June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 24 Jun '09

24 Jun '09

Shouldn't it be loading snd-hda-codec-ca0110 instead? I am using the 6/24 stable snapshot. lspci -vv gives me this. 05:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: Memory at fdffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

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[alsa-devel] RFC: PCM extra attributes
by Takashi Iwai 24 Jun '09

24 Jun '09

Hi, this is yet another topic I'm (currently) working on -- the addition of PCM ioctls to get/set some extra attributes. Basically, it adds two simple ioctls for getting/setting extra attributes to the PCM substream. The attribute has a sort of TLV form, /* PCM extra attributes */ struct snd_pcm_attr { unsigned int type; /* SNDRC_PCM_TYPE_ATTR_XXX */ unsigned int len; /* GET R: the max elements in value array * W: the actually written # elements * SET R/W: # elements to store */ unsigned int value[0]; /* value(s) to read / write */ }; And corresponding two ioctls #define SNDRV_PCM_IOCTL_GET_ATTR _IOWR('A', 0x14, struct snd_pcm_attr) #define SNDRV_PCM_IOCTL_SET_ATTR _IOWR('A', 0x15, struct snd_pcm_attr) The caller sets type and len fields. For getting attributes, the len points the max buffer length, and in return, the actual size is stored there. For setting, it's the size to set. Now, for getting the control ids associated to a PCM substream, it'll be like unsigned int buf[32]; buf[0] = SNDRV_PCM_ATTR_ASSOC_CTLS; buf[1] = ARRAY_SIZE(buf) - 2; /* minus header size */ ioctl(fd, SNDRV_PCM_IOCTL_GET_ATTR, buf); The kernel side implementation (a core part) is attached below. In this prototype, each driver is supposed to allocate and set substream->assoc_ctls and substream->num_assoc_ctls appropriately. I chose it since the control ids are statically associated to the PCM substreams at the driver loading time. It could be, of course, a liked list with kmalloc, but I thought it's overkill. Well, that's a pretty subtle thing and can be changed at any time. But, we need to define ABI properly from the beginning. So, if you have any comments on ABI or on the whole design, please speak up. An important note is that I'm planning to use this framework for getting/setting the PCM (surround) channel mapping. It's straightforward for this ABI. The internal implementation may vary, but the interface is the question now. For example, one could imagine that the channel mapping belongs to hw_params. But, it looks difficult because of the nature of channel mapping; it doesn't match with a simple integer parameter as hw_parmas has. The PCM channel mapping feature will be certainly useful for PA (and even for normal apps with a new alsa-lib plugin to convert the correct the channel mapping properly on demand). So, this is rather more interesting stuff to me, too... thanks, Takashi --- diff --git a/include/sound/asound.h b/include/sound/asound.h index 82aed3f..159826b 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -445,6 +445,21 @@ struct snd_xfern { snd_pcm_uframes_t frames; }; +/* PCM extra attributes */ +struct snd_pcm_attr { + unsigned int type; /* SNDRC_PCM_TYPE_ATTR_XXX */ + unsigned int len; /* GET R: the max elements in value array + * W: the actually written # elements + * SET R/W: # elements to store + */ + unsigned int value[0]; /* value(s) to read / write */ +}; + +/* PCM attribute types */ +enum { + SNDRV_PCM_ATTR_ASSOC_CTLS, /* array of associated ctl ids */ +}; + enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ @@ -459,6 +474,8 @@ enum { #define SNDRV_PCM_IOCTL_HW_PARAMS _IOWR('A', 0x11, struct snd_pcm_hw_params) #define SNDRV_PCM_IOCTL_HW_FREE _IO('A', 0x12) #define SNDRV_PCM_IOCTL_SW_PARAMS _IOWR('A', 0x13, struct snd_pcm_sw_params) +#define SNDRV_PCM_IOCTL_GET_ATTR _IOWR('A', 0x14, struct snd_pcm_attr) +#define SNDRV_PCM_IOCTL_SET_ATTR _IOWR('A', 0x15, struct snd_pcm_attr) #define SNDRV_PCM_IOCTL_STATUS _IOR('A', 0x20, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2389352..98416b0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -392,6 +392,10 @@ struct snd_pcm_substream { struct snd_info_entry *proc_prealloc_entry; struct snd_info_entry *proc_prealloc_max_entry; #endif + /* associated controls */ + unsigned int num_assoc_ctls; + unsigned int *assoc_ctls; + /* misc flags */ unsigned int hw_opened: 1; }; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 145931a..7a46674 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -748,6 +748,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->assoc_ctls); kfree(substream); substream = substream_next; } diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 08bfed5..07512b3 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -485,6 +485,8 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_TTSTAMP: case SNDRV_PCM_IOCTL_HWSYNC: case SNDRV_PCM_IOCTL_PREPARE: + case SNDRV_PCM_IOCTL_GET_ATTR: + case SNDRV_PCM_IOCTL_SET_ATTR: case SNDRV_PCM_IOCTL_RESET: case SNDRV_PCM_IOCTL_START: case SNDRV_PCM_IOCTL_DROP: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 84da3ba..46129c7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2492,6 +2492,43 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return 0; } +/* store back the associated ctl ids to the given PCM substream */ +static int snd_pcm_get_attr_assoc_ctls(struct snd_pcm_substream *substream, + unsigned int __user *arg) +{ + unsigned int len, i; + + arg++; + if (get_user(len, arg)) + return -EFAULT; + for (i = 0; i < len && i < substream->num_assoc_ctls; i++) + put_user(substream->assoc_ctls[i], arg + i + 1); + put_user(i, arg); + return 0; +} + +/* GET_ATTR ioctl */ +static int snd_pcm_get_attr(struct snd_pcm_substream *substream, + unsigned int __user *arg) +{ + unsigned int type; + + if (get_user(type, arg)) + return -EFAULT; + switch (type) { + case SNDRV_PCM_ATTR_ASSOC_CTLS: + return snd_pcm_get_attr_assoc_ctls(substream, arg); + } + return -ENOENT; +} + +/* SET_ATTR ioctl */ +static int snd_pcm_set_attr(struct snd_pcm_substream *substream, + struct snd_pcm_attr __user *arg) +{ + return -ENOENT; +} + static int snd_pcm_common_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) @@ -2519,6 +2556,10 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: return snd_pcm_prepare(substream, file); + case SNDRV_PCM_IOCTL_GET_ATTR: + return snd_pcm_get_attr(substream, arg); + case SNDRV_PCM_IOCTL_SET_ATTR: + return snd_pcm_set_attr(substream, arg); case SNDRV_PCM_IOCTL_RESET: return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START:

