIG_DUMMY=m CONFIG_BONDING=m # CONFIG_MACVLAN is not set CONFIG_EQUALIZER=m CONFIG_TUN=m # CONFIG_VETH is not set CONFIG_NET_SB1000=m CONFIG_ARCNET=m CONFIG_ARCNET_1201=m CONFIG_ARCNET_1051=m CONFIG_ARCNET_RAW=m CONFIG_ARCNET_CAP=m CONFIG_ARCNET_COM90xx=m CONFIG_ARCNET_COM90xxIO=m CONFIG_ARCNET_RIM_I=m CONFIG_ARCNET_COM20020=m CONFIG_ARCNET_COM20020_ISA=m CONFIG_ARCNET_COM20020_PCI=m CONFIG_PHYLIB=m # # MII PHY device drivers # CONFIG_MARVELL_PHY=m CONFIG_DAVICOM_PHY=m CONFIG_QSEMI_PHY=m CONFIG_LXT_PHY=m CONFIG_CICADA_PHY=m CONFIG_VITESSE_PHY=m CONFIG_SMSC_PHY=m CONFIG_BROADCOM_PHY=m # CONFIG_ICPLUS_PHY is not set CONFIG_FIXED_PHY=m # CONFIG_FIXED_MII_10_FDX is not set # CONFIG_FIXED_MII_100_FDX is not set # CONFIG_FIXED_MII_1000_FDX is not set CONFIG_FIXED_MII_AMNT=1 # CONFIG_MDIO_BITBANG is not set CONFIG_NET_ETHERNET=y CONFIG_MII=m CONFIG_HAPPYMEAL=m CONFIG_SUNGEM=m CONFIG_CASSINI=m CONFIG_NET_VENDOR_3COM=y CONFIG_EL1=m CONFIG_EL2=m CONFIG_ELPLUS=m CONFIG_EL16=m CONFIG_EL3=m CONFIG_3C515=m CONFIG_VORTEX=m CONFIG_TYPHOON=m CONFIG_LANCE=m CONFIG_NET_VENDOR_SMC=y CONFIG_WD80x3=m CONFIG_ULTRA=m CONFIG_SMC9194=m CONFIG_NET_VENDOR_RACAL=y CONFIG_NI52=m CONFIG_NI65=m CONFIG_NET_TULIP=y CONFIG_DE2104X=m CONFIG_TULIP=m # CONFIG_TULIP_MWI is not set # CONFIG_TULIP_MMIO is not set CONFIG_TULIP_NAPI=y CONFIG_TULIP_NAPI_HW_MITIGATION=y CONFIG_DE4X5=m CONFIG_WINBOND_840=m CONFIG_DM9102=m CONFIG_ULI526X=m CONFIG_PCMCIA_XIRCOM=m CONFIG_AT1700=m CONFIG_DEPCA=m CONFIG_HP100=m CONFIG_NET_ISA=y CONFIG_E2100=m CONFIG_EWRK3=m CONFIG_EEXPRESS=m CONFIG_EEXPRESS_PRO=m CONFIG_HPLAN_PLUS=m CONFIG_HPLAN=m CONFIG_LP486E=m CONFIG_ETH16I=m CONFIG_NE2000=m CONFIG_ZNET=m CONFIG_SEEQ8005=m # CONFIG_IBM_NEW_EMAC_ZMII is not set # CONFIG_IBM_NEW_EMAC_RGMII is not set # CONFIG_IBM_NEW_EMAC_TAH is not set # CONFIG_IBM_NEW_EMAC_EMAC4 is not set CONFIG_NET_PCI=y CONFIG_PCNET32=m CONFIG_PCNET32_NAPI=y CONFIG_AMD8111_ETH=m CONFIG_AMD8111E_NAPI=y CONFIG_ADAPTEC_STARFIRE=m CONFIG_ADAPTEC_STARFIRE_NAPI=y CONFIG_AC3200=m CONFIG_APRICOT=m CONFIG_B44=m CONFIG_B44_PCI_AUTOSELECT=y CONFIG_B44_PCICORE_AUTOSELECT=y CONFIG_B44_PCI=y CONFIG_FORCEDETH=m # CONFIG_FORCEDETH_NAPI is not set CONFIG_CS89x0=m CONFIG_EEPRO100=m CONFIG_E100=m CONFIG_FEALNX=m CONFIG_NATSEMI=m CONFIG_NE2K_PCI=m CONFIG_8139CP=m CONFIG_8139TOO=m CONFIG_8139TOO_PIO=y CONFIG_8139TOO_TUNE_TWISTER=y CONFIG_8139TOO_8129=y # CONFIG_8139_OLD_RX_RESET is not set CONFIG_SIS900=m CONFIG_EPIC100=m CONFIG_SUNDANCE=m # CONFIG_SUNDANCE_MMIO is not set CONFIG_TLAN=m CONFIG_VIA_RHINE=m # CONFIG_VIA_RHINE_MMIO is not set CONFIG_VIA_RHINE_NAPI=y CONFIG_SC92031=m # CONFIG_NET_POCKET is not set CONFIG_NETDEV_1000=y CONFIG_ACENIC=m # CONFIG_ACENIC_OMIT_TIGON_I is not set CONFIG_DL2K=m CONFIG_E1000=m CONFIG_E1000_NAPI=y # CONFIG_E1000_DISABLE_PACKET_SPLIT is not set # CONFIG_E1000E is not set # CONFIG_IP1000 is not set CONFIG_NS83820=m CONFIG_HAMACHI=m CONFIG_YELLOWFIN=m CONFIG_R8169=m CONFIG_R8169_NAPI=y CONFIG_R8169_VLAN=y CONFIG_SIS190=m CONFIG_SKGE=m # CONFIG_SKGE_DEBUG is not set CONFIG_SKY2=m # CONFIG_SKY2_DEBUG is not set # CONFIG_SK98LIN is not set CONFIG_VIA_VELOCITY=m CONFIG_TIGON3=m CONFIG_BNX2=m CONFIG_QLA3XXX=m CONFIG_ATL1=m CONFIG_NETDEV_10000=y CONFIG_CHELSIO_T1=m CONFIG_CHELSIO_T1_1G=y CONFIG_CHELSIO_T1_NAPI=y CONFIG_CHELSIO_T3=m # CONFIG_IXGBE is not set CONFIG_IXGB=m CONFIG_IXGB_NAPI=y CONFIG_S2IO=m CONFIG_S2IO_NAPI=y CONFIG_MYRI10GE=m CONFIG_NETXEN_NIC=m # CONFIG_NIU is not set CONFIG_MLX4_CORE=m CONFIG_MLX4_DEBUG=y # CONFIG_TEHUTI is not set CONFIG_TR=y CONFIG_IBMTR=m CONFIG_IBMOL=m CONFIG_IBMLS=m # CONFIG_3C359 is not set CONFIG_TMS380TR=m CONFIG_TMSPCI=m CONFIG_SKISA=m CONFIG_PROTEON=m CONFIG_ABYSS=m CONFIG_SMCTR=m # # Wireless LAN # CONFIG_WLAN_PRE80211=y CONFIG_STRIP=m CONFIG_ARLAN=m CONFIG_WAVELAN=m CONFIG_PCMCIA_WAVELAN=m CONFIG_PCMCIA_NETWAVE=m CONFIG_WLAN_80211=y CONFIG_PCMCIA_RAYCS=m CONFIG_IPW2100=m CONFIG_IPW2100_MONITOR=y # CONFIG_IPW2100_DEBUG is not set CONFIG_IPW2200=m CONFIG_IPW2200_MONITOR=y # CONFIG_IPW2200_RADIOTAP is not set # CONFIG_IPW2200_PROMISCUOUS is not set CONFIG_IPW2200_QOS=y # CONFIG_IPW2200_DEBUG is not set CONFIG_LIBERTAS=m CONFIG_LIBERTAS_USB=m # CONFIG_LIBERTAS_CS is not set # CONFIG_LIBERTAS_SDIO is not set # CONFIG_LIBERTAS_DEBUG is not set CONFIG_AIRO=m CONFIG_HERMES=m CONFIG_PLX_HERMES=m CONFIG_TMD_HERMES=m CONFIG_NORTEL_HERMES=m CONFIG_PCI_HERMES=m CONFIG_PCMCIA_HERMES=m CONFIG_PCMCIA_SPECTRUM=m CONFIG_ATMEL=m CONFIG_PCI_ATMEL=m CONFIG_PCMCIA_ATMEL=m CONFIG_AIRO_CS=m CONFIG_PCMCIA_WL3501=m CONFIG_PRISM54=m CONFIG_USB_ZD1201=m # CONFIG_RTL8187 is not set # CONFIG_ADM8211 is not set # CONFIG_P54_COMMON is not set # CONFIG_IWLWIFI is not set CONFIG_HOSTAP=m CONFIG_HOSTAP_FIRMWARE=y # CONFIG_HOSTAP_FIRMWARE_NVRAM is not set CONFIG_HOSTAP_PLX=m CONFIG_HOSTAP_PCI=m CONFIG_HOSTAP_CS=m CONFIG_BCM43XX=m CONFIG_BCM43XX_DEBUG=y CONFIG_BCM43XX_DMA=y CONFIG_BCM43XX_PIO=y CONFIG_BCM43XX_DMA_AND_PIO_MODE=y # CONFIG_BCM43XX_DMA_MODE is not set # CONFIG_BCM43XX_PIO_MODE is not set # CONFIG_B43 is not set # CONFIG_B43LEGACY is not set CONFIG_ZD1211RW=m # CONFIG_ZD1211RW_DEBUG is not set # CONFIG_RT2X00 is not set # # USB Network Adapters # CONFIG_USB_CATC=m CONFIG_USB_KAWETH=m CONFIG_USB_PEGASUS=m CONFIG_USB_RTL8150=m CONFIG_USB_USBNET=m CONFIG_USB_NET_AX8817X=m CONFIG_USB_NET_CDCETHER=m CONFIG_USB_NET_DM9601=m CONFIG_USB_NET_GL620A=m CONFIG_USB_NET_NET1080=m CONFIG_USB_NET_PLUSB=m CONFIG_USB_NET_MCS7830=m CONFIG_USB_NET_RNDIS_HOST=m CONFIG_USB_NET_CDC_SUBSET=m CONFIG_USB_ALI_M5632=y CONFIG_USB_AN2720=y CONFIG_USB_BELKIN=y CONFIG_USB_ARMLINUX=y CONFIG_USB_EPSON2888=y CONFIG_USB_KC2190=y CONFIG_USB_NET_ZAURUS=m CONFIG_NET_PCMCIA=y CONFIG_PCMCIA_3C589=m CONFIG_PCMCIA_3C574=m CONFIG_PCMCIA_FMVJ18X=m CONFIG_PCMCIA_PCNET=m CONFIG_PCMCIA_NMCLAN=m CONFIG_PCMCIA_SMC91C92=m CONFIG_PCMCIA_XIRC2PS=m CONFIG_PCMCIA_AXNET=m CONFIG_ARCNET_COM20020_CS=m CONFIG_PCMCIA_IBMTR=m CONFIG_WAN=y CONFIG_HOSTESS_SV11=m CONFIG_COSA=m CONFIG_LANMEDIA=m CONFIG_SEALEVEL_4021=m CONFIG_HDLC=m CONFIG_HDLC_RAW=m CONFIG_HDLC_RAW_ETH=m CONFIG_HDLC_CISCO=m CONFIG_HDLC_FR=m CONFIG_HDLC_PPP=m CONFIG_HDLC_X25=m CONFIG_PCI200SYN=m CONFIG_WANXL=m CONFIG_PC300=m CONFIG_PC300_MLPPP=y # # Cyclades-PC300 MLPPP support is disabled. # # # Refer to the file README.mlppp, provided by PC300 package. # # CONFIG_PC300TOO is not set CONFIG_N2=m CONFIG_C101=m CONFIG_FARSYNC=m CONFIG_DSCC4=m CONFIG_DSCC4_PCISYNC=y CONFIG_DSCC4_PCI_RST=y CONFIG_DLCI=m CONFIG_DLCI_MAX=8 CONFIG_SDLA=m CONFIG_WAN_ROUTER_DRIVERS=m CONFIG_CYCLADES_SYNC=m CONFIG_CYCLOMX_X25=y CONFIG_LAPBETHER=m CONFIG_X25_ASY=m CONFIG_SBNI=m # CONFIG_SBNI_MULTILINE is not set CONFIG_ATM_DRIVERS=y CONFIG_ATM_DUMMY=m CONFIG_ATM_TCP=m CONFIG_ATM_LANAI=m CONFIG_ATM_ENI=m # CONFIG_ATM_ENI_DEBUG is not set # CONFIG_ATM_ENI_TUNE_BURST is not set CONFIG_ATM_FIRESTREAM=m CONFIG_ATM_ZATM=m # CONFIG_ATM_ZATM_DEBUG is not set CONFIG_ATM_NICSTAR=m # CONFIG_ATM_NICSTAR_USE_SUNI is not set # CONFIG_ATM_NICSTAR_USE_IDT77105 is not set CONFIG_ATM_IDT77252=m # CONFIG_ATM_IDT77252_DEBUG is not set # CONFIG_ATM_IDT77252_RCV_ALL is not set CONFIG_ATM_IDT77252_USE_SUNI=y CONFIG_ATM_AMBASSADOR=m # CONFIG_ATM_AMBASSADOR_DEBUG is not set CONFIG_ATM_HORIZON=m # CONFIG_ATM_HORIZON_DEBUG is not set CONFIG_ATM_IA=m # CONFIG_ATM_IA_DEBUG is not set CONFIG_ATM_FORE200E_MAYBE=m CONFIG_ATM_FORE200E_PCA=y CONFIG_ATM_FORE200E_PCA_DEFAULT_FW=y # CONFIG_ATM_FORE200E_USE_TASKLET is not set CONFIG_ATM_FORE200E_TX_RETRY=16 CONFIG_ATM_FORE200E_DEBUG=0 CONFIG_ATM_FORE200E=m CONFIG_ATM_HE=m CONFIG_ATM_HE_USE_SUNI=y CONFIG_FDDI=y CONFIG_DEFXX=m # CONFIG_DEFXX_MMIO is not set CONFIG_SKFP=m CONFIG_HIPPI=y CONFIG_ROADRUNNER=m # CONFIG_ROADRUNNER_LARGE_RINGS is not set CONFIG_PLIP=m CONFIG_PPP=m CONFIG_PPP_MULTILINK=y CONFIG_PPP_FILTER=y CONFIG_PPP_ASYNC=m CONFIG_PPP_SYNC_TTY=m CONFIG_PPP_DEFLATE=m CONFIG_PPP_BSDCOMP=m CONFIG_PPP_MPPE=m CONFIG_PPPOE=m CONFIG_PPPOATM=m # CONFIG_PPPOL2TP is not set CONFIG_SLIP=m CONFIG_SLIP_COMPRESSED=y CONFIG_SLHC=m CONFIG_SLIP_SMART=y CONFIG_SLIP_MODE_SLIP6=y CONFIG_NET_FC=y CONFIG_SHAPER=m CONFIG_NETCONSOLE=m # CONFIG_NETCONSOLE_DYNAMIC is not set CONFIG_NETPOLL=y # CONFIG_NETPOLL_TRAP is not set CONFIG_NET_POLL_CONTROLLER=y CONFIG_ISDN=m CONFIG_ISDN_I4L=m