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

[alsa-devel] ASoC updates
by Mark Brown 28 Jan '09

28 Jan '09

The following changes since commit 070504ade7a95a0f4395673717f3bb7d41793ca8: Mark Brown (1): ASoC: Fix L3 bus handling in Kconfig are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (1): ASoC: Push the codec runtime storage into the card structure Peter Ujfalusi (5): ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset cancelation ASoC: TWL4030: Code clean up for codec power up and down ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use ASoC: TWL4030: Physical ADC and amplifier power switch change ASoC: TWL4030: Move the twl4030_power_up and _power_down function Timur Tabi (1): ASoC: improve I2C initialization code in CS4270 driver include/sound/soc.h | 3 +- sound/soc/codecs/ac97.c | 20 ++-- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/ad73311.c | 8 +- sound/soc/codecs/ak4535.c | 14 +- sound/soc/codecs/cs4270.c | 254 ++++++++++++++++++---------------------- sound/soc/codecs/pcm3008.c | 12 +- sound/soc/codecs/ssm2602.c | 20 ++-- sound/soc/codecs/tlv320aic23.c | 18 ++-- sound/soc/codecs/tlv320aic26.c | 4 +- sound/soc/codecs/tlv320aic3x.c | 14 +- sound/soc/codecs/twl4030.c | 224 +++++++++++++++++++----------------- sound/soc/codecs/uda134x.c | 18 ++-- sound/soc/codecs/uda1380.c | 18 ++-- sound/soc/codecs/wm8350.c | 10 +- sound/soc/codecs/wm8510.c | 16 ++-- sound/soc/codecs/wm8580.c | 10 +- sound/soc/codecs/wm8728.c | 16 ++-- sound/soc/codecs/wm8731.c | 20 ++-- sound/soc/codecs/wm8750.c | 16 ++-- sound/soc/codecs/wm8753.c | 18 ++-- sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 18 ++-- sound/soc/codecs/wm8971.c | 14 +- sound/soc/codecs/wm8990.c | 14 +- sound/soc/codecs/wm9705.c | 15 ++- sound/soc/codecs/wm9712.c | 21 ++-- sound/soc/codecs/wm9713.c | 17 ++-- sound/soc/soc-core.c | 37 +++--- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-jack.c | 4 +- 31 files changed, 443 insertions(+), 456 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - add another MacBook Pro 4, 1 subsystem ID
by Luke Yelavich 28 Jan '09

28 Jan '09

Add another MacBook Pro 4,1 SSID (106b:3800). It seems that latter revisions, (at least mine), have different IDs to earlier revisions. Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 863ab95..aef543a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7036,6 +7036,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ + case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; default: -- 1.5.6.3

2 1

[alsa-devel] gusclassic alsa configuration
by Krzysztof Helt 28 Jan '09

28 Jan '09

Hi, I have given another go to fix Gravis Ultrasound configuration for ALSA and I have found a working configuration. The Gravis Classic card uses non-mmap and non-interleaved sound format so aplay fails with standard configuration. Even worse, no default pcm device is defined so someone who wants just install Linux on old computer won't get any sound. I have modified the GUS.conf from the /usr/share/alsa/cards and the GUS card now works with alsa and oss. My changes to the GUS.conf are below (as a diff). Does anybody know how to upload MIDI instruments into the GUS card under ALSA? (like with awesfx for SB AWE) Regards, Krzysztof --- /usr/src/alsa/alsa-lib-1.0.18/src/conf/cards/GUS.conf 2008-10-29 13:42:10.000000000 +0100 +++ GUS.conf 2009-01-21 20:42:42.675215220 +0100 @@ -19,10 +19,12 @@ } # -# It's a temporary solution. +# First route two channels to two other channels +# to drop mmap requirement. Then, use the mmap_emul +# plugin to allow using noninterleaved channels. # -!pcm.hw { +!pcm.default { @args [ CARD DEV SUBDEV ] @args.CARD { type string @@ -58,9 +60,16 @@ name defaults.pcm.subdevice } } - type hw - card $CARD - device $DEV - subdevice $SUBDEV - mmap_emulation on + type route + slave.pcm { + type mmap_emul + slave.pcm { + type hw + card $CARD + device $DEV + subdevice $SUBDEV + } + } + ttable.0.0 1 + ttable.1.1 1 } ---------------------------------------------------------------------- Speak Up. Angielski szybko i skutecznie. 3 miesiace nauki gratis. Sprawdz. >> http://link.interia.pl/f2019

3 14

Re: [alsa-devel] [PATCH] Turtle Beach Multisound Classic/Pinnacle driver (4th rev)
by Takashi Iwai 28 Jan '09

