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[alsa-devel] Digidesign Mbox 2 Mini
by Mark Hills 19 Oct '08

19 Oct '08

I have been trying to use a Digidesign Mbox 2 Mini (USB) without success, but with some positive signs. The device is detected. From the software side, playback appears to work -- both aplay and mplayer push audio to somewhere (no crashes, and at the correct rate). But there is no sound from any of the outputs. Recording is not so good. It does not error or crash, but arecord hangs and creates a 44 byte file containing the WAV header. In a test program, snd_pcm_readi() does not return and strace waits on poll(). I found two threads on alsa-devel referring to the Mbox interfaces (a different model to the one I am using though; different ID): http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/004394.h… http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg12273.html In the second one, which is from 2004, Clemens says: > Strange. But having two endpoints in one interface is not what the > driver was written for. I think part of the driver needs to be > rewritten before this device can work. Now this is a while ago, with a different interface. But the Mbox 2 mini also has two endpoints in one interface (see attached lsusb output), and I seem to be getting considerably more out of the interface than the original poster. So is it possible that the driver is now successfully getting audio to the playback endpoint only, and just needs some kind of mixer device control (or similar) for playback to work? Or is the 'success' with aplay/mplayer misleading -- how far off working is this? Thanks, -- Mark

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[alsa-devel] [PATCH] snd-sbxfi: rename erroneous Master mixer control to "PCM Playback Volume".
by William Pitcock 19 Oct '08

19 Oct '08

The OSS4-provided master mixer control is actually relative to EMU20K1 port volume settings. So, it should be named PCM. The Master mixer control will be shortly readded as a control which sets the master volume. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index c006278..3e99a97 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -155,7 +155,7 @@ struct sbxfi { int timer_reprogram; unsigned int base_rate; - int master_vol[2]; + int pcm_vol[2]; int capture_vol[2]; int capsrc; int micboost; @@ -1103,7 +1103,7 @@ static void sbxfi_setup_play_mixer(struct sbxfi *chip, int i, vol, audio_ch; for (i = 0; i < 2; i++) { - vol = port->vol[i] + chip->master_vol[i] - MAX_VOLUME; + vol = port->vol[i] + chip->pcm_vol[i] - MAX_VOLUME; if (vol < 0) vol = 0; vol = amop_vol_table[vol]; @@ -1124,9 +1124,9 @@ static int sbxfi_update_play_mixer(struct sbxfi *chip, int *vol) struct sbxfi_port *port; for (i = 0; i < 2; i++) { - if (vol[i] == chip->master_vol[i]) + if (vol[i] == chip->pcm_vol[i]) continue; - chip->master_vol[i] = vol[i]; + chip->pcm_vol[i] = vol[i]; spin_lock_irq(&chip->port_lock); list_for_each_entry(port, &chip->port_list, list) { if (port->substream->stream != @@ -1516,7 +1516,7 @@ static int __devinit sbxfi_create_pcm(struct sbxfi *chip) /* * Mixer interface */ -static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1526,16 +1526,16 @@ static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, return 0; } -static int sbxfi_master_vol_get(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->master_vol[0]; - ucontrol->value.integer.value[1] = chip->master_vol[1]; + ucontrol->value.integer.value[0] = chip->pcm_vol[0]; + ucontrol->value.integer.value[1] = chip->pcm_vol[1]; return 0; } -static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); @@ -1543,7 +1543,7 @@ static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, vol[0] = ucontrol->value.integer.value[0]; vol[1] = ucontrol->value.integer.value[1]; - if (!memcmp(vol, chip->master_vol, sizeof(vol))) + if (!memcmp(vol, chip->pcm_vol, sizeof(vol))) return 0; if (vol[0] < 0) @@ -1658,13 +1658,13 @@ static int sbxfi_micboost_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new mixers[] __devinitdata = { { - .name = "Master Playback Volume", + .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE /* | SNDRV_CTL_ELEM_ACCESS_TLV_READ*/), - .info = sbxfi_master_vol_info, - .get = sbxfi_master_vol_get, - .put = sbxfi_master_vol_put, + .info = sbxfi_pcm_vol_info, + .get = sbxfi_pcm_vol_get, + .put = sbxfi_pcm_vol_put, /*.tlv = { .p = db_master_vol }*/ }, { -- 1.5.5.4