CONFIG_ISDN_PPP=y CONFIG_ISDN_PPP_VJ=y CONFIG_ISDN_MPP=y CONFIG_IPPP_FILTER=y CONFIG_ISDN_PPP_BSDCOMP=m CONFIG_ISDN_AUDIO=y CONFIG_ISDN_TTY_FAX=y CONFIG_ISDN_X25=y # # ISDN feature submodules # # CONFIG_ISDN_DIVERSION is not set # # ISDN4Linux hardware drivers # # # Passive cards # CONFIG_ISDN_DRV_HISAX=m # # D-channel protocol features # CONFIG_HISAX_EURO=y CONFIG_DE_AOC=y # CONFIG_HISAX_NO_SENDCOMPLETE is not set # CONFIG_HISAX_NO_LLC is not set # CONFIG_HISAX_NO_KEYPAD is not set CONFIG_HISAX_1TR6=y CONFIG_HISAX_NI1=y CONFIG_HISAX_MAX_CARDS=8 # # HiSax supported cards # CONFIG_HISAX_16_0=y CONFIG_HISAX_16_3=y CONFIG_HISAX_TELESPCI=y CONFIG_HISAX_S0BOX=y CONFIG_HISAX_AVM_A1=y CONFIG_HISAX_FRITZPCI=y CONFIG_HISAX_AVM_A1_PCMCIA=y CONFIG_HISAX_ELSA=y CONFIG_HISAX_IX1MICROR2=y CONFIG_HISAX_DIEHLDIVA=y CONFIG_HISAX_ASUSCOM=y CONFIG_HISAX_TELEINT=y CONFIG_HISAX_HFCS=y CONFIG_HISAX_SEDLBAUER=y CONFIG_HISAX_SPORTSTER=y CONFIG_HISAX_MIC=y CONFIG_HISAX_NETJET=y CONFIG_HISAX_NETJET_U=y CONFIG_HISAX_NICCY=y CONFIG_HISAX_ISURF=y CONFIG_HISAX_HSTSAPHIR=y CONFIG_HISAX_BKM_A4T=y CONFIG_HISAX_SCT_QUADRO=y CONFIG_HISAX_GAZEL=y CONFIG_HISAX_HFC_PCI=y CONFIG_HISAX_W6692=y CONFIG_HISAX_HFC_SX=y CONFIG_HISAX_ENTERNOW_PCI=y # CONFIG_HISAX_DEBUG is not set # # HiSax PCMCIA card service modules # CONFIG_HISAX_SEDLBAUER_CS=m CONFIG_HISAX_ELSA_CS=m CONFIG_HISAX_AVM_A1_CS=m CONFIG_HISAX_TELES_CS=m # # HiSax sub driver modules # CONFIG_HISAX_ST5481=m CONFIG_HISAX_HFCUSB=m CONFIG_HISAX_HFC4S8S=m CONFIG_HISAX_FRITZ_PCIPNP=m CONFIG_HISAX_HDLC=y # # Active cards # CONFIG_ISDN_DRV_ICN=m CONFIG_ISDN_DRV_PCBIT=m CONFIG_ISDN_DRV_SC=m CONFIG_ISDN_DRV_ACT2000=m CONFIG_ISDN_DRV_GIGASET=m CONFIG_GIGASET_BASE=m CONFIG_GIGASET_M105=m CONFIG_GIGASET_M101=m # CONFIG_GIGASET_DEBUG is not set # CONFIG_GIGASET_UNDOCREQ is not set CONFIG_ISDN_CAPI=m CONFIG_ISDN_DRV_AVMB1_VERBOSE_REASON=y CONFIG_CAPI_TRACE=y CONFIG_ISDN_CAPI_MIDDLEWARE=y CONFIG_ISDN_CAPI_CAPI20=m CONFIG_ISDN_CAPI_CAPIFS_BOOL=y CONFIG_ISDN_CAPI_CAPIFS=m CONFIG_ISDN_CAPI_CAPIDRV=m # # CAPI hardware drivers # CONFIG_CAPI_AVM=y CONFIG_ISDN_DRV_AVMB1_B1ISA=m CONFIG_ISDN_DRV_AVMB1_B1PCI=m CONFIG_ISDN_DRV_AVMB1_B1PCIV4=y CONFIG_ISDN_DRV_AVMB1_T1ISA=m CONFIG_ISDN_DRV_AVMB1_B1PCMCIA=m CONFIG_ISDN_DRV_AVMB1_AVM_CS=m CONFIG_ISDN_DRV_AVMB1_T1PCI=m CONFIG_ISDN_DRV_AVMB1_C4=m CONFIG_CAPI_EICON=y CONFIG_ISDN_DIVAS=m CONFIG_ISDN_DIVAS_BRIPCI=y CONFIG_ISDN_DIVAS_PRIPCI=y CONFIG_ISDN_DIVAS_DIVACAPI=m CONFIG_ISDN_DIVAS_USERIDI=m CONFIG_ISDN_DIVAS_MAINT=m CONFIG_PHONE=m CONFIG_PHONE_IXJ=m CONFIG_PHONE_IXJ_PCMCIA=m # # Input device support # CONFIG_INPUT=y CONFIG_INPUT_FF_MEMLESS=m CONFIG_INPUT_POLLDEV=m # # Userland interfaces # CONFIG_INPUT_MOUSEDEV=y CONFIG_INPUT_MOUSEDEV_PSAUX=y CONFIG_INPUT_MOUSEDEV_SCREEN_X=1024 CONFIG_INPUT_MOUSEDEV_SCREEN_Y=768 CONFIG_INPUT_JOYDEV=m CONFIG_INPUT_EVDEV=m # CONFIG_INPUT_EVBUG is not set # # Input Device Drivers # CONFIG_INPUT_KEYBOARD=y CONFIG_KEYBOARD_ATKBD=y CONFIG_KEYBOARD_SUNKBD=m CONFIG_KEYBOARD_LKKBD=m CONFIG_KEYBOARD_XTKBD=m CONFIG_KEYBOARD_NEWTON=m CONFIG_KEYBOARD_STOWAWAY=m CONFIG_INPUT_MOUSE=y CONFIG_MOUSE_PS2=m CONFIG_MOUSE_PS2_ALPS=y CONFIG_MOUSE_PS2_LOGIPS2PP=y CONFIG_MOUSE_PS2_SYNAPTICS=y CONFIG_MOUSE_PS2_LIFEBOOK=y CONFIG_MOUSE_PS2_TRACKPOINT=y # CONFIG_MOUSE_PS2_TOUCHKIT is not set CONFIG_MOUSE_SERIAL=m CONFIG_MOUSE_APPLETOUCH=m CONFIG_MOUSE_INPORT=m # CONFIG_MOUSE_ATIXL is not set CONFIG_MOUSE_LOGIBM=m CONFIG_MOUSE_PC110PAD=m CONFIG_MOUSE_VSXXXAA=m CONFIG_INPUT_JOYSTICK=y CONFIG_JOYSTICK_ANALOG=m CONFIG_JOYSTICK_A3D=m CONFIG_JOYSTICK_ADI=m CONFIG_JOYSTICK_COBRA=m CONFIG_JOYSTICK_GF2K=m CONFIG_JOYSTICK_GRIP=m CONFIG_JOYSTICK_GRIP_MP=m CONFIG_JOYSTICK_GUILLEMOT=m CONFIG_JOYSTICK_INTERACT=m CONFIG_JOYSTICK_SIDEWINDER=m CONFIG_JOYSTICK_TMDC=m CONFIG_JOYSTICK_IFORCE=m CONFIG_JOYSTICK_IFORCE_USB=y CONFIG_JOYSTICK_IFORCE_232=y CONFIG_JOYSTICK_WARRIOR=m CONFIG_JOYSTICK_MAGELLAN=m CONFIG_JOYSTICK_SPACEORB=m CONFIG_JOYSTICK_SPACEBALL=m CONFIG_JOYSTICK_STINGER=m CONFIG_JOYSTICK_TWIDJOY=m CONFIG_JOYSTICK_DB9=m CONFIG_JOYSTICK_GAMECON=m CONFIG_JOYSTICK_TURBOGRAFX=m CONFIG_JOYSTICK_JOYDUMP=m CONFIG_JOYSTICK_XPAD=m # CONFIG_JOYSTICK_XPAD_FF is not set # CONFIG_JOYSTICK_XPAD_LEDS is not set CONFIG_INPUT_TABLET=y CONFIG_TABLET_USB_ACECAD=m CONFIG_TABLET_USB_AIPTEK=m CONFIG_TABLET_USB_GTCO=m CONFIG_TABLET_USB_KBTAB=m CONFIG_TABLET_USB_WACOM=m CONFIG_INPUT_TOUCHSCREEN=y CONFIG_TOUCHSCREEN_ADS7846=m # CONFIG_TOUCHSCREEN_FUJITSU is not set CONFIG_TOUCHSCREEN_GUNZE=m CONFIG_TOUCHSCREEN_ELO=m CONFIG_TOUCHSCREEN_MTOUCH=m CONFIG_TOUCHSCREEN_MK712=m CONFIG_TOUCHSCREEN_PENMOUNT=m CONFIG_TOUCHSCREEN_TOUCHRIGHT=m CONFIG_TOUCHSCREEN_TOUCHWIN=m CONFIG_TOUCHSCREEN_UCB1400=m CONFIG_TOUCHSCREEN_USB_COMPOSITE=m CONFIG_TOUCHSCREEN_USB_EGALAX=y CONFIG_TOUCHSCREEN_USB_PANJIT=y CONFIG_TOUCHSCREEN_USB_3M=y CONFIG_TOUCHSCREEN_USB_ITM=y CONFIG_TOUCHSCREEN_USB_ETURBO=y CONFIG_TOUCHSCREEN_USB_GUNZE=y CONFIG_TOUCHSCREEN_USB_DMC_TSC10=y CONFIG_TOUCHSCREEN_USB_IRTOUCH=y CONFIG_TOUCHSCREEN_USB_IDEALTEK=y CONFIG_TOUCHSCREEN_USB_GENERAL_TOUCH=y CONFIG_TOUCHSCREEN_USB_GOTOP=y CONFIG_INPUT_MISC=y CONFIG_INPUT_PCSPKR=m CONFIG_INPUT_WISTRON_BTNS=m CONFIG_INPUT_ATLAS_BTNS=m CONFIG_INPUT_ATI_REMOTE=m CONFIG_INPUT_ATI_REMOTE2=m CONFIG_INPUT_KEYSPAN_REMOTE=m CONFIG_INPUT_POWERMATE=m CONFIG_INPUT_YEALINK=m CONFIG_INPUT_UINPUT=m # # Hardware I/O ports # CONFIG_SERIO=y CONFIG_SERIO_I8042=y CONFIG_SERIO_SERPORT=m CONFIG_SERIO_CT82C710=m CONFIG_SERIO_PARKBD=m CONFIG_SERIO_PCIPS2=m CONFIG_SERIO_LIBPS2=y CONFIG_SERIO_RAW=m CONFIG_GAMEPORT=m CONFIG_GAMEPORT_NS558=m CONFIG_GAMEPORT_L4=m CONFIG_GAMEPORT_EMU10K1=m CONFIG_GAMEPORT_FM801=m # # Character devices # CONFIG_VT=y CONFIG_VT_CONSOLE=y CONFIG_HW_CONSOLE=y # CONFIG_VT_HW_CONSOLE_BINDING is not set CONFIG_SERIAL_NONSTANDARD=y # CONFIG_COMPUTONE is not set CONFIG_ROCKETPORT=m CONFIG_CYCLADES=m # CONFIG_CYZ_INTR is not set # CONFIG_DIGIEPCA is not set # CONFIG_ESPSERIAL is not set # CONFIG_MOXA_INTELLIO is not set CONFIG_MOXA_SMARTIO=m CONFIG_MOXA_SMARTIO_NEW=m # CONFIG_ISI is not set CONFIG_SYNCLINK=m CONFIG_SYNCLINKMP=m CONFIG_SYNCLINK_GT=m CONFIG_N_HDLC=m # CONFIG_SPECIALIX is not set CONFIG_SX=m # CONFIG_RIO is not set CONFIG_STALDRV=y # # Serial drivers # CONFIG_SERIAL_8250=y CONFIG_SERIAL_8250_CONSOLE=y CONFIG_FIX_EARLYCON_MEM=y CONFIG_SERIAL_8250_PCI=y CONFIG_SERIAL_8250_PNP=y CONFIG_SERIAL_8250_CS=m CONFIG_SERIAL_8250_NR_UARTS=16 CONFIG_SERIAL_8250_RUNTIME_UARTS=4 CONFIG_SERIAL_8250_EXTENDED=y CONFIG_SERIAL_8250_MANY_PORTS=y CONFIG_SERIAL_8250_FOURPORT=m CONFIG_SERIAL_8250_ACCENT=m CONFIG_SERIAL_8250_BOCA=m CONFIG_SERIAL_8250_EXAR_ST16C554=m CONFIG_SERIAL_8250_HUB6=m CONFIG_SERIAL_8250_SHARE_IRQ=y # CONFIG_SERIAL_8250_DETECT_IRQ is not set CONFIG_SERIAL_8250_RSA=y # # Non-8250 serial port support # CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y CONFIG_SERIAL_JSM=m CONFIG_UNIX98_PTYS=y # CONFIG_LEGACY_PTYS is not set CONFIG_PRINTER=m # CONFIG_LP_CONSOLE is not set CONFIG_PPDEV=m CONFIG_IPMI_HANDLER=m # CONFIG_IPMI_PANIC_EVENT is not set CONFIG_IPMI_DEVICE_INTERFACE=m CONFIG_IPMI_SI=m CONFIG_IPMI_WATCHDOG=m CONFIG_IPMI_POWEROFF=m CONFIG_HW_RANDOM=y CONFIG_HW_RANDOM_INTEL=m CONFIG_HW_RANDOM_AMD=m CONFIG_HW_RANDOM_GEODE=m CONFIG_HW_RANDOM_VIA=m CONFIG_NVRAM=m CONFIG_RTC=m CONFIG_GEN_RTC=m CONFIG_GEN_RTC_X=y CONFIG_DTLK=m CONFIG_R3964=m CONFIG_APPLICOM=m CONFIG_SONYPI=m # # PCMCIA character devices # CONFIG_SYNCLINK_CS=m CONFIG_CARDMAN_4000=m CONFIG_CARDMAN_4040=m CONFIG_MWAVE=m CONFIG_SCx200_GPIO=m CONFIG_PC8736x_GPIO=m CONFIG_NSC_GPIO=m CONFIG_CS5535_GPIO=m CONFIG_RAW_DRIVER=m CONFIG_MAX_RAW_DEVS=256 CONFIG_HPET=y # CONFIG_HPET_RTC_IRQ is not set CONFIG_HPET_MMAP=y CONFIG_HANGCHECK_TIMER=m # CONFIG_TCG_TPM is not set CONFIG_TELCLOCK=m CONFIG_DEVPORT=y CONFIG_I2C=m CONFIG_I2C_BOARDINFO=y CONFIG_I2C_CHARDEV=m # # I2C Algorithms # CONFIG_I2C_ALGOBIT=m CONFIG_I2C_ALGOPCF=m CONFIG_I2C_ALGOPCA=m # # I2C Hardware Bus support # CONFIG_I2C_ALI1535=m CONFIG_I2C_ALI1563=m CONFIG_I2C_ALI15X3=m CONFIG_I2C_AMD756=m CONFIG_I2C_AMD756_S4882=m CONFIG_I2C_AMD8111=m CONFIG_I2C_I801=m CONFIG_I2C_I810=m CONFIG_I2C_PIIX4=m CONFIG_I2C_NFORCE2=m CONFIG_I2C_OCORES=m CONFIG_I2C_PARPORT=m CONFIG_I2C_PARPORT_LIGHT=m CONFIG_I2C_PROSAVAGE=m CONFIG_I2C_SAVAGE4=m CONFIG_I2C_SIMTEC=m