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[alsa-devel] [PATCH] hda: Add 5.1 support for second headphone jack
by Matthew Ranostay 25 Mar '08

25 Mar '08

Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks, the second headphone jack should be used for the 5.1 surround sound. Add support for 'Headphone as Line Out' switch, which allows it be used in 5.1 surround sound. 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 Tue Mar 18 10:45:08 2008 -0400 @@ -179,6 +179,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; unsigned int clfe_swap; + unsigned int hp_switch; unsigned int aloopback; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -1251,7 +1252,7 @@ static unsigned int ref92hd73xx_pin_conf static unsigned int dell_m6_pin_configs[13] = { 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, - 0x03a11020, 0x03011050, 0x4f0000f0, 0x4f0000f0, + 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, }; @@ -2029,6 +2030,33 @@ static void stac92xx_auto_set_pinctl(str AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info + +static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->hp_switch; + return 0; +} + +static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + spec->hp_switch = ucontrol->value.integer.value[0]; + + /* check to be sure that the ports are upto date with + * switch changes */ + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + + return 1; +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2099,6 +2127,15 @@ static int stac92xx_clfe_switch_put(stru return 1; } + +#define STAC_CODEC_HP_SWITCH(xname) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .info = stac92xx_hp_switch_info, \ + .get = stac92xx_hp_switch_get, \ + .put = stac92xx_hp_switch_put, \ + } #define STAC_CODEC_IO_SWITCH(xname, xpval) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -2124,6 +2161,7 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2132,6 +2170,7 @@ static struct snd_kcontrol_new stac92xx_ HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2396,6 +2435,14 @@ static int stac92xx_auto_create_multi_ou } } + if (cfg->hp_outs > 1) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_HP_SWITCH, + "Headphone as Line Out Switch", 0); + if (err < 0) + return err; + } + if (spec->line_switch) { nid = cfg->input_pins[AUTO_PIN_LINE]; pincap = snd_hda_param_read(codec, nid, @@ -3144,6 +3191,7 @@ static void stac92xx_hp_detect(struct hd { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; int i, presence; presence = 0; @@ -3154,11 +3202,15 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->hp_outs; i++) { if (presence) break; + if (spec->hp_switch && cfg->hp_pins[i] == nid) + break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } if (presence) { /* disable lineouts, enable hp */ + if (spec->hp_switch) + stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); @@ -3167,6 +3219,8 @@ static void stac92xx_hp_detect(struct hd AC_PINCTL_OUT_EN); } else { /* enable lineouts, disable hp */ + if (spec->hp_switch) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); @@ -3174,6 +3228,8 @@ static void stac92xx_hp_detect(struct hd stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); } + if (!spec->hp_switch && cfg->hp_outs > 1 && presence) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); } static void stac92xx_pin_sense(struct hda_codec *codec, int idx) @@ -3427,6 +3483,7 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ + spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break;

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Re: [alsa-devel] PATCH - ESI Juli driver
by Demian Martin 25 Mar '08

25 Mar '08

Pavel: I want to thank you for getting the Juli@ driver sorted out. Its all working right and I can play the 176.4 KHz 24 bit files now properly. Along with 96K, 88.2K 48K and 44.1K. I had to borrow an Apogee 192K mini-dac to test the system. The inexpensive DAC I got off eBay from China was never tested properly for high bit rate files and let me off track until I got a known DAC to test with. The unique clocking scheme of the Juli@ card should give lower jitter and better audio than the other solutions since the clock isn’t generated from a PLL. It’s a relative bargain compared to a Hammerfall or a Lynx card. And now with a good Linux driver very useable. -Demian Demian Martin Product Design Service 784 Cary Drive San Leandro, CA 94577 209 613 6990 No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.8/1340 - Release Date: 3/23/2008 6:50 PM

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[alsa-devel] Wrong offset, step and size passed from alsa-lib layers to external io type plugin
by Pharaoh . 25 Mar '08

25 Mar '08

Hi, In my external io type plugin, the alsa buffer is not accessible in the transfer function, symptom of this is I am unable to memcpy period size data from alsa buffer to my local buffers. after debugging a little I found out that values of offset, size and areas->step are wrong. #aplay --period-size =1024 --buffer-size=8192 test.wav static snd_pcm_sframes_t alsa_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) I get following, Buffer size = 32768 bytes Period size = 4096 bytes *ALSA buffer base addr = 0x2c6e0 //wrong Offset = 0xbeb4c7dc **//wrong* *size = 1063777280 **//wrong* step = 32 What could be the reason? -Pharaoh.