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[alsa-devel] snd-xfi programming, some guidelines.
by James Courtier-Dutton 19 Oct '08

19 Oct '08

Hi, I have not had time to review the current work on the snd-xfi, but I have provided input if I think it would be useful. I have one more piece of information that might help. The xfi has two ways to talk to all the registers. 1) ioports using ioport pointer offset write method as familiar with older emu10k1 cards. 2) iomem using simple memory writes. The same registers on the xfi card can be accessed via both methods. There is some guidance as to when to use each. I.e. Do not use iomem for everything. There are some registers that take longer to write to than others. Unfortunately, the datasheet does not tell me which these slow are, so trial and error might be required. Now, if one uses the iomem to write to these registers that take a long time to update, it ties up the entire PCI bus for the entire duration, so this is not a good idea. So, the recommendation is that one should use ioports to access these slow registers as this will not block the PCI bus. I suggest putting a timer round each ioport and iomem write to find the slow ones. Kind Regards James

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[alsa-devel] [PATCH] snd-sbxfi: use SBXXXX model numbers instead of user-unfriendly names like UAA and ORIG.
by William Pitcock 19 Oct '08

19 Oct '08

This is just something that bothers me. There's no reason to accept or reject it, other than that it might be more informative about what hardware revision we are dealing with. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index 30ad25b..c006278 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -1702,10 +1702,10 @@ static int __devinit sbxfi_create_mixer(struct sbxfi *chip) */ static char *typenames[SBXFI_HW_TYPES] = { - [SBXFI_HW_ORIG] = "ORIG", - [SBXFI_HW_055x] = "055x", - [SBXFI_HW_073x] = "073x", - [SBXFI_HW_UAA] = "UAA" + [SBXFI_HW_ORIG] = "SB0760", + [SBXFI_HW_055x] = "SB0550", + [SBXFI_HW_073x] = "SB0730", + [SBXFI_HW_UAA] = "SB0886" }; #ifdef CONFIG_PROC_FS -- 1.5.5.4

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[alsa-devel] [PATCH] snd-sbxfi: set up the DAC at card initialization, regardless of duplex mode
by William Pitcock 19 Oct '08

19 Oct '08

We should bring up the DAC at card initialization even in half-duplex mode, otherwise we have to deal with unfortunate glitches in the hardware. Constantly resetting the DAC can cause serious state desyncs, resulting in the driver eventually not working until a reboot occurs. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index d64d462..30ad25b 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(debug, "Debug flag: 0 = disable"); /* continuous rate support */ #undef XXX_CONT_RATE -/* full-duplex support */ +/* full duplex support */ #undef XXX_FULL_DUPLEX /* multi playback - not implemented */ @@ -1364,10 +1364,6 @@ static int sbxfi_playback_prepare(struct snd_pcm_substream *substream) port->frag_count = runtime->period_size; port->need_update = 0; -#ifndef XXX_FULL_DUPLEX - sbxfi_init_dac(chip); - sbxfi_setup_i2s(chip); -#endif sbxfi_setup_play_pitch(chip, port); sbxfi_setup_play_mixer(chip, port); sbxfi_setup_play_mapper(chip, port); @@ -1924,10 +1920,8 @@ static int sbxfi_init_hw(struct sbxfi *chip) sbxfi_daoimap_clear(chip); -#ifdef XXX_FULL_DUPLEX sbxfi_init_dac(chip); sbxfi_setup_i2s(chip); -#endif return 0; } -- 1.5.5.4

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[alsa-devel] [PATCH] snd-sbxfi: implement support for 192khz playback/capture.
by William Pitcock 19 Oct '08

