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[alsa-devel] Misusing snd_pcm_avail_update()
by Lennart Poettering 29 Jan '09

29 Jan '09

Heya! Currently in the 'glitch-free' logic of PulseAudio I use snd_pcm_avail_update() to estimate how I need to program my system timers for the next wake-up for the next buffer fill-up. For that I assume that the current fill level of the hardware buffer is the hardware buffer size minus what s_p_a_u() returns. I then convert that fill level from sample units to time units, and fix it up by the deviation of the sound card time from the system time. Finally I substract some extra margin just to make sure. This I assumed would tell me how much time will pass until an underrun happens if I don't write anything. Mostly this logic works fine. But on some setups and cases it doesn't. ALSA will signal an underrun much much earlier than what I estimated like this. I am now wondering why? One possibility of course is that s_p_a_u() is not reliable, due to driver issues (there were problems in the HDA driver about this, right?). Also, s_p_a_u() might simply lag behind quite a bit, or -- what I think is most likely -- because samples are popped in larger blocks form the hw playback buffer we reach the underrun much earlier than expected. I do acknowledge that the way i use s_p_a_u() is probably a misuse of the API. I make asssumptions I probably shouldn't make. Now, considering all this I'd like to ask for a new API function that tells me how much time I really have before the next underrun. It probably should just return a value in sample units, leaving it for the application to deal with system/sound card clock deviations. Any opinions on this? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Strange Config: Fedora 8 running on Amazon EC2
by Patrick Lightbody 29 Jan '09

29 Jan '09

I have a strange request (which is why I'm sending directly to the dev list). I'm trying to run Flash video (v9 or v10) on Firefox 3 on a Fedora 8 image running on Amazon EC2. I don't actually want/need sound (it's of course technically not possible, since EC2 doesn't provide a sound card). Video playback (when watched through VNC) goes fine for a couple minutes, but then the browser and flash lock up entirely. I've looked for ways to tell Firefox and/or Flash to not send audio out, but I've come up empty so far. So now I'm hoping that fixing this error and putting in perhaps some sort of "no op" device might help out. I'm not positive the error is related to ALSA, though the fact that there appear to be a ton of the following logs just before the lockup is suspicious: ALSA lib confmisc.c:768:(parse_card) cannot find card '0' ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default ALSA lib pcm_hw.c:1240:(_snd_pcm_hw_open) Invalid value for card (these lines repeat over and over again until the crash happens, then it stops promptly) As you can see by the alsa-info output at the following URL, I don't have any running snd modules: http://www.alsa-project.org/db/?f=6b2431d0564b3638cdb798a6d1a21eb73982588c Similarly, I don't have any sound-related devices installed (ie: /dev/audio, etc). My hope in emailing this list is that there may be a solution that causes ALSA to no-op the requests that Firefox/Flash send it's way, potentially solving the lockup I'm experiencing. I would really appreciate any ideas that could be shared. Thank you very much! Patrick

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[alsa-devel] ASoC : underrun messages
by Naveen Krishna Ch 28 Jan '09

28 Jan '09

Hello, I am experiencing some audio issue While playing using aplay some times i see the following messages. (Sorry if the messages are too long) [root@OMAP3 audio_samples]#aplay sample_96K.wav Playing WAVE 'sample_96K.wav' : Signed 16 bit Little Endian, Rate 96000 Hz, Stereo underrun!!! (at least 1307.251 ms long) underrun!!! (at least 554.656 ms long) underrun!!! (at least 988.861 ms long) underrun!!! (at least 554.443 ms long) underrun!!! (at least 987.670 ms long) underrun!!! (at least 995.484 ms long) ^CAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_48K.wav Playing WAVE 'sample_48K.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo underrun!!! (at least 69.306 ms long) underrun!!! (at least 130.981 ms long) underrun!!! (at least 205.750 ms long) underrun!!! (at least 178.100 ms long) underrun!!! (at least 146.057 ms long) ^CAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_8K.wav Playing WAVE 'sample_8K.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo ^C^[[AAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_22K.wav Playing WAVE 'sample_22K.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Stereo ^CAborted by signal Interrupt... [root@OMAP3 audio_samples]#aplay sample_44K.wav Playing WAVE 'sample_44K.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 1602.264 ms long) underrun!!! (at least 873.078 ms long) underrun!!! (at least 1693.268 ms long) underrun!!! (at least 870.483 ms long) underrun!!! (at least 540.466 ms long) underrun!!! (at least 1122.711 ms long) I am using OMAP3430 + twl4030 ASoC audio codec drivers. I understand that the prints are coming from alsa-lib aplay.c file As a result of the client not reading the buffers fast enough. These are not conditionally reproducible, i see them uncertainly. I saw the same isue with 2.6.26 some one suggested me to find if there is any bug in the asoc layer. Now i am using 2.6.28 kernel from kernel.org but still got the same Can any one suggest me a way to find a fix for this.. -- Cheers (: Naveen Krishna Ch :)

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[alsa-devel] alsa-lib failing at runtime
by Will Wagner 28 Jan '09

28 Jan '09

Hi All, I have found a problem with alsa-lib 1.0.19 running on an embedded arm board. It all compiles ok but aplay fails to play a .wav file and the kernel reports this error: ALSA sound/core/pcm_native.c:2587: unknown ioctl = 0x80184132 Decoding this ioctl it is SNDRV_PCM_IOCTL_CHANNEL_INFO defined in asound.h. The reason it is not working is that the kernel and alsa-lib think that the sndrv_pcm_channel_info struct is different sizes. The definition in asound.h is: struct sndrv_pcm_channel_info { unsigned int channel; off_t offset; /* mmap offset */ unsigned int first; /* offset to first sample in bits */ unsigned int step; /* samples distance in bits */ }; The problem is that when compiling linux it thinks off_t is 4 bytes but when compiling alsa-lib it thought it was 8 bytes. After discussion on the buildroot mailing list (see http://lists.busybox.net/pipermail/buildroot/2009-January/025458.html) it seems like for the kernel off_t is always 4 bytes but for user space it can be either 4 or 8 bytes depending on whether large file support is enabled. It appears that the fix is that the type of offset should be changed to long. Hope this all makes sense. Thanks, Will. -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Development Manager Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------

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[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(-)

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[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

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[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

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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

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[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(-)

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[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; }

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