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[alsa-devel] ALSA Debugging
by Vinod Koul 24 Mar '08

24 Mar '08

Hello, I am new to ALSA and Linux. I was trying to understand the ALSA architecture. I have Linux Box with Intel sound card. with Suse 10.1 on it. I launched the audio player (Real player) and audio plays fine. Then, I built the kernel and installed it. I put some printk's in intel8x0.cand control.c. (could see the my code was complied) Even though I changed volume muted etc couldn't see any of my printk in dmesg. One default printk in intel8x0.c was there, so I have put my print before that line as well, but couldn't see anything. Apologies If my problem is very basic. I read on forums that best way to learn is to put printk's. Can anyone help me with where I am doing incorrectly, Am I supposed to do it this way or some other way? Thanks in advance -Vinod

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[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

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[alsa-devel] Buffer size/periods (Layla3G)
by Mark Hills 22 Mar '08

22 Mar '08

Hello, I am investigating problems I'm having with recording over long periods. I'm using a Layla3G and tested various ALSA versions including 1.0.14 and 1.0.16 on different machines. I've noticed that when I open the first pair of subdevices, my request for 1000ms buffer size is handled differently to other devices: $ ./test_buffers "hw:Layla3G,0,0" 1000 buffer_size=16384 period_size=128 $ ./test_buffers "hw:Layla3G,0,2" 1000 buffer_size=32768 period_size=256 $ ./test_buffers "hw:Layla3G,0,4" 1000 buffer_size=32768 period_size=256 $ ./test_buffers "hw:Layla3G,0,6" 1000 buffer_size=32768 period_size=256 If I try other buffer sizes, I can see is a limit to buffer size of 16384 samples on the ,0 subdevice, which is also affecting the chosen period size. But only on the first subdevice. Is there a reason, or is this a bug? Also, there seem to be failures at certain buffer sizes: $ ./test_buffers "hw:Layla3G,0,2" 429 buffer_size=18912 period_size=96 $ ./test_buffers "hw:Layla3G,0,2" 430 set_buffer_time_near: Invalid argument $ ./test_buffers "hw:Layla3G,0,2" 431 buffer_size=19008 period_size=96 Can anyone explain to me what's going on here, or is this also a bug? The overall problem I'm investigating is snd_pcm_readi() returning incomplete buffers after running for long periods until another snd_pcm_prepare(). It seems possible that this is related. Any help is greatly appreciated. Thanks, Mark

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[alsa-devel] [PATCH] Support mic automute for Clevo M720R/SR
by Herton Ronaldo Krzesinski 22 Mar '08

22 Mar '08

Add support for mic automute in clevo-m720r ALC883 model, and rename it to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt accordingly. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/Documentation/ALSA-Configuration.txt.orig alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa-kernel/Documentation/ALSA-Configuration.txt.orig 2008-03-20 16:51:03.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-20 16:52:00.000000000 -0300 @@ -886,7 +886,7 @@ Prior to version 0.9.0rc4 options had a 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D - clevo-m720r Clevo laptop M720R + clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) 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 2008-03-20 16:43:55.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-20 16:49:22.000000000 -0300 @@ -201,7 +201,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, - ALC883_CLEVO_M720R, + ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, ALC883_AUTO, ALC883_MODEL_LAST, @@ -6661,7 +6661,7 @@ static struct snd_kcontrol_new alc883_mi { } /* end */ }; -static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { +static struct snd_kcontrol_new alc883_clevo_m720_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), @@ -7130,7 +7130,7 @@ static struct hda_verb alc883_mitac_verb { } /* end */ }; -static struct hda_verb alc883_clevo_m720r_verbs[] = { +static struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -7140,6 +7140,7 @@ static struct hda_verb alc883_clevo_m720 /* enable unsolicited event */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, { } /* end */ }; @@ -7330,7 +7331,7 @@ static void alc883_tagra_unsol_event(str } /* toggle speaker-output according to the hp-jack state */ -static void alc883_clevo_m720r_automute(struct hda_codec *codec) +static void alc883_clevo_m720_hp_automute(struct hda_codec *codec) { unsigned int present; unsigned char bits; @@ -7342,11 +7343,33 @@ static void alc883_clevo_m720r_automute( HDA_AMP_MUTE, bits); } -static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, +static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +static void alc883_clevo_m720_automute(struct hda_codec *codec) +{ + alc883_clevo_m720_hp_automute(codec); + alc883_clevo_m720_mic_automute(codec); +} + +static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc883_clevo_m720r_automute(codec); + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_clevo_m720_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc883_clevo_m720_mic_automute(codec); + break; + } } /* toggle speaker-output according to the hp-jack state */ @@ -7605,7 +7628,7 @@ 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_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC883_AUTO] = "auto", }; @@ -7649,8 +7672,8 @@ 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, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720R), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), + SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), 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), @@ -7794,17 +7817,17 @@ 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 }, + [ALC883_CLEVO_M720] = { + .mixers = { alc883_clevo_m720_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720_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, + .unsol_event = alc883_clevo_m720_unsol_event, + .init_hook = alc883_clevo_m720_automute, }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, -- []'s Herton