CONFIG_SCx200_I2C=m CONFIG_SCx200_I2C_SCL=12 CONFIG_SCx200_I2C_SDA=13 CONFIG_SCx200_ACB=m CONFIG_I2C_SIS5595=m CONFIG_I2C_SIS630=m CONFIG_I2C_SIS96X=m # CONFIG_I2C_TAOS_EVM is not set CONFIG_I2C_STUB=m CONFIG_I2C_TINY_USB=m CONFIG_I2C_VIA=m CONFIG_I2C_VIAPRO=m CONFIG_I2C_VOODOO3=m CONFIG_I2C_PCA_ISA=m # # Miscellaneous I2C Chip support # CONFIG_SENSORS_DS1337=m CONFIG_SENSORS_DS1374=m # CONFIG_DS1682 is not set CONFIG_SENSORS_EEPROM=m CONFIG_SENSORS_PCF8574=m CONFIG_SENSORS_PCA9539=m CONFIG_SENSORS_PCF8591=m CONFIG_SENSORS_MAX6875=m # CONFIG_SENSORS_TSL2550 is not set # CONFIG_I2C_DEBUG_CORE is not set # CONFIG_I2C_DEBUG_ALGO is not set # CONFIG_I2C_DEBUG_BUS is not set # CONFIG_I2C_DEBUG_CHIP is not set # # SPI support # CONFIG_SPI=y # CONFIG_SPI_DEBUG is not set CONFIG_SPI_MASTER=y # # SPI Master Controller Drivers # CONFIG_SPI_BITBANG=m CONFIG_SPI_BUTTERFLY=m # CONFIG_SPI_LM70_LLP is not set # # SPI Protocol Masters # CONFIG_SPI_AT25=m # CONFIG_SPI_SPIDEV is not set # CONFIG_SPI_TLE62X0 is not set CONFIG_W1=m CONFIG_W1_CON=y # # 1-wire Bus Masters # CONFIG_W1_MASTER_MATROX=m CONFIG_W1_MASTER_DS2490=m CONFIG_W1_MASTER_DS2482=m # # 1-wire Slaves # CONFIG_W1_SLAVE_THERM=m CONFIG_W1_SLAVE_SMEM=m CONFIG_W1_SLAVE_DS2433=m # CONFIG_W1_SLAVE_DS2433_CRC is not set # CONFIG_W1_SLAVE_DS2760 is not set CONFIG_POWER_SUPPLY=y # CONFIG_POWER_SUPPLY_DEBUG is not set # CONFIG_PDA_POWER is not set # CONFIG_BATTERY_DS2760 is not set CONFIG_HWMON=y CONFIG_HWMON_VID=m CONFIG_SENSORS_ABITUGURU=m # CONFIG_SENSORS_ABITUGURU3 is not set CONFIG_SENSORS_AD7418=m CONFIG_SENSORS_ADM1021=m CONFIG_SENSORS_ADM1025=m CONFIG_SENSORS_ADM1026=m CONFIG_SENSORS_ADM1029=m CONFIG_SENSORS_ADM1031=m CONFIG_SENSORS_ADM9240=m # CONFIG_SENSORS_ADT7470 is not set CONFIG_SENSORS_K8TEMP=m CONFIG_SENSORS_ASB100=m CONFIG_SENSORS_ATXP1=m CONFIG_SENSORS_DS1621=m # CONFIG_SENSORS_I5K_AMB is not set CONFIG_SENSORS_F71805F=m # CONFIG_SENSORS_F71882FG is not set CONFIG_SENSORS_F75375S=m CONFIG_SENSORS_FSCHER=m CONFIG_SENSORS_FSCPOS=m # CONFIG_SENSORS_FSCHMD is not set CONFIG_SENSORS_GL518SM=m CONFIG_SENSORS_GL520SM=m CONFIG_SENSORS_CORETEMP=m # CONFIG_SENSORS_IBMPEX is not set CONFIG_SENSORS_IT87=m CONFIG_SENSORS_LM63=m CONFIG_SENSORS_LM70=m CONFIG_SENSORS_LM75=m CONFIG_SENSORS_LM77=m CONFIG_SENSORS_LM78=m CONFIG_SENSORS_LM80=m CONFIG_SENSORS_LM83=m CONFIG_SENSORS_LM85=m CONFIG_SENSORS_LM87=m CONFIG_SENSORS_LM90=m CONFIG_SENSORS_LM92=m # CONFIG_SENSORS_LM93 is not set CONFIG_SENSORS_MAX1619=m CONFIG_SENSORS_MAX6650=m CONFIG_SENSORS_PC87360=m CONFIG_SENSORS_PC87427=m CONFIG_SENSORS_SIS5595=m # CONFIG_SENSORS_DME1737 is not set CONFIG_SENSORS_SMSC47M1=m CONFIG_SENSORS_SMSC47M192=m CONFIG_SENSORS_SMSC47B397=m # CONFIG_SENSORS_THMC50 is not set CONFIG_SENSORS_VIA686A=m CONFIG_SENSORS_VT1211=m CONFIG_SENSORS_VT8231=m CONFIG_SENSORS_W83781D=m CONFIG_SENSORS_W83791D=m CONFIG_SENSORS_W83792D=m CONFIG_SENSORS_W83793=m CONFIG_SENSORS_W83L785TS=m CONFIG_SENSORS_W83627HF=m CONFIG_SENSORS_W83627EHF=m CONFIG_SENSORS_HDAPS=m CONFIG_SENSORS_APPLESMC=m # CONFIG_HWMON_DEBUG_CHIP is not set CONFIG_WATCHDOG=y # CONFIG_WATCHDOG_NOWAYOUT is not set # # Watchdog Device Drivers # CONFIG_SOFT_WATCHDOG=m CONFIG_ACQUIRE_WDT=m CONFIG_ADVANTECH_WDT=m CONFIG_ALIM1535_WDT=m CONFIG_ALIM7101_WDT=m CONFIG_SC520_WDT=m CONFIG_EUROTECH_WDT=m CONFIG_IB700_WDT=m CONFIG_IBMASR=m CONFIG_WAFER_WDT=m CONFIG_I6300ESB_WDT=m CONFIG_ITCO_WDT=m # CONFIG_ITCO_VENDOR_SUPPORT is not set # CONFIG_IT8712F_WDT is not set CONFIG_SC1200_WDT=m CONFIG_SCx200_WDT=m CONFIG_PC87413_WDT=m CONFIG_60XX_WDT=m CONFIG_SBC8360_WDT=m # CONFIG_SBC7240_WDT is not set CONFIG_CPU5_WDT=m CONFIG_SMSC37B787_WDT=m CONFIG_W83627HF_WDT=m CONFIG_W83697HF_WDT=m CONFIG_W83877F_WDT=m CONFIG_W83977F_WDT=m CONFIG_MACHZ_WDT=m CONFIG_SBC_EPX_C3_WATCHDOG=m # # ISA-based Watchdog Cards # CONFIG_PCWATCHDOG=m CONFIG_MIXCOMWD=m CONFIG_WDT=m CONFIG_WDT_501=y # # PCI-based Watchdog Cards # CONFIG_PCIPCWATCHDOG=m CONFIG_WDTPCI=m CONFIG_WDT_501_PCI=y # # USB-based Watchdog Cards # CONFIG_USBPCWATCHDOG=m # # Sonics Silicon Backplane # CONFIG_SSB_POSSIBLE=y CONFIG_SSB=m CONFIG_SSB_PCIHOST_POSSIBLE=y CONFIG_SSB_PCIHOST=y CONFIG_SSB_PCMCIAHOST_POSSIBLE=y # CONFIG_SSB_PCMCIAHOST is not set # CONFIG_SSB_DEBUG is not set CONFIG_SSB_DRIVER_PCICORE_POSSIBLE=y CONFIG_SSB_DRIVER_PCICORE=y # # Multifunction device drivers # CONFIG_MFD_SM501=m # # Multimedia devices # CONFIG_VIDEO_DEV=m CONFIG_VIDEO_V4L1=y CONFIG_VIDEO_V4L1_COMPAT=y CONFIG_VIDEO_V4L2=y CONFIG_VIDEO_CAPTURE_DRIVERS=y # CONFIG_VIDEO_ADV_DEBUG is not set CONFIG_VIDEO_HELPER_CHIPS_AUTO=y CONFIG_VIDEO_TVAUDIO=m CONFIG_VIDEO_TDA7432=m CONFIG_VIDEO_TDA9840=m CONFIG_VIDEO_TDA9875=m CONFIG_VIDEO_TEA6415C=m CONFIG_VIDEO_TEA6420=m CONFIG_VIDEO_MSP3400=m CONFIG_VIDEO_CS53L32A=m CONFIG_VIDEO_WM8775=m CONFIG_VIDEO_WM8739=m CONFIG_VIDEO_VP27SMPX=m CONFIG_VIDEO_BT819=m CONFIG_VIDEO_BT856=m CONFIG_VIDEO_KS0127=m CONFIG_VIDEO_OV7670=m CONFIG_VIDEO_SAA7110=m CONFIG_VIDEO_SAA7111=m CONFIG_VIDEO_SAA7114=m CONFIG_VIDEO_SAA711X=m CONFIG_VIDEO_TVP5150=m CONFIG_VIDEO_VPX3220=m CONFIG_VIDEO_CX25840=m CONFIG_VIDEO_CX2341X=m CONFIG_VIDEO_SAA7127=m CONFIG_VIDEO_SAA7185=m CONFIG_VIDEO_ADV7170=m CONFIG_VIDEO_ADV7175=m CONFIG_VIDEO_UPD64031A=m CONFIG_VIDEO_UPD64083=m CONFIG_VIDEO_VIVI=m CONFIG_VIDEO_BT848=m CONFIG_VIDEO_BT848_DVB=y CONFIG_VIDEO_SAA6588=m CONFIG_VIDEO_PMS=m CONFIG_VIDEO_BWQCAM=m CONFIG_VIDEO_CQCAM=m CONFIG_VIDEO_W9966=m CONFIG_VIDEO_CPIA=m CONFIG_VIDEO_CPIA_PP=m CONFIG_VIDEO_CPIA_USB=m CONFIG_VIDEO_CPIA2=m CONFIG_VIDEO_SAA5246A=m CONFIG_VIDEO_SAA5249=m CONFIG_TUNER_3036=m CONFIG_VIDEO_STRADIS=m CONFIG_VIDEO_ZORAN_ZR36060=m CONFIG_VIDEO_ZORAN=m CONFIG_VIDEO_ZORAN_BUZ=m CONFIG_VIDEO_ZORAN_DC10=m CONFIG_VIDEO_ZORAN_DC30=m CONFIG_VIDEO_ZORAN_LML33=m CONFIG_VIDEO_ZORAN_LML33R10=m CONFIG_VIDEO_ZORAN_AVS6EYES=m CONFIG_VIDEO_MEYE=m CONFIG_VIDEO_SAA7134=m CONFIG_VIDEO_SAA7134_ALSA=m CONFIG_VIDEO_SAA7134_OSS=m CONFIG_VIDEO_SAA7134_DVB=m CONFIG_VIDEO_MXB=m CONFIG_VIDEO_DPC=m CONFIG_VIDEO_HEXIUM_ORION=m CONFIG_VIDEO_HEXIUM_GEMINI=m CONFIG_VIDEO_CX88=m CONFIG_VIDEO_CX88_ALSA=m CONFIG_VIDEO_CX88_BLACKBIRD=m CONFIG_VIDEO_CX88_DVB=m CONFIG_VIDEO_CX88_VP3054=m # CONFIG_VIDEO_CX23885 is not set CONFIG_VIDEO_IVTV=m # CONFIG_VIDEO_FB_IVTV is not set CONFIG_VIDEO_CAFE_CCIC=m CONFIG_V4L_USB_DRIVERS=y CONFIG_VIDEO_PVRUSB2=m CONFIG_VIDEO_PVRUSB2_29XXX=y CONFIG_VIDEO_PVRUSB2_24XXX=y CONFIG_VIDEO_PVRUSB2_SYSFS=y # CONFIG_VIDEO_PVRUSB2_DEBUGIFC is not set CONFIG_VIDEO_EM28XX=m CONFIG_VIDEO_USBVISION=m CONFIG_VIDEO_USBVIDEO=m CONFIG_USB_VICAM=m CONFIG_USB_IBMCAM=m CONFIG_USB_KONICAWC=m CONFIG_USB_QUICKCAM_MESSENGER=m CONFIG_USB_ET61X251=m CONFIG_VIDEO_OVCAMCHIP=m # CONFIG_USB_W9968CF is not set CONFIG_USB_OV511=m CONFIG_USB_SE401=m CONFIG_USB_SN9C102=m CONFIG_USB_STV680=m CONFIG_USB_ZC0301=m CONFIG_USB_PWC=m # CONFIG_USB_PWC_DEBUG is not set CONFIG_USB_ZR364XX=m CONFIG_RADIO_ADAPTERS=y CONFIG_RADIO_CADET=m CONFIG_RADIO_RTRACK=m CONFIG_RADIO_RTRACK2=m CONFIG_RADIO_AZTECH=m CONFIG_RADIO_GEMTEK=m CONFIG_RADIO_GEMTEK_PCI=m CONFIG_RADIO_MAXIRADIO=m CONFIG_RADIO_MAESTRO=m CONFIG_RADIO_SF16FMI=m CONFIG_RADIO_SF16FMR2=m CONFIG_RADIO_TERRATEC=m CONFIG_RADIO_TRUST=m CONFIG_RADIO_TYPHOON=m CONFIG_RADIO_TYPHOON_PROC_FS=y CONFIG_RADIO_ZOLTRIX=m CONFIG_USB_DSBR=m CONFIG_DVB_CORE=m CONFIG_DVB_CORE_ATTACH=y CONFIG_DVB_CAPTURE_DRIVERS=y # # Supported SAA7146 based PCI Adapters # CONFIG_DVB_AV7110=m CONFIG_DVB_AV7110_OSD=y CONFIG_DVB_BUDGET=m CONFIG_DVB_BUDGET_CI=m CONFIG_DVB_BUDGET_AV=m CONFIG_DVB_BUDGET_PATCH=m # # Supported USB Adapters # CONFIG_DVB_USB=m # CONFIG_DVB_USB_DEBUG is not set CONFIG_DVB_USB_A800=m CONFIG_DVB_USB_DIBUSB_MB=m CONFIG_DVB_USB_DIBUSB_MB_FAULTY=y CONFIG_DVB_USB_DIBUSB_MC=m CONFIG_DVB_USB_DIB0700=m CONFIG_DVB_USB_UMT_010=m CONFIG_DVB_USB_CXUSB=m CONFIG_DVB_USB_M920X=m CONFIG_DVB_USB_GL861=m CONFIG_DVB_USB_AU6610=m CONFIG_DVB_USB_DIGITV=m CONFIG_DVB_USB_VP7045=m CONFIG_DVB_USB_VP702X=m CONFIG_DVB_USB_GP8PSK=m CONFIG_DVB_USB_NOVA_T_USB2=m CONFIG_DVB_USB_TTUSB2=m CONFIG_DVB_USB_DTT200U=m CONFIG_DVB_USB_OPERA1=m # CONFIG_DVB_USB_AF9005 is not set CONFIG_DVB_TTUSB_BUDGET=m CONFIG_DVB_TTUSB_DEC=m CONFIG_DVB_CINERGYT2=m # CONFIG_DVB_CINERGYT2_TUNING is not set # # Supported FlexCopII (B2C2) Adapters # CONFIG_DVB_B2C2_FLEXCOP=m