March 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Same pci id for different ASUS machine
by Jiang zhe 19 Mar '08

19 Mar '08

Hi, According to the [0003839] https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3839 The PCI ID of ASUS F9S is 1043:1339 and "model=lenovo" works for it. But I found that ASUS G1 have the same id and model is 3stack in the alsa driver. Which one is true? --- alsa-driver-hg20080313/alsa-kernel/pci/hda/patch_realtek.c 2008-03-13 09:00:08.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-03-13 10:38:57.000000000 +0800 @@ -12591,7 +12591,7 @@ static struct snd_pci_quirk alc861vd_cfg SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x1339, "Asus F9S", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),

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[alsa-devel] HW decoding abstracted in ALSA
by Harsha priya gupta 19 Mar '08

19 Mar '08

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? Thanks in advance, Harsha

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[alsa-devel] About question of "ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default"
by wangxf 19 Mar '08

19 Mar '08

alsa-devel，你好： I am new to ALSA, now I try to add one sound soc chip in linux 2.6.21 kernel on samsung arm designed board. I have do some underside work: 1: I have porting linux 2.6.21 kernel on samsung arm designed board.(log.txt add accessories) 2. kernel have recognise sound chip form log information. 3. I have a test app working in linux 2.6.18.(this test app is tested ok!) 4. I have download "alsa-lib-1.0.15.tar.bz2" and buid in linux host(Linux 2.6.18-5-686 #1 SMP Fri Jun 1 00:47:00 UTC 2007),create libasound.so.2.0.0. build order: 1)./configure --host=arm-linux CC=arm-linux-gcc ( i do not affirm this config) 2) make 3) and then change to su to use to make install. 5. download libasound.so.2.0.0 lib to samsung designed board and download test app to samsung designed board. 6. but i run test ,system prompt err: ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: No such file or directory Segmentation fault now,I doubt libasound.so.2.0.0 lib is conflict with kernel 2.6.21. hope help . 夏新移动通讯有限公司软件设计工程师王小锋电话：86－592－6516777EXT6643 E-MAIL:wangxf@amoi.com.cn wangxf 2008-03-19

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[alsa-devel] Having a fight with snd_sctl_install:src/control/setup.c in alsa-lib-1.0.14a
by Bouterse, Bob 18 Mar '08

18 Mar '08

I am using alsa-lib-1.0.14a on a VIA EPIA 15000g board. I am hoping that someone can help me. I have been in an entrenched fight with the sctl functions setting my ctl setting back to the config file values. I am working on some extensions to the alsa audio output plugins in xine to provide simultaneous analog and iec958 on a Via EPIA 15000g, board. The issue is that I need to set the iec958 status bits at runtime, but everytime I call: /*---------------------------------------------------------------------- -----*/ ... snd_ctl_elem_value_get_iec958(cval, &iec958); llprintf(LOG_MIXER,"Read AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958.status[0], iec958.status[1], iec958.status[2], iec958.status[3]); /* AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2 */ iec958.status[0] = IEC958_AES0_NONAUDIO | IEC958_AES0_CON_NOT_COPYRIGHT; iec958.status[1] = IEC958_AES1_CON_PCM_CODER | IEC958_AES1_CON_ORIGINAL; iec958.status[2] = 0x00; iec958.status[3] = IEC958_AES3_CON_FS_48000; llprintf(LOG_MIXER,"Writing AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958.status[0], iec958.status[1], iec958.status[2], iec958.status[3]); /* store the values */ snd_ctl_elem_value_set_iec958(cval, &iec958); if ((err = snd_ctl_elem_write(ctl_handle, cval)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_write(): %s\n", snd_strerror(err)); } ... /*---------------------------------------------------------------------- -----*/ snd_sctl_install resets the IEC958 status bits back to the original values! I am pulling my hair out trying to understand how to stop this. Unfortunately I don't have much time to fix this, please help. ---------------------------------------------------- Bob Bouterse Sr. Software Engineer Office of the CTO D&M Holdings Inc. 1-317-616-6541 (fax) 1-317-616-6541

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[alsa-devel] [PATCH] Use base ALC883 mixer for 6stack-dell model
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the same as alc883_base_mixer. Avoid duplicated code and use alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer definition. 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 17:54:42.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 18:00:08.000000000 -0300 @@ -6992,46 +6992,6 @@ static struct snd_kcontrol_new alc888_3s { } /* end */ }; -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", 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), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, 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("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, 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_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7941,7 +7901,7 @@ static struct alc_config_preset alc883_p .input_mux = &alc883_capture_source, }, [ALC888_6ST_DELL] = { - .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer }, + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, -- []'s Herton

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[alsa-devel] [PATCH] Remove now uneeded 6stack-hp model from ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