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[alsa-devel] Kernel Oops unloading snd_ymfpci, 2.6.24
by Gerard Neil 22 Mar '08

22 Mar '08

Hi, I'm just upgraded to kernel 2.6.24 and I'm seeing a kernel OOPS every time I unload the snd_ymfpci module... Below is what I see in my log. My naive interpretation of this is that the system is barfing in the interrupt handler at the time it is trying to free the interrupt.... so maybe the interrupt hasn't been disabled properly before the free_irq() call? As I say, it's a naive interpretation. Anyway, I'd appreciate it if somebody could look this over and tell me what my next step should be. I'm available to debug if necessary. Thanks, Gerard Mar 18 00:16:58 {kern,info} ACPI: PCI Interrupt 0000:00:08.0[A] -> GSI 18 (level, low) -> IRQ 16 Mar 18 00:17:00 {kern,err} codec_ready: codec 0 is not ready [0x802a] Mar 18 00:17:00 {kern,alert} BUG: unable to handle kernel paging request at virtual address f8e10100 Mar 18 00:17:00 {kern,alert} printing eip: f8e2b430 *pde = 37770067 *pte = 00000000 Mar 18 00:17:00 {kern,emerg} Oops: 0000 [#1] SMP Mar 18 00:17:00 {kern,warning} Modules linked in: snd_ymfpci snd_seq snd_ac97_codec snd_pcm snd_opl3_lib snd_timer snd_hwdep snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device snd w83627hf hwmon_vid hwmon eeprom nfsd lockd nfs_acl sunrpc exportfs af_packet loop usbhid floppy ac97_bus soundcore ohci_hcd via_agp agpgart ehci_hcd skge uhci_hcd i2c_viapro i2c_core e100 Mar 18 00:17:00 {kern,warning} Mar 18 00:17:00 {kern,warning} Pid: 8543, comm: rmmod Not tainted (2.6.24-gentoo-r3 #1) Mar 18 00:17:00 {kern,warning} EIP: 0060:[<f8e2b430>] EFLAGS: 00210092 CPU: 1 Mar 18 00:17:00 {kern,warning} EIP is at snd_ymfpci_interrupt+0x10/0x1e0 [snd_ymfpci] Mar 18 00:17:00 {kern,warning} EAX: 00000010 EBX: f77f0200 ECX: e5a36980 EDX: f8e10000 Mar 18 00:17:00 {kern,warning} ESI: 00000010 EDI: dd74f000 EBP: c041e1d0 ESP: dd7a7e4c Mar 18 00:17:00 {kern,warning} DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Mar 18 00:17:00 {kern,emerg} Process rmmod (pid: 8543, ti=dd7a6000 task=e5a8aa60 task.ti=dd7a6000) Mar 18 00:17:00 {kern,emerg} Stack: f7c82960 00200286 000007b0 00000010 00000003 00200246 f77f0200 00000010 Mar 18 00:17:00 {kern,emerg} 00200282 c041e1d0 c01415b2 dd7b8000 1d7b8000 f7c57c4c e9e2a000 dd74f000 Mar 18 00:17:00 {kern,emerg} 00200216 dd74f000 e9e2a140 00002fff e9e2a000 f8e28d4f 00001fff f77f0d40 Mar 18 00:17:00 {kern,emerg} Call Trace: Mar 18 00:17:00 {kern,emerg} [<c01415b2>] free_irq+0x102/0x160 Mar 18 00:17:00 {kern,emerg} [<f8e28d4f>] snd_ymfpci_free+0x10f/0x150 [snd_ymfpci] Mar 18 00:17:00 {kern,emerg} [<f88bd6f3>] snd_device_free+0x63/0xb0 [snd] Mar 18 00:17:00 {kern,emerg} [<f88bd798>] snd_device_free_all+0x58/0x60 [snd] Mar 18 00:17:00 {kern,emerg} [<f88b96cd>] snd_card_do_free+0x2d/0xe0 [snd] Mar 18 00:17:00 {kern,emerg} [<f88ba361>] snd_card_free+0x91/0xa0 [snd] Mar 18 00:17:00 {kern,emerg} [<c022318b>] kref_put+0x2b/0x80 Mar 18 00:17:00 {kern,emerg} [<c03417cc>] wait_for_common+0x1c/0x140 Mar 18 00:17:00 {kern,emerg} [<f8e2801e>] snd_card_ymfpci_remove+0xe/0x20 [snd_ymfpci] Mar 18 00:17:00 {kern,emerg} [<c02309f6>] pci_device_remove+0x16/0x40 Mar 18 00:17:00 {kern,emerg} [<c027c6d4>] __device_release_driver+0x64/0xa0 Mar 18 00:17:00 {kern,emerg} [<c027cc21>] driver_detach+0xc1/0xd0 Mar 18 00:17:00 {kern,emerg} [<c027c235>] bus_remove_driver+0x75/0xa0 Mar 18 00:17:00 {kern,emerg} [<c0230ace>] pci_unregister_driver+0xe/0x70 Mar 18 00:17:00 {kern,emerg} [<c013f27a>] sys_delete_module+0x11a/0x1c0 Mar 18 00:17:00 {kern,emerg} [<c0114297>] do_page_fault+0xe7/0x620 Mar 18 00:17:00 {kern,emerg} [<c01553b7>] do_munmap+0x197/0x1f0 Mar 18 00:17:00 {kern,emerg} [<c0102b8e>] sysenter_past_esp+0x5f/0x85 Mar 18 00:17:00 {kern,emerg} [<c0340000>] unix_bind+0x170/0x2b0 Mar 18 00:17:00 {kern,emerg} ======================= Mar 18 00:17:00 {kern,emerg} Code: a8 c8 00 00 00 8b 47 10 8b 14 24 89 90 c0 00 00 00 8b 43 2c e9 40 ff ff ff 66 90 55 57 89 d7 56 53 83 ec 18 89 44 24 0c 8b 52 10 <8b> 82 00 01 00 00 85 c0 78 50 0f b7 42 04 0f b7 d8 f6 c3 01 74 Mar 18 00:17:00 {kern,emerg} EIP: [<f8e2b430>] snd_ymfpci_interrupt+0x10/0x1e0 [snd_ymfpci] SS:ESP 0068:dd7a7e4c Mar 18 00:17:00 {kern,warning} ---[ end trace 8697168af0baea4f ]---