CONFIG_DVB_B2C2_FLEXCOP_PCI=m CONFIG_DVB_B2C2_FLEXCOP_USB=m # CONFIG_DVB_B2C2_FLEXCOP_DEBUG is not set # # Supported BT878 Adapters # CONFIG_DVB_BT8XX=m # # Supported Pluto2 Adapters # CONFIG_DVB_PLUTO2=m # # Supported DVB Frontends # # # Customise DVB Frontends # # CONFIG_DVB_FE_CUSTOMISE is not set # # DVB-S (satellite) frontends # CONFIG_DVB_STV0299=m CONFIG_DVB_CX24110=m CONFIG_DVB_CX24123=m CONFIG_DVB_TDA8083=m CONFIG_DVB_MT312=m CONFIG_DVB_VES1X93=m CONFIG_DVB_S5H1420=m CONFIG_DVB_TDA10086=m # # DVB-T (terrestrial) frontends # CONFIG_DVB_SP8870=m CONFIG_DVB_SP887X=m CONFIG_DVB_CX22700=m CONFIG_DVB_CX22702=m CONFIG_DVB_L64781=m CONFIG_DVB_TDA1004X=m CONFIG_DVB_NXT6000=m CONFIG_DVB_MT352=m CONFIG_DVB_ZL10353=m CONFIG_DVB_DIB3000MB=m CONFIG_DVB_DIB3000MC=m CONFIG_DVB_DIB7000M=m CONFIG_DVB_DIB7000P=m # # DVB-C (cable) frontends # CONFIG_DVB_VES1820=m CONFIG_DVB_TDA10021=m CONFIG_DVB_TDA10023=m CONFIG_DVB_STV0297=m # # ATSC (North American/Korean Terrestrial/Cable DTV) frontends # CONFIG_DVB_NXT200X=m CONFIG_DVB_OR51211=m CONFIG_DVB_OR51132=m CONFIG_DVB_BCM3510=m CONFIG_DVB_LGDT330X=m # CONFIG_DVB_S5H1409 is not set # # Tuners/PLL support # CONFIG_DVB_PLL=m CONFIG_DVB_TDA826X=m CONFIG_DVB_TDA827X=m CONFIG_DVB_TUNER_QT1010=m CONFIG_DVB_TUNER_MT2060=m CONFIG_DVB_TUNER_MT2266=m # CONFIG_DVB_TUNER_MT2131 is not set CONFIG_DVB_TUNER_DIB0070=m # # Miscellaneous devices # CONFIG_DVB_LNBP21=m CONFIG_DVB_ISL6421=m CONFIG_DVB_TUA6100=m CONFIG_VIDEO_SAA7146=m CONFIG_VIDEO_SAA7146_VV=m CONFIG_VIDEO_TUNER=m # CONFIG_VIDEO_TUNER_CUSTOMIZE is not set CONFIG_TUNER_MT20XX=m CONFIG_TUNER_TDA8290=m CONFIG_TUNER_TEA5761=m CONFIG_TUNER_TEA5767=m CONFIG_TUNER_SIMPLE=m CONFIG_VIDEOBUF_GEN=m CONFIG_VIDEOBUF_DMA_SG=m CONFIG_VIDEOBUF_VMALLOC=m CONFIG_VIDEOBUF_DVB=m CONFIG_VIDEO_BTCX=m CONFIG_VIDEO_IR_I2C=m CONFIG_VIDEO_IR=m CONFIG_VIDEO_TVEEPROM=m CONFIG_DAB=y # CONFIG_USB_DABUSB is not set # # Graphics support # CONFIG_AGP=m CONFIG_AGP_ALI=m CONFIG_AGP_ATI=m CONFIG_AGP_AMD=m CONFIG_AGP_AMD64=m CONFIG_AGP_INTEL=m CONFIG_AGP_NVIDIA=m CONFIG_AGP_SIS=m CONFIG_AGP_SWORKS=m CONFIG_AGP_VIA=m CONFIG_AGP_EFFICEON=m CONFIG_DRM=m CONFIG_DRM_TDFX=m CONFIG_DRM_R128=m CONFIG_DRM_RADEON=m CONFIG_DRM_I810=m CONFIG_DRM_I830=m CONFIG_DRM_I915=m CONFIG_DRM_MGA=m CONFIG_DRM_SIS=m CONFIG_DRM_VIA=m CONFIG_DRM_SAVAGE=m CONFIG_VGASTATE=m # CONFIG_VIDEO_OUTPUT_CONTROL is not set CONFIG_FB=y CONFIG_FIRMWARE_EDID=y CONFIG_FB_DDC=m CONFIG_FB_CFB_FILLRECT=y CONFIG_FB_CFB_COPYAREA=y CONFIG_FB_CFB_IMAGEBLIT=y # CONFIG_FB_CFB_REV_PIXELS_IN_BYTE is not set CONFIG_FB_SYS_FILLRECT=m CONFIG_FB_SYS_COPYAREA=m CONFIG_FB_SYS_IMAGEBLIT=m CONFIG_FB_SYS_FOPS=m CONFIG_FB_DEFERRED_IO=y CONFIG_FB_SVGALIB=m # CONFIG_FB_MACMODES is not set CONFIG_FB_BACKLIGHT=y CONFIG_FB_MODE_HELPERS=y CONFIG_FB_TILEBLITTING=y # # Frame buffer hardware drivers # CONFIG_FB_CIRRUS=m CONFIG_FB_PM2=m CONFIG_FB_PM2_FIFO_DISCONNECT=y CONFIG_FB_CYBER2000=m CONFIG_FB_ARC=m # CONFIG_FB_ASILIANT is not set # CONFIG_FB_IMSTT is not set CONFIG_FB_VGA16=m # CONFIG_FB_UVESA is not set CONFIG_FB_VESA=y # CONFIG_FB_EFI is not set # CONFIG_FB_IMAC is not set CONFIG_FB_HECUBA=m CONFIG_FB_HGA=m # CONFIG_FB_HGA_ACCEL is not set CONFIG_FB_S1D13XXX=m CONFIG_FB_NVIDIA=m CONFIG_FB_NVIDIA_I2C=y # CONFIG_FB_NVIDIA_DEBUG is not set CONFIG_FB_NVIDIA_BACKLIGHT=y # CONFIG_FB_RIVA is not set CONFIG_FB_I810=m # CONFIG_FB_I810_GTF is not set CONFIG_FB_LE80578=m CONFIG_FB_CARILLO_RANCH=m CONFIG_FB_INTEL=m # CONFIG_FB_INTEL_DEBUG is not set CONFIG_FB_INTEL_I2C=y CONFIG_FB_MATROX=m CONFIG_FB_MATROX_MILLENIUM=y CONFIG_FB_MATROX_MYSTIQUE=y CONFIG_FB_MATROX_G=y CONFIG_FB_MATROX_I2C=m CONFIG_FB_MATROX_MAVEN=m CONFIG_FB_MATROX_MULTIHEAD=y CONFIG_FB_RADEON=m CONFIG_FB_RADEON_I2C=y CONFIG_FB_RADEON_BACKLIGHT=y # CONFIG_FB_RADEON_DEBUG is not set CONFIG_FB_ATY128=m CONFIG_FB_ATY128_BACKLIGHT=y CONFIG_FB_ATY=m CONFIG_FB_ATY_CT=y CONFIG_FB_ATY_GENERIC_LCD=y CONFIG_FB_ATY_GX=y CONFIG_FB_ATY_BACKLIGHT=y CONFIG_FB_S3=m CONFIG_FB_SAVAGE=m CONFIG_FB_SAVAGE_I2C=y # CONFIG_FB_SAVAGE_ACCEL is not set CONFIG_FB_SIS=m CONFIG_FB_SIS_300=y CONFIG_FB_SIS_315=y CONFIG_FB_NEOMAGIC=m CONFIG_FB_KYRO=m CONFIG_FB_3DFX=m # CONFIG_FB_3DFX_ACCEL is not set CONFIG_FB_VOODOO1=m CONFIG_FB_VT8623=m CONFIG_FB_CYBLA=m CONFIG_FB_TRIDENT=m # CONFIG_FB_TRIDENT_ACCEL is not set CONFIG_FB_ARK=m CONFIG_FB_PM3=m CONFIG_FB_GEODE=y # CONFIG_FB_GEODE_LX is not set CONFIG_FB_GEODE_GX=m # CONFIG_FB_GEODE_GX_SET_FBSIZE is not set CONFIG_FB_GEODE_GX1=m CONFIG_FB_SM501=m CONFIG_FB_VIRTUAL=m CONFIG_BACKLIGHT_LCD_SUPPORT=y # CONFIG_LCD_CLASS_DEVICE is not set CONFIG_BACKLIGHT_CLASS_DEVICE=y # CONFIG_BACKLIGHT_CORGI is not set CONFIG_BACKLIGHT_PROGEAR=m # # Display device support # CONFIG_DISPLAY_SUPPORT=m # # Display hardware drivers # # # Console display driver support # CONFIG_VGA_CONSOLE=y # CONFIG_VGACON_SOFT_SCROLLBACK is not set CONFIG_VIDEO_SELECT=y CONFIG_MDA_CONSOLE=m CONFIG_DUMMY_CONSOLE=y CONFIG_FRAMEBUFFER_CONSOLE=y # CONFIG_FRAMEBUFFER_CONSOLE_DETECT_PRIMARY is not set CONFIG_FRAMEBUFFER_CONSOLE_ROTATION=y # CONFIG_FONTS is not set CONFIG_FONT_8x8=y CONFIG_FONT_8x16=y # CONFIG_LOGO is not set # # Sound # CONFIG_SOUND=m # # Advanced Linux Sound Architecture # 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_RTCTIMER=m CONFIG_SND_SEQ_RTCTIMER_DEFAULT=y # CONFIG_SND_DYNAMIC_MINORS is not set CONFIG_SND_SUPPORT_OLD_API=y CONFIG_SND_VERBOSE_PROCFS=y # CONFIG_SND_VERBOSE_PRINTK is not set # CONFIG_SND_DEBUG is not set # # Generic devices # 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_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_AD1848_LIB=m CONFIG_SND_CS4231_LIB=m CONFIG_SND_SB_COMMON=m CONFIG_SND_SB8_DSP=m CONFIG_SND_SB16_DSP=m # # ISA devices # 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 is not set CONFIG_SND_GUS_SYNTH=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_SB16_CSP_FIRMWARE_IN_KERNEL=y CONFIG_SND_SGALAXY=m CONFIG_SND_SSCAPE=m CONFIG_SND_WAVEFRONT=m CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL=y # # PCI devices # 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_AZT3328=m CONFIG_SND_BT87X=m # CONFIG_SND_BT87X_OVERCLOCK is not set CONFIG_SND_CA0106=m CONFIG_SND_CMIPCI=m CONFIG_SND_CS4281=m CONFIG_SND_CS46XX=m CONFIG_SND_CS46XX_NEW_DSP=y # CONFIG_SND_CS5530 is not set 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_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_CONEXANT=y CONFIG_SND_HDA_CODEC_CMEDIA=y CONFIG_SND_HDA_CODEC_SI3054=y CONFIG_SND_HDA_GENERIC=y # CONFIG_SND_HDA_POWER_SAVE is not set CONFIG_SND_HDSP=m CONFIG_SND_HDSPM=m CONFIG_SND_ICE1712=m CONFIG_SND_ICE1724=m CONFIG_SND_INTEL8X0=m CONFIG_SND_INTEL8X0M=m CONFIG_SND_KORG1212=m CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL=y CONFIG_SND_MAESTRO3=m CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL=y 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_SONICVIBES=m CONFIG_SND_TRIDENT=m CONFIG_SND_VIA82XX=m CONFIG_SND_VIA82XX_MODEM=m CONFIG_SND_VX222=m CONFIG_SND_YMFPCI=m CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL=y CONFIG_SND_AC97_POWER_SAVE=y CONFIG_SND_AC97_POWER_SAVE_DEFAULT=0 # # SPI devices # # # USB devices # CONFIG_SND_USB_AUDIO=m CONFIG_SND_USB_USX2Y=m CONFIG_SND_USB_CAIAQ=m CONFIG_SND_USB_CAIAQ_INPUT=y # # PCMCIA devices # CONFIG_SND_VXPOCKET=m CONFIG_SND_PDAUDIOCF=m # # System on Chip audio support # # CONFIG_SND_SOC is not set # # SoC Audio support for SuperH # # # Open Sound System # CONFIG_SOUND_PRIME=m CONFIG_SOUND_TRIDENT=m # CONFIG_SOUND_MSNDCLAS is not set # CONFIG_SOUND_MSNDPIN is not set CONFIG_SOUND_OSS=m # CONFIG_SOUND_TRACEINIT is not set # CONFIG_SOUND_DMAP is not set 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_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=0x0 # CONFIG_AEDSP16_MSS is not set # CONFIG_AEDSP16_SBPRO is not set CONFIG_SOUND_KAHLUA=m CONFIG_AC97_BUS=m CONFIG_HID_SUPPORT=y CONFIG_HID=m # CONFIG_HID_DEBUG is not set # CONFIG_HIDRAW is not set # # USB Input Devices # CONFIG_USB_HID=m CONFIG_USB_HIDINPUT_POWERBOOK=y # CONFIG_HID_FF is not set CONFIG_USB_HIDDEV=y # # USB HID Boot Protocol drivers # CONFIG_USB_KBD=m CONFIG_USB_MOUSE=m CONFIG_USB_SUPPORT=y CONFIG_USB_ARCH_HAS_HCD=y CONFIG_USB_ARCH_HAS_OHCI=y CONFIG_USB_ARCH_HAS_EHCI=y CONFIG_USB=m # CONFIG_USB_DEBUG is not set # # Miscellaneous USB options # CONFIG_USB_DEVICEFS=y CONFIG_USB_DEVICE_CLASS=y # CONFIG_USB_DYNAMIC_MINORS is not set CONFIG_USB_SUSPEND=y # CONFIG_USB_PERSIST is not set # CONFIG_USB_OTG is not set # # USB Host Controller Drivers # CONFIG_USB_EHCI_HCD=m CONFIG_USB_EHCI_SPLIT_ISO=y CONFIG_USB_EHCI_ROOT_HUB_TT=y # CONFIG_USB_EHCI_TT_NEWSCHED is not set CONFIG_USB_ISP116X_HCD=m CONFIG_USB_OHCI_HCD=m # CONFIG_USB_OHCI_HCD_SSB is not set # CONFIG_USB_OHCI_BIG_ENDIAN_DESC is not set # CONFIG_USB_OHCI_BIG_ENDIAN_MMIO is not set CONFIG_USB_OHCI_LITTLE_ENDIAN=y CONFIG_USB_UHCI_HCD=m CONFIG_USB_U132_HCD=m CONFIG_USB_SL811_HCD=m CONFIG_USB_SL811_CS=m # CONFIG_USB_R8A66597_HCD is not set # # USB Device Class drivers # CONFIG_USB_ACM=m CONFIG_USB_PRINTER=m # # NOTE: USB_STORAGE enables SCSI, and 'SCSI disk support' # # # may also be needed; see USB_STORAGE Help for more information # CONFIG_USB_STORAGE=m # CONFIG_USB_STORAGE_DEBUG is not set CONFIG_USB_STORAGE_DATAFAB=y CONFIG_USB_STORAGE_FREECOM=y CONFIG_USB_STORAGE_ISD200=y CONFIG_USB_STORAGE_DPCM=y CONFIG_USB_STORAGE_USBAT=y CONFIG_USB_STORAGE_SDDR09=y CONFIG_USB_STORAGE_SDDR55=y CONFIG_USB_STORAGE_JUMPSHOT=y CONFIG_USB_STORAGE_ALAUDA=y CONFIG_USB_STORAGE_KARMA=y # CONFIG_USB_LIBUSUAL is not set # # USB Imaging devices # CONFIG_USB_MDC800=m CONFIG_USB_MICROTEK=m CONFIG_USB_MON=y # # USB port drivers # CONFIG_USB_USS720=m # # USB Serial Converter support # CONFIG_USB_SERIAL=m CONFIG_USB_SERIAL_GENERIC=y CONFIG_USB_SERIAL_AIRCABLE=m CONFIG_USB_SERIAL_AIRPRIME=m CONFIG_USB_SERIAL_ARK3116=m CONFIG_USB_SERIAL_BELKIN=m # CONFIG_USB_SERIAL_CH341 is not set # CONFIG_USB_SERIAL_WHITEHEAT is not set CONFIG_USB_SERIAL_DIGI_ACCELEPORT=m CONFIG_USB_SERIAL_CP2101=m CONFIG_USB_SERIAL_CYPRESS_M8=m CONFIG_USB_SERIAL_EMPEG=m CONFIG_USB_SERIAL_FTDI_SIO=m CONFIG_USB_SERIAL_FUNSOFT=m CONFIG_USB_SERIAL_VISOR=m CONFIG_USB_SERIAL_IPAQ=m CONFIG_USB_SERIAL_IR=m # CONFIG_USB_SERIAL_EDGEPORT is not set # CONFIG_USB_SERIAL_EDGEPORT_TI is not set CONFIG_USB_SERIAL_GARMIN=m CONFIG_USB_SERIAL_IPW=m CONFIG_USB_SERIAL_KEYSPAN_PDA=m # CONFIG_USB_SERIAL_KEYSPAN is not set CONFIG_USB_SERIAL_KLSI=m CONFIG_USB_SERIAL_KOBIL_SCT=m CONFIG_USB_SERIAL_MCT_U232=m CONFIG_USB_SERIAL_MOS7720=m CONFIG_USB_SERIAL_MOS7840=m CONFIG_USB_SERIAL_NAVMAN=m CONFIG_USB_SERIAL_PL2303=m # CONFIG_USB_SERIAL_OTI6858 is not set CONFIG_USB_SERIAL_HP4X=m CONFIG_USB_SERIAL_SAFE=m # CONFIG_USB_SERIAL_SAFE_PADDED is not set CONFIG_USB_SERIAL_SIERRAWIRELESS=m # CONFIG_USB_SERIAL_TI is not set CONFIG_USB_SERIAL_CYBERJACK=m CONFIG_USB_SERIAL_XIRCOM=m CONFIG_USB_SERIAL_OPTION=m CONFIG_USB_SERIAL_OMNINET=m CONFIG_USB_SERIAL_DEBUG=m CONFIG_USB_EZUSB=y # # USB Miscellaneous drivers # # CONFIG_USB_EMI62 is not set # CONFIG_USB_EMI26 is not set CONFIG_USB_ADUTUX=m CONFIG_USB_AUERSWALD=m CONFIG_USB_RIO500=m CONFIG_USB_LEGOTOWER=m CONFIG_USB_LCD=m CONFIG_USB_BERRY_CHARGE=m CONFIG_USB_LED=m CONFIG_USB_CYPRESS_CY7C63=m CONFIG_USB_CYTHERM=m CONFIG_USB_PHIDGET=m CONFIG_USB_PHIDGETKIT=m CONFIG_USB_PHIDGETMOTORCONTROL=m CONFIG_USB_PHIDGETSERVO=m CONFIG_USB_IDMOUSE=m CONFIG_USB_FTDI_ELAN=m CONFIG_USB_APPLEDISPLAY=m CONFIG_USB_SISUSBVGA=m CONFIG_USB_SISUSBVGA_CON=y CONFIG_USB_LD=m CONFIG_USB_TRANCEVIBRATOR=m CONFIG_USB_IOWARRIOR=m CONFIG_USB_TEST=m # # USB DSL modem support # CONFIG_USB_ATM=m CONFIG_USB_SPEEDTOUCH=m CONFIG_USB_CXACRU=m CONFIG_USB_UEAGLEATM=m CONFIG_USB_XUSBATM=m # # USB Gadget Support # CONFIG_USB_GADGET=m # CONFIG_USB_GADGET_DEBUG is not set # CONFIG_USB_GADGET_DEBUG_FILES is not set # CONFIG_USB_GADGET_DEBUG_FS is not set CONFIG_USB_GADGET_SELECTED=y # CONFIG_USB_GADGET_AMD5536UDC is not set # CONFIG_USB_GADGET_ATMEL_USBA is not set # CONFIG_USB_GADGET_FSL_USB2 is not set CONFIG_USB_GADGET_NET2280=y CONFIG_USB_NET2280=m # CONFIG_USB_GADGET_PXA2XX is not set # CONFIG_USB_GADGET_M66592 is not set # CONFIG_USB_GADGET_GOKU is not set # CONFIG_USB_GADGET_LH7A40X is not set # CONFIG_USB_GADGET_OMAP is not set # CONFIG_USB_GADGET_S3C2410 is not set # CONFIG_USB_GADGET_AT91 is not set # CONFIG_USB_GADGET_DUMMY_HCD is not set CONFIG_USB_GADGET_DUALSPEED=y CONFIG_USB_ZERO=m CONFIG_USB_ETH=m CONFIG_USB_ETH_RNDIS=y CONFIG_USB_GADGETFS=m CONFIG_USB_FILE_STORAGE=m # CONFIG_USB_FILE_STORAGE_TEST is not set CONFIG_USB_G_SERIAL=m CONFIG_USB_MIDI_GADGET=m CONFIG_MMC=m # CONFIG_MMC_DEBUG is not set # CONFIG_MMC_UNSAFE_RESUME is not set # # MMC/SD Card Drivers # CONFIG_MMC_BLOCK=m CONFIG_MMC_BLOCK_BOUNCE=y # CONFIG_SDIO_UART is not set # # MMC/SD Host Controller Drivers # CONFIG_MMC_SDHCI=m # CONFIG_MMC_RICOH_MMC is not set CONFIG_MMC_WBSD=m CONFIG_MMC_TIFM_SD=m CONFIG_NEW_LEDS=y CONFIG_LEDS_CLASS=m # # LED drivers # CONFIG_LEDS_NET48XX=m CONFIG_LEDS_WRAP=m # # LED Triggers # CONFIG_LEDS_TRIGGERS=y CONFIG_LEDS_TRIGGER_TIMER=m CONFIG_LEDS_TRIGGER_IDE_DISK=y CONFIG_LEDS_TRIGGER_HEARTBEAT=m CONFIG_INFINIBAND=m CONFIG_INFINIBAND_USER_MAD=m CONFIG_INFINIBAND_USER_ACCESS=m CONFIG_INFINIBAND_USER_MEM=y CONFIG_INFINIBAND_ADDR_TRANS=y CONFIG_INFINIBAND_MTHCA=m CONFIG_INFINIBAND_MTHCA_DEBUG=y CONFIG_INFINIBAND_AMSO1100=m # CONFIG_INFINIBAND_AMSO1100_DEBUG is not set CONFIG_INFINIBAND_CXGB3=m # CONFIG_INFINIBAND_CXGB3_DEBUG is not set CONFIG_MLX4_INFINIBAND=m CONFIG_INFINIBAND_IPOIB=m # CONFIG_INFINIBAND_IPOIB_CM is not set CONFIG_INFINIBAND_IPOIB_DEBUG=y # CONFIG_INFINIBAND_IPOIB_DEBUG_DATA is not set CONFIG_INFINIBAND_SRP=m CONFIG_INFINIBAND_ISER=m # CONFIG_EDAC is not set CONFIG_RTC_LIB=m CONFIG_RTC_CLASS=m # # RTC interfaces # CONFIG_RTC_INTF_SYSFS=y CONFIG_RTC_INTF_PROC=y CONFIG_RTC_INTF_DEV=y # CONFIG_RTC_INTF_DEV_UIE_EMUL is not set # CONFIG_RTC_DRV_TEST is not set # # I2C RTC drivers # CONFIG_RTC_DRV_DS1307=m # CONFIG_RTC_DRV_DS1374 is not set CONFIG_RTC_DRV_DS1672=m CONFIG_RTC_DRV_MAX6900=m CONFIG_RTC_DRV_RS5C372=m CONFIG_RTC_DRV_ISL1208=m CONFIG_RTC_DRV_X1205=m CONFIG_RTC_DRV_PCF8563=m CONFIG_RTC_DRV_PCF8583=m # CONFIG_RTC_DRV_M41T80 is not set # # SPI RTC drivers # CONFIG_RTC_DRV_RS5C348=m CONFIG_RTC_DRV_MAX6902=m # # Platform RTC drivers # CONFIG_RTC_DRV_CMOS=m CONFIG_RTC_DRV_DS1553=m # CONFIG_RTC_DRV_STK17TA8 is not set CONFIG_RTC_DRV_DS1742=m CONFIG_RTC_DRV_M48T86=m # CONFIG_RTC_DRV_M48T59 is not set CONFIG_RTC_DRV_V3020=m # # on-CPU RTC drivers # # CONFIG_DMADEVICES is not set # CONFIG_AUXDISPLAY is not set # CONFIG_VIRTUALIZATION is not set # # Userspace I/O # # CONFIG_UIO is not set # # Firmware Drivers # CONFIG_EDD=m CONFIG_EFI_VARS=m CONFIG_DELL_RBU=m CONFIG_DCDBAS=m CONFIG_DMIID=y # # File systems # CONFIG_EXT2_FS=m CONFIG_EXT2_FS_XATTR=y CONFIG_EXT2_FS_POSIX_ACL=y CONFIG_EXT2_FS_SECURITY=y # CONFIG_EXT2_FS_XIP is not set CONFIG_EXT3_FS=m CONFIG_EXT3_FS_XATTR=y CONFIG_EXT3_FS_POSIX_ACL=y CONFIG_EXT3_FS_SECURITY=y CONFIG_EXT4DEV_FS=m CONFIG_EXT4DEV_FS_XATTR=y CONFIG_EXT4DEV_FS_POSIX_ACL=y CONFIG_EXT4DEV_FS_SECURITY=y CONFIG_JBD=m # CONFIG_JBD_DEBUG is not set CONFIG_JBD2=m # CONFIG_JBD2_DEBUG is not set CONFIG_FS_MBCACHE=m CONFIG_REISERFS_FS=m # CONFIG_REISERFS_CHECK is not set # CONFIG_REISERFS_PROC_INFO is not set CONFIG_REISERFS_FS_XATTR=y CONFIG_REISERFS_FS_POSIX_ACL=y CONFIG_REISERFS_FS_SECURITY=y CONFIG_JFS_FS=m CONFIG_JFS_POSIX_ACL=y CONFIG_JFS_SECURITY=y # CONFIG_JFS_DEBUG is not set # CONFIG_JFS_STATISTICS is not set CONFIG_FS_POSIX_ACL=y CONFIG_XFS_FS=m CONFIG_XFS_QUOTA=y CONFIG_XFS_SECURITY=y CONFIG_XFS_POSIX_ACL=y CONFIG_XFS_RT=y CONFIG_GFS2_FS=m CONFIG_GFS2_FS_LOCKING_NOLOCK=m CONFIG_GFS2_FS_LOCKING_DLM=m CONFIG_OCFS2_FS=m CONFIG_OCFS2_DEBUG_MASKLOG=y # CONFIG_OCFS2_DEBUG_FS is not set CONFIG_MINIX_FS=m CONFIG_ROMFS_FS=m CONFIG_INOTIFY=y CONFIG_INOTIFY_USER=y CONFIG_QUOTA=y # CONFIG_QUOTA_NETLINK_INTERFACE is not set CONFIG_PRINT_QUOTA_WARNING=y CONFIG_QFMT_V1=m CONFIG_QFMT_V2=m CONFIG_QUOTACTL=y CONFIG_DNOTIFY=y CONFIG_AUTOFS_FS=m CONFIG_AUTOFS4_FS=m CONFIG_FUSE_FS=m CONFIG_GENERIC_ACL=y # # CD-ROM/DVD Filesystems # CONFIG_ISO9660_FS=m CONFIG_JOLIET=y CONFIG_ZISOFS=y CONFIG_UDF_FS=m CONFIG_UDF_NLS=y # # DOS/FAT/NT Filesystems # CONFIG_FAT_FS=m CONFIG_MSDOS_FS=m CONFIG_VFAT_FS=m CONFIG_FAT_DEFAULT_CODEPAGE=437 CONFIG_FAT_DEFAULT_IOCHARSET="iso8859-1" CONFIG_NTFS_FS=m # CONFIG_NTFS_DEBUG is not set CONFIG_NTFS_RW=y # # Pseudo filesystems # CONFIG_PROC_FS=y CONFIG_PROC_KCORE=y CONFIG_PROC_SYSCTL=y