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Re: [alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 24 Jun '09

24 Jun '09

Bill, My creative x-fi card used to work great with the daily build, and the module I mentioned. I've been away for it for a couple of months because I need sound for wine programs and there were issues with wine sound, and I just got back. I was surprised to see such a big step backward. When I was last using a daily build, both unstable and stable worked for me. Susan Susan, I also had a CA110-IBG chip on my Creative X-Fi. Never could get it to work, not under Ubuntu or Fedora, not with the Creative-supplied driver and also not with ctxfi. So we dropped that card and went to using an Asus PCI-e sound card instead. As for what module should load, here is what Takashi said on June 11: Are you sure that your board has really emu20k1 or emu20k2 chip? X-Fi isn't always with these chips. Some are HD-audio only (thus snd-hda-intel is the right driver) and some are with CA0106 chip. -------- Original Message -------- Shouldn't it be loading snd-hda-codec-ca0110 instead? I am using the 6/24 stable snapshot. lspci -vv gives me this. 05:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: Memory at fdffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

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[alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Jun '09

24 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_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_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > 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 > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

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[alsa-devel] Need help on mmap
by Christopher Tsui 24 Jun '09

24 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

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[alsa-devel] Getting board to work on 2.6.30
by Tony Hoyle 24 Jun '09

24 Jun '09

Hi, I'm trying to get sound working on a board I'm working with. The sound works correctly on 2.6.24.7 but not 2.6.30. I has an intel HDA (Poulson) chipset and a sigmatel 9202 codec. Physically there's only one output - the speakers, and no input, so from a testing point of view it either works or it doesn't. Kernel output from both looks similar: 2.6.24.7: [ 19.942029] ALSA sound/pci/hda/hda_intel.c:916: HDA snoop disabled, try to enable ... OK [ 20.157458] ALSA sound/pci/hda/hda_codec.c:2760: autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) [ 20.157467] ALSA sound/pci/hda/hda_codec.c:2764: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 20.157474] ALSA sound/pci/hda/hda_codec.c:2768: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) [ 20.157481] ALSA sound/pci/hda/hda_codec.c:2776: inputs: mic=0x0, fmic=0xb, line=0xc, fline=0x0, cd=0x11, aux=0x0 2.6.30 (running alsa snapshot from last night): 19.379811] ALSA sound/pci/hda/hda_codec.c:3795: autoconfig: line_outs=1 (0x10/0x0/0x0/0x0/0x0) [ 19.379824] ALSA sound/pci/hda/hda_codec.c:3799: speaker_outs=2 (0xa/0x7/0x0/0x0/0x0) [ 19.379833] ALSA sound/pci/hda/hda_codec.c:3803: hp_outs=1 (0xd/0x0/0x0/0x0/0x0) [ 19.379842] ALSA sound/pci/hda/hda_codec.c:3804: mono: mono_out=0x10 [ 19.379849] ALSA sound/pci/hda/hda_codec.c:3807: dig-out=0x8/0x0 [ 19.379856] ALSA sound/pci/hda/hda_codec.c:3815: inputs: mic=0xb, fmic=0x0, line=0xc, fline=0x0, cd=0x11, aux=0x0 [ 19.390010] input: HDA Intel MID CD at Ext Rear Jack as /class/input/input3 [ 19.401109] input: HDA Intel MID Line In at Ext Rear Jack as /class/input/input4 [ 19.414726] input: HDA Intel MID Mic at Ext Rear Jack as /class/input/input5 [ 19.430427] input: HDA Intel MID Line Out at Ext Rear Jack as /class/input/input6 [ 19.447561] input: HDA Intel MID HP Out at Ext Front Jack as /class/input/input7 It seems to be creating the outputs as inputs? This seems wrong. I've tried every model setting I could find and reckon I need to create a new one.. but don't actually know what to create to get things working - there are lots of differences in the alsa between the two kernels and it isn't something that looks easy to pin down short of going nuts with git bisect. Can someone give me a pointer where to start looking at this? Tony Codec: SigmaTel STAC9202 Address: 0 Function Id: 0x1 Vendor Id: 0x83847632 Subsystem Id: 0x00000100 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=4, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x14 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x07 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x06 [Audio Selector] wcaps 0x300901: Stereo R/L Connection: 3 0x02* 0x07 0x14 Node 0x07 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x01110014: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x08 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x01410013: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x1, Sequence = 0x3 Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x14 Node 0x09 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=0 Amp-Out vals: [0x07 0x07] Connection: 1 0x0f Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02111011: [Jack] Speaker at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a10021: [Jack] Mic at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01810022: [Jack] Line In at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=03, enabled=1 Connection: 1 0x0e Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02214012: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Connection: 1 0x0e Node 0x0e [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x17 0x17] Connection: 1 0x06 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 5 0x0b* 0x0c 0x0d 0x0a 0x11 Node 0x10 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x01010031: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x40: OUT Connection: 1 0x12 Node 0x11 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x01310023: [Jack] CD at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Pin-ctls: 0x20: IN Node 0x12 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x06 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x14 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 2 0x09* 0x15 Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x01d10024: [Jack] Digitial In at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Pin-ctls: 0x20: IN

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[alsa-devel] [PATCH 1/3] ASoC: TWL4030: Enable voice filters for voice sidetone
by Lopez Cruz, Misael 24 Jun '09

24 Jun '09

Digital voice loopback (sidetone) requires voice filters to be enabled: VTXL, VTXR, VRX. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 67 +++++++++++++++++++++++++------------------- 1 files changed, 38 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4dbb853..780217a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -332,6 +332,28 @@ static void twl4030_power_down(struct snd_soc_codec *codec) twl4030_codec_enable(codec, 0); } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + /* Earpiece */ static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), @@ -712,7 +734,22 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) { + twl4030->bypass_state |= (1 << 5); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_PLAYBACK, 1); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_CAPTURE, 1); + } else { + twl4030->bypass_state &= ~(1 << 5); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_PLAYBACK, 0); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_CAPTURE, 0); + } + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +763,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) @@ -1806,28 +1837,6 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R - * (VTXL, VTXR) for uplink has to be enabled/disabled. */ -static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, - int enable) -{ - u8 reg, mask; - - reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); - - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - mask = TWL4030_ARXL1_VRX_EN; - else - mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; - - if (enable) - reg |= mask; - else - reg &= ~mask; - - twl4030_write(codec, TWL4030_REG_OPTION, reg); -} - static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { -- 1.5.4.3

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[alsa-devel] USB microphone not working. No sound, although device visible.
by Shapeshifter 23 Jun '09