19 Oct '08

Implements support for 192khz initialization for playback and capture. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index 4c65125..d64d462 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -514,7 +514,7 @@ static void sbxfi_write_amop(struct sbxfi *chip, static void sbxfi_playback_start(struct sbxfi *chip, struct sbxfi_port *port) { unsigned int start, loop; - unsigned int ctrl, ratec; + unsigned int ctrl, ratec = 0; struct snd_pcm_runtime *runtime; if (port->tlb_index < 0) @@ -523,12 +523,19 @@ static void sbxfi_playback_start(struct sbxfi *chip, struct sbxfi_port *port) start = port->tlb_index * SBXFI_PAGE_SIZE; loop = start + frames_to_bytes(runtime, runtime->buffer_size); + /* + * ratec is the rate selection mode, 0x4c = 48000hz, 0x5c = 96000hz, + * 0x6c = 192000hz. + */ switch (runtime->rate) { case 48000: - ratec = 0x4c; /* XXX what meaning? */ + ratec = 0x4c; break; case 96000: - ratec = 0x5c; /* XXX what meaning? */ + ratec = 0x5c; + break; + case 192000: + ratec = 0x6c; break; } ctrl = ratec; @@ -559,7 +566,7 @@ static void sbxfi_playback_stop(struct sbxfi *chip, struct sbxfi_port *port) static void sbxfi_capture_start(struct sbxfi *chip, struct sbxfi_port *port) { unsigned int start, loop; - unsigned int ctrl, ratec; + unsigned int ctrl, ratec = 0; struct snd_pcm_runtime *runtime; if (port->tlb_index < 0) @@ -575,6 +582,9 @@ static void sbxfi_capture_start(struct sbxfi *chip, struct sbxfi_port *port) case 96000: ratec = 0x5d | SRCCTL_RUN_STATE; break; + case 192000: + ratec = 0x6d | SRCCTL_RUN_STATE; + break; } ctrl = ratec; -- 1.5.5.4

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[alsa-devel] [PATCH] snd-sbxfi: rename erroneous Master mixer control to "PCM Playback Volume".
by William Pitcock 19 Oct '08

19 Oct '08

The OSS4-provided master mixer control is actually relative to EMU20K1 port volume settings. So, it should be named PCM. The Master mixer control will be shortly readded as a control which sets the master volume. Signed-off-by: William Pitcock <nenolod(a)petrie.sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index c006278..3e99a97 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -155,7 +155,7 @@ struct sbxfi { int timer_reprogram; unsigned int base_rate; - int master_vol[2]; + int pcm_vol[2]; int capture_vol[2]; int capsrc; int micboost; @@ -1103,7 +1103,7 @@ static void sbxfi_setup_play_mixer(struct sbxfi *chip, int i, vol, audio_ch; for (i = 0; i < 2; i++) { - vol = port->vol[i] + chip->master_vol[i] - MAX_VOLUME; + vol = port->vol[i] + chip->pcm_vol[i] - MAX_VOLUME; if (vol < 0) vol = 0; vol = amop_vol_table[vol]; @@ -1124,9 +1124,9 @@ static int sbxfi_update_play_mixer(struct sbxfi *chip, int *vol) struct sbxfi_port *port; for (i = 0; i < 2; i++) { - if (vol[i] == chip->master_vol[i]) + if (vol[i] == chip->pcm_vol[i]) continue; - chip->master_vol[i] = vol[i]; + chip->pcm_vol[i] = vol[i]; spin_lock_irq(&chip->port_lock); list_for_each_entry(port, &chip->port_list, list) { if (port->substream->stream != @@ -1516,7 +1516,7 @@ static int __devinit sbxfi_create_pcm(struct sbxfi *chip) /* * Mixer interface */ -static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1526,16 +1526,16 @@ static int sbxfi_master_vol_info(struct snd_kcontrol *kcontrol, return 0; } -static int sbxfi_master_vol_get(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->master_vol[0]; - ucontrol->value.integer.value[1] = chip->master_vol[1]; + ucontrol->value.integer.value[0] = chip->pcm_vol[0]; + ucontrol->value.integer.value[1] = chip->pcm_vol[1]; return 0; } -static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, +static int sbxfi_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct sbxfi *chip = snd_kcontrol_chip(kcontrol); @@ -1543,7 +1543,7 @@ static int sbxfi_master_vol_put(struct snd_kcontrol *kcontrol, vol[0] = ucontrol->value.integer.value[0]; vol[1] = ucontrol->value.integer.value[1]; - if (!memcmp(vol, chip->master_vol, sizeof(vol))) + if (!memcmp(vol, chip->pcm_vol, sizeof(vol))) return 0; if (vol[0] < 0) @@ -1658,13 +1658,13 @@ static int sbxfi_micboost_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new mixers[] __devinitdata = { { - .name = "Master Playback Volume", + .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE /* | SNDRV_CTL_ELEM_ACCESS_TLV_READ*/), - .info = sbxfi_master_vol_info, - .get = sbxfi_master_vol_get, - .put = sbxfi_master_vol_put, + .info = sbxfi_pcm_vol_info, + .get = sbxfi_pcm_vol_get, + .put = sbxfi_pcm_vol_put, /*.tlv = { .p = db_master_vol }*/ }, { -- 1.5.5.4