After DAC assignment fix in ALC883, the 6stack-hp model is now the same as 6stack-dig. So just remove 6stack-hp model and replace its use with 6stack-dig. 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 2008-03-18 17:41:33.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-18 17:43:06.000000000 -0300 @@ -879,7 +879,6 @@ Prior to version 0.9.0rc4 options had a lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 - 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D 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 17:30:24.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 17:41:13.000000000 -0300 @@ -197,7 +197,6 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC883_HAIER_W66, - ALC888_6ST_HP, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, @@ -6955,46 +6954,6 @@ static struct snd_kcontrol_new alc883_me { } /* end */ }; -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", 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), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, 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("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, 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 alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7720,7 +7679,6 @@ static const char *alc883_models[ALC883_ [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC883_HAIER_W66] = "haier-w66", - [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", @@ -7739,7 +7697,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), @@ -7972,17 +7930,6 @@ static struct alc_config_preset alc883_p .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, }, - [ALC888_6ST_HP] = { - .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, [ALC888_3ST_HP] = { .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, -- []'s Herton

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[alsa-devel] Issues with alsa-lib-1.0.13 built using arm gcc 4.1.1
by sandeep lahane 18 Mar '08

18 Mar '08

Hello, I have written an external io type plugin for an arm based board. While debugging the plugin, I encountered unexpected errors which I doubt are because of gcc optimizations, and after looking up on web it seems that arm-gcc 4.1.1 which we are using has some known bugs due to which alsa-library functions behave unexpectedly. E.g. snd_mask_refine() reportedly behaves differently when it is compiled using different optimization levels, I am facing the same issue where snd_mask_refine() returns unexpected (-EINVAL) and after I modify the code as in src/pcm/mask_inline.h MASK_INLINE void snd_mask_copy(snd_mask_t *mask, const snd_mask_t *v) { - *mask = *v; + memcpy(mask->bits, v->bits, MASK_SIZE * 4); } snd_mask_refine() returns 0 which is valid but I get segfaults just after returning from snd_pcm_ioplug_hw_refine(). Is this a known bug and if yes/no then is there any list of known bugs/quirks when using alsa-lib with arm gcc 4.1.1? I am attaching a log below, pardon for non-alsa flavour of debug messages in the log. # aplay -D myplug etc/startup.wav AlsaVersion=1.0.13, Opened for Playback ----- INFO (ALSA) : Successfully Registered the IO plugin ALSA-OPENMAX PLUGIN Playing WAVE 'etc/startup.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Ste reo ####aplay.c set_params calling snd_pcm_hw_params_any ###### snd_pcm_hw_params_any 1 ##### _snd_pcm_hw_params_any 0 ##### _snd_pcm_hw_params_any 2 ##### _snd_pcm_hw_params_any 8 ##### _snd_pcm_hw_params_any 19 _snd_pcm_hw_param_any in hw_is_mask _snd_pcm_hw_param_any in hw_is_mask _snd_pcm_hw_param_any in hw_is_mask _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval _snd_pcm_hw_param_any in hw_is_interval ##### snd_pcm_hw_params_any 1 #####snd_pcm_hw_refine #####snd_pcm_hw_refine calling hw_refine #### snd_pcm_ioplug_hw_refine 1 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 1 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 1 snd_pcm_ioplug_hw_refine 2 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 3 snd_mask_refine 1 snd_mask_refine 1 snd_mask_refine 1 snd_pcm_ioplug_hw_refine 4 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 change is = 1 ##### return val of hw_refine = 1 #####snd_pcm_hw_refine #####snd_pcm_hw_refine calling hw_refine #### snd_pcm_ioplug_hw_refine 1 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 0 snd_ext_parm_mask_refine 1 snd_mask_refine 1 snd_ext_parm_mask_refine return val is = 0 snd_pcm_ioplug_hw_refine 2 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 3 snd_mask_refine 1 snd_pcm_ioplug_hw_refine 4 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 snd_ext_parm_interval_refine 1 snd_ext_parm_interval_refine 1 snd_pcm_ioplug_hw_refine 5 change is = 0 ##### return val of hw_refine = 0 Dec 31 16:27:59 kernel: aplay: unhandled page fault (11) at 0x00000022, code 0x0 17 Dec 31 16:27:59 kernel: pgd = c30d8000 Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Pid: 206, comm: aplay Dec 31 16:27:59 kernel: CPU: 0 Not tainted (2.6.23.14-BROADCOM #1) Dec 31 16:27:59 kernel: PC is at 0x40170770 Dec 31 16:27:59 kernel: LR is at 0x40168210 Dec 31 16:27:59 kernel: pc : [<40170770>] lr : [<40168210>] psr: a0000010 Dec 31 16:27:59 kernel: sp : bee6f698 ip : 401855f8 fp : bee6fb64 Dec 31 16:27:59 kernel: r10: 401852b0 r9 : 00000006 r8 : bee6f894 Dec 31 16:27:59 kernel: r7 : 0001cc64 r6 : 00000022 r5 : 000130fc r4 : bee6f7 04 Dec 31 16:27:59 kernel: r3 : ffffffff r2 : 00000022 r1 : fffffffe r0 : 000000 22 Dec 31 16:27:59 kernel: Flags: NzCv IRQs on FIQs on Mode USER_32 ISA ARM Se gment user Dec 31 16:27:59 kernel: Control: 00e5387f Table: c30d8008 DAC: 00000015 Dec 31 16:27:59 kernel: (__do_user_fault+0x5c/0xa4) Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: (do_page_fault+0x200/0x238) Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: (do_DataAbort+0x3c/0xa0) Dec 31 16:27:59 kernel: (ret_from_exception+0x0/0x10) Dec 31 16:27:59 kernel: Exception stack(0xc31a1fb0 to 0xc31a1ff8) Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Dec 31 16:27:59 kernel: Segmentation fault Regards, Sandeep.