28 Jan '09

At Sat, 24 Jan 2009 13:35:28 +0100, Krzysztof Helt wrote: > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > This is driver for Turtle Beach Multisound cards: > Classic, Fiji and Pinnacle. > > Tested pcm playback and recording and MIDI playback > on Multisound Pinnacle. > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > --- > > All issues pointed by Takashi are resolved. I moved > more code to the common part and made a common > library. Thanks, this looks pretty good. But, one thing I forgot to ask... > +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) > +{ > + int err; > + struct snd_card *card; > + struct snd_msnd *chip; > + > + if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { > + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); > + return -ENODEV; > + } > + > + card = snd_card_new(index[idx], id[idx], THIS_MODULE, > + sizeof(struct snd_msnd)); > + if (!card) > + return -ENOMEM; Could you use the new API snd_card_create() introduced recently? > +#ifdef CONFIG_PNP > +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, > + const struct pnp_card_device_id *pid) > +{ ... > + /* > + * Create a new ALSA sound card entry, in anticipation > + * of detecting our hardware ... > + */ > + card = snd_card_new(index[idx], id[idx], THIS_MODULE, > + sizeof(struct snd_msnd)); > + if (!card) > + return -ENOMEM; Ditto. thanks, Takashi

2 2

[alsa-devel] [PATCH 0/6] ASoC: TWL4030: Cleanups and analog loopback support
by Peter Ujfalusi 27 Jan '09

27 Jan '09

Hello, The following series - at the end - will add analog loopback/bypass support for the twl4030 codec. The main changes: [1] ANAMICL register self cleaning bit handling [2] Simplified power up/down function [3] Move the Headset anti-pop/bias ramp enabling to be happend only if the headset is actually in use [4] Change the place for the DAC power switch [5] Preparation for the loopback implementation [6] Analog loopback/bypass implementation You can find a bit more detailed notes in the actuall patches. In short: if none of the loopbacks are enabled, the codec behaves in the same way as it used to. If the loopback is enabled the codec will be kept powered as long as the loopback is enabled. P�ter --- Peter Ujfalusi (6): ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset cancelation ASoC: TWL4030: Code clean up for codec power up and down ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use ASoC: TWL4030: Physical ADC and amplifier power switch change ASoC: TWL4030: Move the twl4030_power_up and _power_down function ASoC: TWL4030: Add analog loopback support sound/soc/codecs/twl4030.c | 340 ++++++++++++++++++++++++++++++-------------- 1 files changed, 232 insertions(+), 108 deletions(-)

2 12

[alsa-devel] [PATCH] Detailed messages in usbaudio.c
by LCID Fire 27 Jan '09

27 Jan '09

Attached are 3 patches which make the error messages in usbaudio.c more useful and also some code optimizations. commit 191cf816a5a618a84cc3627c5f9f96355f5d22c4 Author: Andreas Bergmeier <lcid-fire(a)gmx.net> Date: Sat Jan 24 12:17:01 2009 +0100 Print device number which did not support low speed audio streaming Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8b8eb9e..2cfb8a3 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2929,7 +2929,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) continue; } if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + snd_printk(KERN_ERR "%d: low speed audio streaming not supported\n", dev->devnum); continue; } if (! parse_audio_endpoints(chip, j)) { commit c6d62cbaf6d36b252768ef109b8d85a036e1d451 Author: Andreas Bergmeier <lcid-fire(a)gmx.net> Date: Sat Jan 24 12:16:16 2009 +0100 Replaced 'return NULL' lines by break in snd_usb_find_desc so there is only one default error. Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index d35b51c..8b8eb9e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2054,10 +2054,10 @@ void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype) end = p + desclen; for (; p < end;) { if (p[0] < 2) - return NULL; + break; next = p + p[0]; if (next > end) - return NULL; + break; if (p[1] == dtype && (!after || (void *)p > after)) { return p; } commit cd0180399253b17ece4fc1177457b21b8b70fe8f Author: Andreas Bergmeier <lcid-fire(a)gmx.net> Date: Sat Jan 24 12:22:13 2009 +0100 Add device and control number so it's apparent for which control this error is showing up. Signed-off-by: Andreas Bergmeier <lcid-fire(a)gmx.net> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 037435d..e578a5c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2867,12 +2867,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "%d:%u: cannot find HEADER\n", dev->devnum, ctrlif); return -EINVAL; } if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + snd_printk(KERN_ERR "%d:%u: invalid HEADER\n", dev->devnum, ctrlif); return -EINVAL; }

2 3

Re: [alsa-devel] handle an external PWM amplifier via alsa interface
by Mark Brown 27 Jan '09