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[alsa-devel] alsa + resampler + jack: possible?
by Markus Grabner 19 Oct '08

19 Oct '08

Hi all! A while ago I started development of an ALSA driver for a device which natively only supports a sampling rate of 39062.5 Hz and 24 bit samples (see http://www.tanzband-scream.at/line6). The ALSA rate plugin does a perfect job for using this device with standard applications (such as xmms) to play audio files at "conventional" sample rates (e.g., 44.1 kHz) with a configuration like this: pcm.resampler { type rate slave { pcm hw:1 format S24_3LE rate 39062 } converter "samplerate" } However, this doesn't work for jack since jack requires a power-of-two period size, and ALSA can't find an integer period size for the raw device such that the target rate is met with sufficient precision. An example: the jack daemon can't be started with period size 512 and sample rate 48000 Hz, but it works for 47961 Hz since 512 * 39062.5 / 48000 = 416.666... ~= 417 bytes 512 / 417 * 39062.5 = 47961.63 ~= 47961 Hz When using this approach together with other audio data from a 48 kHz source, they would drift apart over time, which basically renders the alsa+jack setup useless for this device. I understand that resampling is easier to implement if the sample rate ratio equals the ratio of the (integer) period sizes of client and slave, but at least the underlying libsamplerate has no such restriction. I tried to find the relevant pieces in the ALSA source code (pcm_rate.c), but I'm not sure where these calculations are done and how much effort it would be to support arbitrary sample rate ratios for the resampler. Can this be done in a plugin, or does it require modifications in the ALSA library code base? Could the developers please give a few hints on that? Thanks & kind regards, Markus -- Markus Grabner - Computer Graphics and Vision Graz University of Technology, Inffeldgasse 16a/II, 8010 Graz, Austria Phone: +43/316/873-5041, Fax: +43/316/873-5050 WWW: http://www.icg.tugraz.at/Members/grabner

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[alsa-devel] snd_pcm_sw_params_get_xfer_align deprecated?
by Erik de Castro Lopo 19 Oct '08

19 Oct '08

Hi all, I've got some code that gives me a waring that the above function is deprecated. What should I do: a) Call some other function? If so which one? b) Just drop the call to that function and everything will continue to work as it should? c) Something else? Cheers, Erik -- ----------------------------------------------------------------- Erik de Castro Lopo ----------------------------------------------------------------- "The very definition of hell is having to maintain someone else's Perl code." -- Anonymous

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[alsa-devel] New BIOS, new vendor and device IDs
by Chris Pemberton 17 Oct '08

17 Oct '08

My machine is a Lenovo 3000 N100 laptop (0768 model). I had to recently update the BIOS. Old BIOS: lspci -> device id: [17aa:2066] which is a Realtek ALC861VD (found in patch_realtek.c) New BIOS: lspci -> device id: [8086:27d8] (found in hda_intel.c) In both cases, I use the module option "model=3stack" to get sound working: Laptop speakers work. Built-in mic works. External mic works Headphones work; but automatic speaker muting does not. I tried to debug why I needed this module option at all, several months ago (old BIOS), but I didn't get far and just became complacent with using the module option. If anyone is interested, I'll do what I can to help debug. It would be nice to be able to use Skype with headphones... Output of the alsa info script is at: http://www.alsa-project.org/db/?f=701ca6efa0500080dc78a555813f66c1d42096e6 That is with driver 1.0.17. I'll try to update to the latest snapshot and run the script again. Thanks, Chris

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