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[alsa-devel] If you have a copy callback, do you need to have a silence?
by Barclay, Bruce 18 Mar '08

18 Mar '08

I was finally able to get my driver working. I had to use the copy callback since my memory is external to Alsa. I have noticed that I commented out the silence callback and the ack. All seems to work and just for grins, I put them in with some printk's and they were never invoked. I don't fully understand these callbacks. When are they called and am I just lucky? Bruce --------------------- Bruce Barclay Intel Corporation ( Phone : 480-554-0553 ( Cell : 480-274-7493 * Email:bruce.barclay@intel.com * MS : CH7-308 "Real Gold Fears No Fire" - Li Quan

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[alsa-devel] arecord | aplay expected delay
by van der Linden 18 Mar '08

18 Mar '08

Hi, I'm running arecord and aplay under uCLinux on a Blackfin DSP. In this way I loopback my audio. Does anyone know what delay I can expect between audio input and audio output? I measure at my test board 50ms. I want to have about 1ms! E-J

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[alsa-devel] [PATCH] 1/2 alsa-python: API coverage documentation tool
by Aldrin Martoq 18 Mar '08

18 Mar '08

HG Log: ------- Added python tool of alsa-lib C API coverage for alsa-python binding. Description: ------------ The following patch is a python tool that shows how much of the asoundlib API is being covered by the alsa-python binding. It works by parsing the C source code and comparing with the official Doxygen documentation from alsa website. It will help the following users: - alsa-python developers: know how is mapped the original C API, so which python function/variable may be used. - python binding developers: know how much of the C API is covered. It could help for mistakes (free() instead of using snd_ctl_card_info_free), know what is missing, and statistic interest ;) Sample output: -------------- Functions --------- 1 const char* snd_asoundlib_version ( void ) Returns the ALSA sound library version in ASCII format. => Used in : alsacard.c: asoundlib_version Defines ------- N/A #define SND_CONTROL_DLSYM_VERSION _dlsym_control_001 dlsym version for interface entry callback **** NOT AVAILABLE/USED SND_CONTROL_DLSYM_VERSION **** 2 #define SND_CTL_ASYNC Async notification (flag for open mode) => As constant: alsacontrol.c: open_mode["ASYNC"] => As constant: alsahcontrol.c: open_mode["ASYNC"] 1 #define snd_ctl_card_info_alloca ( ptr ) allocate an invalid snd_ctl_card_info_t using standard alloca => Used in : alsacontrol.c: pyalsacontrol_cardinfo Stats: ------ After every section, the program shows some stats about usage. This is the current usage of each doxygen module: Global defines and functions 16 missing ( 94%) 0 excluded ( 0%) of 17 total ( 6% covered). Error handling 9 missing ( 90%) 0 excluded ( 0%) of 10 total ( 10% covered). Control Interface 137 missing ( 53%) 0 excluded ( 0%) of 257 total ( 47% covered). High level Control Interface 21 missing ( 46%) 0 excluded ( 0%) of 46 total ( 54% covered). Setup Control Interface 4 missing (100%) 0 excluded ( 0%) of 4 total ( 0% covered). MIDI Sequencer 25 missing ( 61%) 7 excluded ( 17%) of 41 total ( 39% covered). Sequencer Client Interface 15 missing ( 35%) 14 excluded ( 33%) of 43 total ( 65% covered). Sequencer Port Interface 48 missing ( 74%) 6 excluded ( 9%) of 65 total ( 26% covered). Sequencer Port Subscription 3 missing ( 7%) 15 excluded ( 33%) of 46 total ( 93% covered). Sequencer Queue Interface 58 missing ( 84%) 0 excluded ( 0%) of 69 total ( 16% covered). Sequencer Event API 42 missing ( 93%) 0 excluded ( 0%) of 45 total ( 7% covered). Sequencer Miscellaneous 0 missing ( N/A) 0 excluded ( N/A) of 0 total ( N/A covered). Sequencer Event Type Checks 7 missing ( 23%) 0 excluded ( 0%) of 30 total ( 77% covered). Sequencer Event Definitions 35 missing ( 40%) 0 excluded ( 0%) of 87 total ( 60% covered). Sequencer Middle Level Interfa 38 missing ( 78%) 0 excluded ( 0%) of 49 total ( 22% covered). Sequencer event <-> MIDI byte 11 missing (100%) 0 excluded ( 0%) of 11 total ( 0% covered). Mixer Interface 44 missing ( 75%) 0 excluded ( 0%) of 59 total ( 25% covered). Simple Mixer Interface 14 missing ( 18%) 0 excluded ( 0%) of 78 total ( 82% covered). Cheers! -- Aldrin Martoq <amartoq(a)dcc.uchile.cl>

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