27 Jan '09

On Tue, Jan 27, 2009 at 11:15:48AM +0100, Matteo Canella wrote: > I'm trying to implement an alsa interface for the audio section of my board. > I have an mpc5200 that control via i2s a pcm1774 stereo DAC and I also have > an amplifier between the DAC and the speakers. > At the moment I have the pcm1774 driver properly working, but I have a few > ideas on how to integrate the control of the external amplifier in the alsa > structure. I need to set the amplifier level with a PWM input to it. > Now I'm trying to use "non dapm" controls, adding the PWM settings into this > structure... > What do you guys think about this? > Am I completely wrong or could this be a good way? Using a non-DAPM control to set gain sounds sensible. Depending on how exactly this works you may want to also have a DAPM control so you can power the amplifier off when the path isn't in use (though your application may not be sufficiently sensitive to power consumption to make it worth bothering or the hardware may not let you do anything useful).

1 0

[alsa-devel] Help with Behringer UCG102 Quirks
by Keith A. Milner 27 Jan '09

27 Jan '09

Hi, I'm new to alsa development work so I would appreciate some advice. I'm trying to add specific support for the Behringer UCG102 Guitar Link USB interface. This is a simple device based on a TI codec. The aim is twofold: a) To provide a more descriptive name, as it currently describes itself as the generic TI device This was easy... I've included the following in usbquirks.h: /* Behringer UCG102 Guitar Link */ { USB_DEVICE(0x08bb, 0x2902), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Behringer", .product_name = "Guitar Link UCG102", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, } }, b) To include more specific support for the characteristics of the UCG102, such as sample rate and so on (if required). I suspect something is required in this respect because when you connect it, dmesg reports "cannot get freq at ep 0x84". I also get a lot of xruns with this device. I suspect it's not reporting itself properly to ALSA and isn't getting set up with the right settings. Of course, I may be completely wrong. I've attached the lsusb output for this. Can anyone help me decode this and help me understand what settings I need to configure? Cheers, -- Keith A. Milner

1 1

[alsa-devel] handle an external PWM amplifier via alsa interface
by Matteo Canella 27 Jan '09

27 Jan '09

Hi, I'm a newbie in ALSA development. I'm trying to implement an alsa interface for the audio section of my board. I have an mpc5200 that control via i2s a pcm1774 stereo DAC and I also have an amplifier between the DAC and the speakers. At the moment I have the pcm1774 driver properly working, but I have a few ideas on how to integrate the control of the external amplifier in the alsa structure. I need to set the amplifier level with a PWM input to it. Now I'm trying to use "non dapm" controls, adding the PWM settings into this structure... What do you guys think about this? Am I completely wrong or could this be a good way? Thanks a lot Ing. Matteo Canella Firmware Engineer Digitek - MTA SpA Via L. Romagnoli, 24 41033 Concordia s.S. (MO) - Italy T. +39 0535 57732 F. +39 0535 54550 email: matteo.canella(a)mta.it

1 0

[alsa-devel] CMI8788/AV200/Oxygen Volume Controls in alsamixer (oxygen_mixer.c:dac_volume_get())
by John L. Utz III 27 Jan '09

27 Jan '09

Hi Clemens and all; ( To those following along at home, kindly open alsa-driver-1.0.18a/alsa-kernel/pci/oxygen/oxygen_mixer.c and look at dac_volume_get() and dac_volume_put() ) It appears to me that these functions are attempting to work on all of the channels at once instead of just choosing Master, Front, Rear, Surround, or what have you: static int dac_volume_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; unsigned int i; mutex_lock(&chip->mutex); for (i = 0; i < chip->model.dac_channels; ++i) value->value.integer.value[i] = chip->dac_volume[i]; mutex_unlock(&chip->mutex); return 0; } This is then surfaced to userland thusly in the controls array: static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Stereo Upmixing", .info = upmix_info, .get = upmix_get, .put = upmix_put, }, { blah }, { blah }, { blah } }; this leads to a rather awkward result in alsamixer where there are 5 volume controls called "Master": < Master > Master Master Master Master IMHO, it would be more desirable for them to say: < Master > Front C/LFE Surround Rear Surr Our app would certainly be much happier :-) So, i *think* that i can fix this thusly: static const struct snd_kcontrol_new controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, .private_value = 0 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, .private_value = 0 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = dac_volume_info, .get = dac_volume_get, .put = dac_volume_put, .private_value = 1 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Front Playback Switch", .info = snd_ctl_boolean_mono_info, .get = dac_mute_get, .put = dac_mute_put, .private_value = 1 }, and alter dac_volume_*() to not use a for() loop and just work off the private_value. anybody have any opinions? i'll probably have tried this already before anybody responds but it seemed reasonable to surface the idea. tnx! johnu

4 12