CONFIG_SYSFS=y CONFIG_TMPFS=y CONFIG_TMPFS_POSIX_ACL=y # CONFIG_HUGETLBFS is not set # CONFIG_HUGETLB_PAGE is not set CONFIG_CONFIGFS_FS=m # # Miscellaneous filesystems # CONFIG_ADFS_FS=m # CONFIG_ADFS_FS_RW is not set CONFIG_AFFS_FS=m CONFIG_ECRYPT_FS=m CONFIG_HFS_FS=m CONFIG_HFSPLUS_FS=m CONFIG_BEFS_FS=m # CONFIG_BEFS_DEBUG is not set CONFIG_BFS_FS=m CONFIG_EFS_FS=m CONFIG_JFFS2_FS=m CONFIG_JFFS2_FS_DEBUG=0 CONFIG_JFFS2_FS_WRITEBUFFER=y # CONFIG_JFFS2_FS_WBUF_VERIFY is not set # CONFIG_JFFS2_SUMMARY is not set CONFIG_JFFS2_FS_XATTR=y CONFIG_JFFS2_FS_POSIX_ACL=y CONFIG_JFFS2_FS_SECURITY=y # CONFIG_JFFS2_COMPRESSION_OPTIONS is not set CONFIG_JFFS2_ZLIB=y # CONFIG_JFFS2_LZO is not set CONFIG_JFFS2_RTIME=y # CONFIG_JFFS2_RUBIN is not set CONFIG_CRAMFS=y CONFIG_VXFS_FS=m CONFIG_HPFS_FS=m CONFIG_QNX4FS_FS=m CONFIG_SYSV_FS=m CONFIG_UFS_FS=m CONFIG_UFS_FS_WRITE=y # CONFIG_UFS_DEBUG is not set CONFIG_NETWORK_FILESYSTEMS=y CONFIG_NFS_FS=m CONFIG_NFS_V3=y CONFIG_NFS_V3_ACL=y CONFIG_NFS_V4=y CONFIG_NFS_DIRECTIO=y CONFIG_NFSD=m CONFIG_NFSD_V2_ACL=y CONFIG_NFSD_V3=y CONFIG_NFSD_V3_ACL=y CONFIG_NFSD_V4=y CONFIG_NFSD_TCP=y CONFIG_LOCKD=m CONFIG_LOCKD_V4=y CONFIG_EXPORTFS=m CONFIG_NFS_ACL_SUPPORT=m CONFIG_NFS_COMMON=y CONFIG_SUNRPC=m CONFIG_SUNRPC_GSS=m CONFIG_SUNRPC_XPRT_RDMA=m CONFIG_SUNRPC_BIND34=y CONFIG_RPCSEC_GSS_KRB5=m CONFIG_RPCSEC_GSS_SPKM3=m CONFIG_SMB_FS=m # CONFIG_SMB_NLS_DEFAULT is not set CONFIG_CIFS=m # CONFIG_CIFS_STATS is not set # CONFIG_CIFS_WEAK_PW_HASH is not set # CONFIG_CIFS_XATTR is not set # CONFIG_CIFS_DEBUG2 is not set # CONFIG_CIFS_EXPERIMENTAL is not set CONFIG_NCP_FS=m CONFIG_NCPFS_PACKET_SIGNING=y CONFIG_NCPFS_IOCTL_LOCKING=y CONFIG_NCPFS_STRONG=y CONFIG_NCPFS_NFS_NS=y CONFIG_NCPFS_OS2_NS=y # CONFIG_NCPFS_SMALLDOS is not set CONFIG_NCPFS_NLS=y CONFIG_NCPFS_EXTRAS=y CONFIG_CODA_FS=m # CONFIG_CODA_FS_OLD_API is not set CONFIG_AFS_FS=m # CONFIG_AFS_DEBUG is not set # # Partition Types # CONFIG_PARTITION_ADVANCED=y CONFIG_ACORN_PARTITION=y # CONFIG_ACORN_PARTITION_CUMANA is not set # CONFIG_ACORN_PARTITION_EESOX is not set CONFIG_ACORN_PARTITION_ICS=y # CONFIG_ACORN_PARTITION_ADFS is not set # CONFIG_ACORN_PARTITION_POWERTEC is not set CONFIG_ACORN_PARTITION_RISCIX=y CONFIG_OSF_PARTITION=y CONFIG_AMIGA_PARTITION=y CONFIG_ATARI_PARTITION=y CONFIG_MAC_PARTITION=y CONFIG_MSDOS_PARTITION=y CONFIG_BSD_DISKLABEL=y CONFIG_MINIX_SUBPARTITION=y CONFIG_SOLARIS_X86_PARTITION=y CONFIG_UNIXWARE_DISKLABEL=y CONFIG_LDM_PARTITION=y # CONFIG_LDM_DEBUG is not set CONFIG_SGI_PARTITION=y CONFIG_ULTRIX_PARTITION=y CONFIG_SUN_PARTITION=y CONFIG_KARMA_PARTITION=y CONFIG_EFI_PARTITION=y # CONFIG_SYSV68_PARTITION is not set CONFIG_NLS=y CONFIG_NLS_DEFAULT="iso8859-1" CONFIG_NLS_CODEPAGE_437=m CONFIG_NLS_CODEPAGE_737=m CONFIG_NLS_CODEPAGE_775=m CONFIG_NLS_CODEPAGE_850=m CONFIG_NLS_CODEPAGE_852=m CONFIG_NLS_CODEPAGE_855=m CONFIG_NLS_CODEPAGE_857=m CONFIG_NLS_CODEPAGE_860=m CONFIG_NLS_CODEPAGE_861=m CONFIG_NLS_CODEPAGE_862=m CONFIG_NLS_CODEPAGE_863=m CONFIG_NLS_CODEPAGE_864=m CONFIG_NLS_CODEPAGE_865=m CONFIG_NLS_CODEPAGE_866=m CONFIG_NLS_CODEPAGE_869=m CONFIG_NLS_CODEPAGE_936=m CONFIG_NLS_CODEPAGE_950=m CONFIG_NLS_CODEPAGE_932=m CONFIG_NLS_CODEPAGE_949=m CONFIG_NLS_CODEPAGE_874=m CONFIG_NLS_ISO8859_8=m CONFIG_NLS_CODEPAGE_1250=m CONFIG_NLS_CODEPAGE_1251=m CONFIG_NLS_ASCII=m CONFIG_NLS_ISO8859_1=m CONFIG_NLS_ISO8859_2=m CONFIG_NLS_ISO8859_3=m CONFIG_NLS_ISO8859_4=m CONFIG_NLS_ISO8859_5=m CONFIG_NLS_ISO8859_6=m CONFIG_NLS_ISO8859_7=m CONFIG_NLS_ISO8859_9=m CONFIG_NLS_ISO8859_13=m CONFIG_NLS_ISO8859_14=m CONFIG_NLS_ISO8859_15=m CONFIG_NLS_KOI8_R=m CONFIG_NLS_KOI8_U=m CONFIG_NLS_UTF8=m CONFIG_DLM=m CONFIG_DLM_DEBUG=y CONFIG_INSTRUMENTATION=y CONFIG_PROFILING=y CONFIG_OPROFILE=m # CONFIG_KPROBES is not set # CONFIG_MARKERS is not set # # Kernel hacking # CONFIG_TRACE_IRQFLAGS_SUPPORT=y # CONFIG_PRINTK_TIME is not set CONFIG_ENABLE_WARN_DEPRECATED=y CONFIG_ENABLE_MUST_CHECK=y CONFIG_MAGIC_SYSRQ=y CONFIG_UNUSED_SYMBOLS=y CONFIG_DEBUG_FS=y # CONFIG_HEADERS_CHECK is not set CONFIG_DEBUG_KERNEL=y # CONFIG_DEBUG_SHIRQ is not set CONFIG_DETECT_SOFTLOCKUP=y CONFIG_SCHED_DEBUG=y # CONFIG_SCHEDSTATS is not set CONFIG_TIMER_STATS=y # CONFIG_DEBUG_SLAB is not set # CONFIG_DEBUG_RT_MUTEXES is not set # CONFIG_RT_MUTEX_TESTER is not set # CONFIG_DEBUG_SPINLOCK is not set # CONFIG_DEBUG_MUTEXES is not set # CONFIG_DEBUG_LOCK_ALLOC is not set # CONFIG_PROVE_LOCKING is not set # CONFIG_LOCK_STAT is not set # CONFIG_DEBUG_SPINLOCK_SLEEP is not set # CONFIG_DEBUG_LOCKING_API_SELFTESTS is not set # CONFIG_DEBUG_KOBJECT is not set # CONFIG_DEBUG_HIGHMEM is not set CONFIG_DEBUG_BUGVERBOSE=y # CONFIG_DEBUG_INFO is not set # CONFIG_DEBUG_VM is not set # CONFIG_DEBUG_LIST is not set # CONFIG_DEBUG_SG is not set # CONFIG_FRAME_POINTER is not set # CONFIG_FORCED_INLINING is not set # CONFIG_BOOT_PRINTK_DELAY is not set # CONFIG_RCU_TORTURE_TEST is not set # CONFIG_FAULT_INJECTION is not set # CONFIG_SAMPLES is not set CONFIG_EARLY_PRINTK=y # CONFIG_DEBUG_STACKOVERFLOW is not set # CONFIG_DEBUG_STACK_USAGE is not set # CONFIG_DEBUG_PAGEALLOC is not set # CONFIG_DEBUG_RODATA is not set # CONFIG_4KSTACKS is not set CONFIG_X86_FIND_SMP_CONFIG=y CONFIG_X86_MPPARSE=y CONFIG_DOUBLEFAULT=y # # Security options # CONFIG_KEYS=y # CONFIG_KEYS_DEBUG_PROC_KEYS is not set CONFIG_SECURITY=y CONFIG_SECURITY_NETWORK=y CONFIG_SECURITY_NETWORK_XFRM=y CONFIG_SECURITY_CAPABILITIES=y # CONFIG_SECURITY_FILE_CAPABILITIES is not set CONFIG_SECURITY_SELINUX=y CONFIG_SECURITY_SELINUX_BOOTPARAM=y CONFIG_SECURITY_SELINUX_BOOTPARAM_VALUE=0 CONFIG_SECURITY_SELINUX_DISABLE=y CONFIG_SECURITY_SELINUX_DEVELOP=y CONFIG_SECURITY_SELINUX_AVC_STATS=y CONFIG_SECURITY_SELINUX_CHECKREQPROT_VALUE=1 # CONFIG_SECURITY_SELINUX_ENABLE_SECMARK_DEFAULT is not set # CONFIG_SECURITY_SELINUX_POLICYDB_VERSION_MAX is not set CONFIG_XOR_BLOCKS=m CONFIG_ASYNC_CORE=m CONFIG_ASYNC_MEMCPY=m CONFIG_ASYNC_XOR=m CONFIG_CRYPTO=y CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_BLKCIPHER=m CONFIG_CRYPTO_HASH=y CONFIG_CRYPTO_MANAGER=y CONFIG_CRYPTO_HMAC=y CONFIG_CRYPTO_XCBC=m CONFIG_CRYPTO_NULL=m CONFIG_CRYPTO_MD4=m CONFIG_CRYPTO_MD5=y CONFIG_CRYPTO_SHA1=m CONFIG_CRYPTO_SHA256=m CONFIG_CRYPTO_SHA512=m CONFIG_CRYPTO_WP512=m CONFIG_CRYPTO_TGR192=m CONFIG_CRYPTO_GF128MUL=m CONFIG_CRYPTO_ECB=m CONFIG_CRYPTO_CBC=m CONFIG_CRYPTO_PCBC=m CONFIG_CRYPTO_LRW=m # CONFIG_CRYPTO_XTS is not set # CONFIG_CRYPTO_CRYPTD is not set CONFIG_CRYPTO_DES=m CONFIG_CRYPTO_FCRYPT=m CONFIG_CRYPTO_BLOWFISH=m CONFIG_CRYPTO_TWOFISH=m CONFIG_CRYPTO_TWOFISH_COMMON=m CONFIG_CRYPTO_TWOFISH_586=m CONFIG_CRYPTO_SERPENT=m CONFIG_CRYPTO_AES=m CONFIG_CRYPTO_AES_586=m CONFIG_CRYPTO_CAST5=m CONFIG_CRYPTO_CAST6=m CONFIG_CRYPTO_TEA=m CONFIG_CRYPTO_ARC4=m CONFIG_CRYPTO_KHAZAD=m CONFIG_CRYPTO_ANUBIS=m # CONFIG_CRYPTO_SEED is not set CONFIG_CRYPTO_DEFLATE=m CONFIG_CRYPTO_MICHAEL_MIC=m CONFIG_CRYPTO_CRC32C=m CONFIG_CRYPTO_CAMELLIA=m CONFIG_CRYPTO_TEST=m # CONFIG_CRYPTO_AUTHENC is not set CONFIG_CRYPTO_HW=y CONFIG_CRYPTO_DEV_PADLOCK=m CONFIG_CRYPTO_DEV_PADLOCK_AES=m CONFIG_CRYPTO_DEV_PADLOCK_SHA=m CONFIG_CRYPTO_DEV_GEODE=m # # Library routines # CONFIG_BITREVERSE=y CONFIG_CRC_CCITT=m CONFIG_CRC16=m CONFIG_CRC_ITU_T=m CONFIG_CRC32=y # CONFIG_CRC7 is not set CONFIG_LIBCRC32C=m CONFIG_AUDIT_GENERIC=y CONFIG_ZLIB_INFLATE=y CONFIG_ZLIB_DEFLATE=m CONFIG_GENERIC_ALLOCATOR=y CONFIG_REED_SOLOMON=m CONFIG_REED_SOLOMON_DEC16=y CONFIG_TEXTSEARCH=y CONFIG_TEXTSEARCH_KMP=m CONFIG_TEXTSEARCH_BM=m CONFIG_TEXTSEARCH_FSM=m CONFIG_PLIST=y CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y CONFIG_CHECK_SIGNATURE=y ================================== unfortunately, lspci -v shows nothing. Thanks. rmgls(a)free.fr

1 0

Re: [alsa-devel] Trouble building alsa-lib
by Takashi Iwai 14 Feb '08

14 Feb '08

At Thu, 14 Feb 2008 10:18:33 -0600, Linux Mercedes wrote: > > Hi Takashi, > > Where should I pass -D_POSIX_C_SOURCE? $ CFLAGS="-O2 -Wall -D_POSIX_C_SOURCE" ./hgcompile ... In the case of csh, use env. Takashi

1 0

[alsa-devel] Trouble building alsa-lib
by Linux Mercedes 14 Feb '08

14 Feb '08