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Re: [alsa-devel] PATCH - ESI Juli driver
by Demian Martin 21 Mar '08

21 Mar '08

Pavel; I have been following this closely since I have the juli@ card and have been having problems getting it to work at rates and bit depths other than 44.1/16 and 48/16 on the spdif output. I will be happy to provide a Juli@ card to support the work and keep this going. I’m working with a record company that is about to start releasing audio at 176.4/24 and the current playback options are very limited and expensive for that format. The Juli@ can open a lot of new options but it needs proper support. Let me know if you need it. Demian Martin Product Design Service No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1331 - Release Date: 3/16/2008 10:34 AM

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[alsa-devel] Accessing pcm samples from snd_pcm_channel_area_t in an io type plugin
by Pharaoh . 21 Mar '08

21 Mar '08

Hi, I am writing an external io type plugin for an ARM based board. In the transfer function of the plugin, I try to grab the audio samples in a temp buffer and send it to the DSP in chunks of period sizes. Somehow the the alsa buffer's audio samples are not copied to the temp buffer which I want to send further. What I do to copy the buffers, static snd_pcm_sframes_t my_transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { char *temp = NULL; temp = malloc(io->period_size * frame_size); //For malloc, //I tried all the three approaches here one by one, including the malloc/memcpy one 1. memcpy(temp,areas->addr+ offset*io->channels*2, io->period_size * frame_size); 2. memcpy(temp, (char *)areas->addr + (areas->first + areas->step * offset) /8, io->period_size * frame_size); //This should work straight off.... 3. temp = (char *)areas->addr + (areas->first + areas->step * offset) / 8; //write temp to DSP here ......... } Somehow the empty buffers are being sent to DSP, when I try to send some data from a static array then it goes to DSP and is played well. Any idea what I am missing? -phraoh.

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