23 Jun '09

Hi there. Following problem, details below: PROBLEM: I can't record anything using the microphone which is integrated into my USB webcam. The cam shows up as a device, and I can adjust the only available mixer which is for <Mic> in [Capture]. It's not muted and it's maxed out. DETAILS and PASTES: Alsa-info output: http://www.alsa-project.org/db/?f=f12fd357bc269d27ebdb4b5d562227bbe22a1086 You might notice the device comes up as [default]. If I run... $ arecord -Dplughw:1,0 -f cd -vv /dev/null or $ arecord -Ddefault -f cd -vv /dev/null then it out puts a lot of stuff (both the same) which I pasted in the below pastebin, but at the bottom it doesn't show any #'s and %'s to indicate the volume. The blue light on the webcam also turns on. If I save to a file instead of /dev/null and aplay it, it's just silence. Here's the output from the above command: http://pastie.org/521697 Although if I run $ arecord -Dhw:1,0 -f cd -vv /dev/null then it fails with: -> Recording WAVE '/dev/null' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo -> arecord: set_params:984: Channels count non available" I made a screenshot illustrating that everything is turned up and unmuted: http://omploader.org/vMXZldg ALREADY TRIED: - Switching the two devices so that 0 becomes the webcam and 1 the intel-hda through modprobe.conf but that had no other effect other than the switching. - Playing a lot with the device, muting, triplechecking volumes etc. etc. - Trying recording something in audacity. First time, the light goes on but it doesn't record, on subsequent tries, audacity spits an error stating "error opening sound device". - Trying it in linphone, light turns on, but other party can't hear anything. Any ideas how to fix hits? Btw, it used to work in ubuntu 9.04 for some reason. Tried it on two Archlinux boxes, it's exactly the same on both. But everything works really, everything is very up to date (I'm sure ubuntu has older versions of stuff). Well, all the info is in the alsa-info link above. Hope I can work this out! Thanks for any input! Regards.

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[alsa-devel] ASoC CODEC power management on reboot
by Ben Dooks 23 Jun '09

23 Jun '09

I've run into a problem where when the system reboots the audio starts playing random data. I suspect that this is due to the CODEC being left powered and when the during/after the reset process the GPIO lines which form the I2S bus are left floating. Should the ASoC (possibly DAPM) layer sort out powering down the CODEC before reboot, or should it be down to the codec or the mapping driver to do this? -- Ben Q: What's a light-year? A: One-third less calories than a regular year.

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[alsa-devel] Please help in adding ams-delta support to ASoC
by Janusz Krzysztofik 23 Jun '09

23 Jun '09

Hi, I am trying to add sound support for ams-delta omap machine, yet without much success. Three years ago, Mark Underwood created an omap-alsa compatible driver that basically worked[1]. It was derieved from similiar driver for aic23 codec found on omap osk machine. It looks like Mark has never managed to finish his work. It's not clear for me if he has found that getting a working fullduplex sound is technically impossible on a voice modem codec, shared among modem and cpu, controlable only from modem side afaik, but I have decided to give it a try. Since Mark's initial work, omap-alsa framework has been depreciated in favour of soc-omap. API changes are so significant that Mark's code is rather not useable directly any more. However, I am trying to use it as a starting point, by comparing it against it's prototype osk/aic23 code. Following Mark, I am trying to derieve the new ams-delta sound driver from current asoc driver for omap osk9512. For codec part, I decided to base my work on much more simple ad73311 rather that tlv320aic23. Comparing Mark's code agaist it's osk/aic23 prototype, I can see the folowing significant changes: 1. rate tables/bitmaps found in hw_constraint_rates, snd_omap_alsa_playback and snd_omap_alsa_capture structures limited to 8kHz, 2. hardware related code found in codec_configure_dev(), codec_clock_on() and codec_clock_off() callback functions replaced with ams-delta hardware specific code that switches the codec DAI pins from modem chip to mcbsp cpu interface and back, 3. codec_set_samplerate() and all mixer related functions replaced with stubs. 4. the following McBSP register settings changes: - .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) | - XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG, + .xcr2 = XPHASE | XWDLEN2(OMAP_MCBSP_WORD_16) | XFRLEN2(0), - .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1), + .srgr1 = CLKGDV(0), - .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1), + .srgr2 = GSYNC, - .pcr0 = CLKXP | CLKRP, /* mcbsp: slave */ I have found points 1. to 3. rather trivial to implement in new framework. Regarding point 4., it looks like detailed register settings are now done inside omap-mcbsp.c, based on machine specified format, so I have to find out what format should be specified for ams-delta. Please correct me if I am missing something. There was one more modification in Mark's code, addressing dma chaining problem on omap15xx hardware, but as far as I could see, the problem was already solved in the asoc omap framework. Initially, I based my work on ompenembedded provided linux-omap.git revision 90e758af52ba803cba233fabee81176d99589f09. The results were rather poor - total system hangup after first device access, with no single message. So I have switched to linux-2.6.30-rc5 and now I can safely access the device, however it does not work as expected. aplay and arecord wait forever, cat to/from /dev/dsp breaks with hardware error messgae. DMA interrput counters stay at 0. However, codec switching that I do from machine->ops->startup/shutdown seems working, as modem stops producing any sounds while the alsa device is in use and gets back thereafter. First of all, I'd like to make sure if my problem is related to my code only. As I am new in these areas, I would like to ask you if the omap asoc framework is stable enough to relay on. If yes, could you please look at my dirty code (attached) an give me some hints? I can provide you with more information if necessary. Regards, Janusz [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html

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