Hi all, I'm trying to build alsa from source. I'm guessing (I was not able to find any instructions on how to build alsa) that I build alsa-lib first. I'm compiling it with gcc 4.2.1. However, when I build it, I get the following errors: In file included from ../../include/local.h:105, from control_local.h:22, from cards.c:35: ../../include/sound/asound.h:148: warning: implicit declaration of function '_IO R' ../../include/sound/asound.h:148: error: expected expression before 'int' ../../include/sound/asound.h:148: error: enumerator value for 'SNDRV_HWDEP_IOCTL _PVERSION' is not an integer constant ../../include/sound/asound.h:149: error: expected expression before 'struct' ../../include/sound/asound.h:149: error: enumerator value for 'SNDRV_HWDEP_IOCTL _INFO' is not an integer constant ../../include/sound/asound.h:150: error: expected expression before 'struct' ../../include/sound/asound.h:150: error: enumerator value for 'SNDRV_HWDEP_IOCTL _DSP_STATUS' is not an integer constant ../../include/sound/asound.h:151: warning: implicit declaration of function '_IO W' ../../include/sound/asound.h:151: error: expected expression before 'struct' ../../include/sound/asound.h:152: error: enumerator value for 'SNDRV_HWDEP_IOCTL _DSP_LOAD' is not an integer constant ../../include/sound/asound.h:471: error: expected expression before 'int' ../../include/sound/asound.h:471: error: enumerator value for 'SNDRV_PCM_IOCTL_P VERSION' is not an integer constant ../../include/sound/asound.h:472: error: expected expression before 'struct' ../../include/sound/asound.h:472: error: enumerator value for 'SNDRV_PCM_IOCTL_I NFO' is not an integer constant ../../include/sound/asound.h:473: error: expected expression before 'int' ../../include/sound/asound.h:473: error: enumerator value for 'SNDRV_PCM_IOCTL_T STAMP' is not an integer constant ../../include/sound/asound.h:474: error: expected expression before 'int' ../../include/sound/asound.h:474: error: enumerator value for 'SNDRV_PCM_IOCTL_T TSTAMP' is not an integer constant ../../include/sound/asound.h:475: warning: implicit declaration of function '_IO WR' ../../include/sound/asound.h:475: error: expected expression before 'struct' ../../include/sound/asound.h:475: error: enumerator value for 'SNDRV_PCM_IOCTL_H W_REFINE' is not an integer constant ../../include/sound/asound.h:476: error: expected expression before 'struct' ../../include/sound/asound.h:476: error: enumerator value for 'SNDRV_PCM_IOCTL_H W_PARAMS' is not an integer constant ../../include/sound/asound.h:477: warning: implicit declaration of function '_IO ' ../../include/sound/asound.h:477: error: enumerator value for 'SNDRV_PCM_IOCTL_H W_FREE' is not an integer constant ../../include/sound/asound.h:478: error: expected expression before 'struct' ../../include/sound/asound.h:478: error: enumerator value for 'SNDRV_PCM_IOCTL_S W_PARAMS' is not an integer constant ../../include/sound/asound.h:479: error: expected expression before 'struct' ../../include/sound/asound.h:479: error: enumerator value for 'SNDRV_PCM_IOCTL_S TATUS' is not an integer constant ../../include/sound/asound.h:480: error: expected expression before 'sndrv_pcm_s frames_t' ../../include/sound/asound.h:480: error: enumerator value for 'SNDRV_PCM_IOCTL_D ELAY' is not an integer constant ../../include/sound/asound.h:481: error: enumerator value for 'SNDRV_PCM_IOCTL_H WSYNC' is not an integer constant ../../include/sound/asound.h:482: error: expected expression before 'struct' ../../include/sound/asound.h:482: error: enumerator value for 'SNDRV_PCM_IOCTL_S YNC_PTR' is not an integer constant ../../include/sound/asound.h:483: error: expected expression before 'struct' ../../include/sound/asound.h:483: error: enumerator value for 'SNDRV_PCM_IOCTL_C HANNEL_INFO' is not an integer constant ../../include/sound/asound.h:484: error: enumerator value for 'SNDRV_PCM_IOCTL_P REPARE' is not an integer constant ../../include/sound/asound.h:485: error: enumerator value for 'SNDRV_PCM_IOCTL_R ESET' is not an integer constant ../../include/sound/asound.h:486: error: enumerator value for 'SNDRV_PCM_IOCTL_S TART' is not an integer constant ../../include/sound/asound.h:487: error: enumerator value for 'SNDRV_PCM_IOCTL_D ROP' is not an integer constant ../../include/sound/asound.h:488: error: enumerator value for 'SNDRV_PCM_IOCTL_D RAIN' is not an integer constant ../../include/sound/asound.h:489: error: expected expression before 'int' ../../include/sound/asound.h:489: error: enumerator value for 'SNDRV_PCM_IOCTL_P AUSE' is not an integer constant ../../include/sound/asound.h:490: error: expected expression before 'sndrv_pcm_u frames_t' ../../include/sound/asound.h:490: error: enumerator value for 'SNDRV_PCM_IOCTL_R EWIND' is not an integer constant ../../include/sound/asound.h:491: error: enumerator value for 'SNDRV_PCM_IOCTL_R ESUME' is not an integer constant ../../include/sound/asound.h:492: error: enumerator value for 'SNDRV_PCM_IOCTL_X RUN' is not an integer constant ../../include/sound/asound.h:493: error: expected expression before 'sndrv_pcm_u frames_t' ../../include/sound/asound.h:493: error: enumerator value for 'SNDRV_PCM_IOCTL_F ORWARD' is not an integer constant ../../include/sound/asound.h:494: error: expected expression before 'struct' ../../include/sound/asound.h:494: error: enumerator value for 'SNDRV_PCM_IOCTL_W RITEI_FRAMES' is not an integer constant ../../include/sound/asound.h:495: error: expected expression before 'struct' ../../include/sound/asound.h:495: error: enumerator value for 'SNDRV_PCM_IOCTL_R EADI_FRAMES' is not an integer constant ../../include/sound/asound.h:496: error: expected expression before 'struct' ../../include/sound/asound.h:496: error: enumerator value for 'SNDRV_PCM_IOCTL_W RITEN_FRAMES' is not an integer constant ../../include/sound/asound.h:497: error: expected expression before 'struct' ../../include/sound/asound.h:497: error: enumerator value for 'SNDRV_PCM_IOCTL_R EADN_FRAMES' is not an integer constant ../../include/sound/asound.h:498: error: expected expression before 'int' ../../include/sound/asound.h:498: error: enumerator value for 'SNDRV_PCM_IOCTL_L INK' is not an integer constant ../../include/sound/asound.h:499: error: enumerator value for 'SNDRV_PCM_IOCTL_U NLINK' is not an integer constant ../../include/sound/asound.h:558: error: expected expression before 'int' ../../include/sound/asound.h:558: error: enumerator value for 'SNDRV_RAWMIDI_IOC TL_PVERSION' is not an integer constant ../../include/sound/asound.h:559: error: expected expression before 'struct' ../../include/sound/asound.h:559: error: enumerator value for 'SNDRV_RAWMIDI_IOC TL_INFO' is not an integer constant ../../include/sound/asound.h:560: error: expected expression before 'struct' ../../include/sound/asound.h:560: error: enumerator value for 'SNDRV_RAWMIDI_IOC TL_PARAMS' is not an integer constant ../../include/sound/asound.h:561: error: expected expression before 'struct' ../../include/sound/asound.h:561: error: enumerator value for 'SNDRV_RAWMIDI_IOC TL_STATUS' is not an integer constant ../../include/sound/asound.h:562: error: expected expression before 'int' ../../include/sound/asound.h:562: error: enumerator value for 'SNDRV_RAWMIDI_IOC TL_DROP' is not an integer constant ../../include/sound/asound.h:563: error: expected expression before 'int' ../../include/sound/asound.h:563: error: enumerator value for 'SNDRV_RAWMIDI_IOC TL_DRAIN' is not an integer constant ../../include/sound/asound.h:673: error: expected expression before 'int' ../../include/sound/asound.h:673: error: enumerator value for 'SNDRV_TIMER_IOCTL _PVERSION' is not an integer constant ../../include/sound/asound.h:674: error: expected expression before 'struct' ../../include/sound/asound.h:674: error: enumerator value for 'SNDRV_TIMER_IOCTL _NEXT_DEVICE' is not an integer constant ../../include/sound/asound.h:675: error: expected expression before 'int' ../../include/sound/asound.h:675: error: enumerator value for 'SNDRV_TIMER_IOCTL _TREAD' is not an integer constant ../../include/sound/asound.h:676: error: expected expression before 'struct' ../../include/sound/asound.h:676: error: enumerator value for 'SNDRV_TIMER_IOCTL _GINFO' is not an integer constant ../../include/sound/asound.h:677: error: expected expression before 'struct' ../../include/sound/asound.h:677: error: enumerator value for 'SNDRV_TIMER_IOCTL _GPARAMS' is not an integer constant ../../include/sound/asound.h:678: error: expected expression before 'struct' ../../include/sound/asound.h:678: error: enumerator value for 'SNDRV_TIMER_IOCTL _GSTATUS' is not an integer constant ../../include/sound/asound.h:679: error: expected expression before 'struct' ../../include/sound/asound.h:679: error: enumerator value for 'SNDRV_TIMER_IOCTL _SELECT' is not an integer constant ../../include/sound/asound.h:680: error: expected expression before 'struct' ../../include/sound/asound.h:680: error: enumerator value for 'SNDRV_TIMER_IOCTL _INFO' is not an integer constant ../../include/sound/asound.h:681: error: expected expression before 'struct' ../../include/sound/asound.h:681: error: enumerator value for 'SNDRV_TIMER_IOCTL _PARAMS' is not an integer constant ../../include/sound/asound.h:682: error: expected expression before 'struct' ../../include/sound/asound.h:682: error: enumerator value for 'SNDRV_TIMER_IOCTL _STATUS' is not an integer constant ../../include/sound/asound.h:684: error: enumerator value for 'SNDRV_TIMER_IOCTL _START' is not an integer constant ../../include/sound/asound.h:685: error: enumerator value for 'SNDRV_TIMER_IOCTL _STOP' is not an integer constant ../../include/sound/asound.h:686: error: enumerator value for 'SNDRV_TIMER_IOCTL _CONTINUE' is not an integer constant ../../include/sound/asound.h:687: error: enumerator value for 'SNDRV_TIMER_IOCTL _PAUSE' is not an integer constant ../../include/sound/asound.h:869: error: expected expression before 'int' ../../include/sound/asound.h:869: error: enumerator value for 'SNDRV_CTL_IOCTL_P VERSION' is not an integer constant ../../include/sound/asound.h:870: error: expected expression before 'struct' ../../include/sound/asound.h:870: error: enumerator value for 'SNDRV_CTL_IOCTL_C ARD_INFO' is not an integer constant ../../include/sound/asound.h:871: error: expected expression before 'struct' ../../include/sound/asound.h:871: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_LIST' is not an integer constant ../../include/sound/asound.h:872: error: expected expression before 'struct' ../../include/sound/asound.h:872: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_INFO' is not an integer constant ../../include/sound/asound.h:873: error: expected expression before 'struct' ../../include/sound/asound.h:873: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_READ' is not an integer constant ../../include/sound/asound.h:874: error: expected expression before 'struct' ../../include/sound/asound.h:874: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_WRITE' is not an integer constant ../../include/sound/asound.h:875: error: expected expression before 'struct' ../../include/sound/asound.h:875: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_LOCK' is not an integer constant ../../include/sound/asound.h:876: error: expected expression before 'struct' ../../include/sound/asound.h:876: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_UNLOCK' is not an integer constant ../../include/sound/asound.h:877: error: expected expression before 'int' ../../include/sound/asound.h:877: error: enumerator value for 'SNDRV_CTL_IOCTL_S UBSCRIBE_EVENTS' is not an integer constant ../../include/sound/asound.h:878: error: expected expression before 'struct' ../../include/sound/asound.h:878: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_ADD' is not an integer constant ../../include/sound/asound.h:879: error: expected expression before 'struct' ../../include/sound/asound.h:879: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_REPLACE' is not an integer constant ../../include/sound/asound.h:880: error: expected expression before 'struct' ../../include/sound/asound.h:880: error: enumerator value for 'SNDRV_CTL_IOCTL_E LEM_REMOVE' is not an integer constant ../../include/sound/asound.h:881: error: expected expression before 'struct' ../../include/sound/asound.h:881: error: enumerator value for 'SNDRV_CTL_IOCTL_T LV_READ' is not an integer constant ../../include/sound/asound.h:882: error: expected expression before 'struct' ../../include/sound/asound.h:882: error: enumerator value for 'SNDRV_CTL_IOCTL_T LV_WRITE' is not an integer constant ../../include/sound/asound.h:883: error: expected expression before 'struct' ../../include/sound/asound.h:883: error: enumerator value for 'SNDRV_CTL_IOCTL_T LV_COMMAND' is not an integer constant ../../include/sound/asound.h:884: error: expected expression before 'int' ../../include/sound/asound.h:884: error: enumerator value for 'SNDRV_CTL_IOCTL_H WDEP_NEXT_DEVICE' is not an integer constant ../../include/sound/asound.h:885: error: expected expression before 'struct' ../../include/sound/asound.h:885: error: enumerator value for 'SNDRV_CTL_IOCTL_H WDEP_INFO' is not an integer constant ../../include/sound/asound.h:886: error: expected expression before 'int' ../../include/sound/asound.h:886: error: enumerator value for 'SNDRV_CTL_IOCTL_P CM_NEXT_DEVICE' is not an integer constant ../../include/sound/asound.h:887: error: expected expression before 'struct' ../../include/sound/asound.h:887: error: enumerator value for 'SNDRV_CTL_IOCTL_P CM_INFO' is not an integer constant ../../include/sound/asound.h:888: error: expected expression before 'int' ../../include/sound/asound.h:888: error: enumerator value for 'SNDRV_CTL_IOCTL_P CM_PREFER_SUBDEVICE' is not an integer constant ../../include/sound/asound.h:889: error: expected expression before 'int' ../../include/sound/asound.h:889: error: enumerator value for 'SNDRV_CTL_IOCTL_R AWMIDI_NEXT_DEVICE' is not an integer constant ../../include/sound/asound.h:890: error: expected expression before 'struct' ../../include/sound/asound.h:890: error: enumerator value for 'SNDRV_CTL_IOCTL_R AWMIDI_INFO' is not an integer constant ../../include/sound/asound.h:891: error: expected expression before 'int' ../../include/sound/asound.h:891: error: enumerator value for 'SNDRV_CTL_IOCTL_R AWMIDI_PREFER_SUBDEVICE' is not an integer constant ../../include/sound/asound.h:892: error: expected expression before 'int' ../../include/sound/asound.h:892: error: enumerator value for 'SNDRV_CTL_IOCTL_P OWER' is not an integer constant ../../include/sound/asound.h:893: error: expected expression before 'int' ../../include/sound/asound.h:893: error: enumerator value for 'SNDRV_CTL_IOCTL_P OWER_STATE' is not an integer constant ../../include/sound/asound.h:949: error: expected expression before 'struct' ../../include/sound/asound.h:949: error: enumerator value for 'SNDRV_IOCTL_READV ' is not an integer constant ../../include/sound/asound.h:950: error: expected expression before 'struct' ../../include/sound/asound.h:950: error: enumerator value for 'SNDRV_IOCTL_WRITE V' is not an integer constant In file included from ../../include/local.h:109, from control_local.h:22, from cards.c:35: ../../include/global.h:137: error: redefinition of 'struct timeval' ../../include/global.h:142: error: redefinition of 'struct timespec' make[2]: *** [cards.lo] Error 1 make[2]: Leaving directory `/nj/alsa-source/alsa-lib/src/control' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/nj/alsa-source/alsa-lib/src' make: *** [all-recursive] Error 1 (I know it's a lot of text, but I'm not sure how much is needed to determine what's wrong, not being overly familiar with alsa myself.) Can someone help me out please? Remember that since I'm using cygwin, it could be as simple as not having all the packages installed. Thanks, Nathan

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[alsa-devel] midi problem
by rmgls＠free.fr 14 Feb '08

14 Feb '08

Hello, upgrading to kernel 2.6.24.1 and alsa-hg-20080213, (libs in sync), on debian: trying to play some midi files, with aplaymidi to: timidity/mpu/rme midi port, i receive the following message: rtc: lost some interrupt at 1024hz perhaps, it's not alsa related but! Do you think it is related to the config? the symptom appeared since 2.6.24 kernel. in case, can you tell me what config options can influence this behaviour, (i have read and applied the low_latency doc), and if this is missing in the doc, to avoid such questions, can you add some words somewhere please? Thanks in advance for reply. rmgls(a)free.fr rmgls(a)free.fr

2 1