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[alsa-devel] Strange code in
by Pete Zaitcev 23 Apr '07

23 Apr '07

Dear Takashi: Hans de Goede (on cc:) hits a problem when snd-usb-audio breaks with: "USB device is in the shutdown state, cannot create a card instance". This is how the place looks: chip = NULL; for (i = 0; i < SNDRV_CARDS; i++) { if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); goto __error; } chip = usb_chip[i]; break; } } The code looks broken to me, because dev at this time is a new device structure, freshly allocated. You cannot compare it with an old instance, and I don't know why you would want to. Unfortunately (for fortunately), I have zero clue about the way ALSA works, specifically rules for snd_device_ops.dev_free, so I am asking you to look at this problem. Yours, -- Pete

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[alsa-devel] [PATCH] sound: fix incorrect use of platform_device_register()
by Akinobu Mita 23 Apr '07

23 Apr '07

The platform_device allocated by platform_device_alloc() should be added to the device hierarchy by platform_device_add() instead of platform_device_register(). Otherwise it will hit WARN_ON() in platform_device_register(). by illegal refcount. This patch fixes such incorrect usages in portman2x4 and mts64 drivers. Also it removes unnecessary trailing whitespaces. Cc: Jaroslav Kysela <perex(a)suse.cz> Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Index: 2.6-mm/sound/drivers/portman2x4.c =================================================================== --- 2.6-mm.orig/sound/drivers/portman2x4.c +++ 2.6-mm/sound/drivers/portman2x4.c @@ -676,13 +676,13 @@ static void __devinit snd_portman_attach struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; } Index: 2.6-mm/sound/drivers/mts64.c =================================================================== --- 2.6-mm.orig/sound/drivers/mts64.c +++ 2.6-mm/sound/drivers/mts64.c @@ -892,13 +892,13 @@ static void __devinit snd_mts64_attach(s struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; }

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Re: [alsa-devel] Problem with Conexant
by Tobin Davis 23 Apr '07

23 Apr '07

Did you try adjusting the PCM volume levels? They may be set too low. That particular sound chip required a lot of manipulating to get functional, and is still very problematic. Unfortunately, I have no way of testing it, so users (like yourself) have to be my ears. The last time I worked with that particular configuration, the user that was testing said that it took a bit of tweaking with the PCM and other volume levels to get good sound. Tobin On Mon, 2007-04-23 at 08:11 +0307, Black X wrote: > after I install the last alsa-driver from the link you give me it > ( alsa-driver-hg20070421.tar.bz2 ) I still have the same problem. now > what I most to do to inform you about this issue? > > > Thank you. > > On Sun, 2007-04-22 at 19:27 -0700, Tobin Davis wrote: > > Could you try the latest alsa-driver snapshot from > > ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver? > > > > I can't tell what revision of Conexant code you are using based on > > your email. Anything after 20070413 should work better, and it will > > also be the most current. I haven't made any changes to the code > > since ~April 6. If you are still having audio issues, let me know. > > > > Tobin > > > > > > On Sun, 2007-04-22 at 23:31 +0307, Black X wrote: > > > Hello > > > > > > First at all I wanna thanks Tobin Davis for his support to get Conexant > > > audio devices working under Linux, and I was testing his last patchs, > > > but I had a little problems with my sound card, the first one is about > > > headphone, it is works but with noise, and when I'm maximize the volume > > > of headphone the sound quality starting so weak. > > > > > > The Second problem I had it is when I run the 3D Desktop Effects (XGL > > > with Compiz), that when I play any sound clip and moving any window or > > > rotate desktop cube that's making a noise with the sound, and the > > > quality of the sound clip it is so weak with slow playing in the sound. > > > > > > I make a little screen cast about this problem and you can see it at > > > this link: > > > http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html > > > > > > any way this is my alsa information: http://pastebin.ca/452453 > > > finaly my device information : > > > http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… > > > > > > I 'm sorry, but the English it's not my mother language. > > > > > > Thank you. > > > > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > > Tobin Davis <tdavis(a)dsl-only.net> > -- Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Ok, I try that before and try it now, still no change, and still have the same problem. When I highest the sound volume of headphone this makes sound starting noisy (and that's the same with PCM). About speaker it's work fine, but IF I moving any window or rotate the desktop cube this makes the sound very slow with noise.

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[alsa-devel] partial fix for bug #2708: [ALSA - driver] PCI - hda-intel
by Jeffrey Arbuckle 23 Apr '07

23 Apr '07

Using alsa-driver-1.0.14rc3 with an iMac Core2 Duo 24" results in audio only available through the headphones. Additionally the sound is very low, even at the highest mixer setting. This problem is very similar to the bug recently fixed in 1.0.14rc2 with the audio on the MacPro. I was able to make a patch that applies the same fix for the MacPro to my iMac. The patch fixes the audio so that the built in speakers work normally, but now the headphones do not work at all. Obviously some more work is needed on this. The patch I've attached just adds my device's' PCI SSID to an existing switch statement. I am willing to provide some more testing if necessary. Here is my systems info: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Realtek Semiconductor Co., Ltd. Unknown device 0885 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 92400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 # cat /proc/asound/card0/codec#0 Codec: Realtek ALC885 Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x106b1000 Revision Id: 0x100103 # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3. Compiled on Apr 22 2007 for kernel 2.6.20-15-generic (SMP).

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[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Hello First at all I wanna thanks Tobin Davis for his support to get Conexant audio devices working under Linux, and I was testing his last patchs, but I had a little problems with my sound card, the first one is about headphone, it is works but with noise, and when I'm maximize the volume of headphone the sound quality starting so weak. The Second problem I had it is when I run the 3D Desktop Effects (XGL with Compiz), that when I play any sound clip and moving any window or rotate desktop cube that's making a noise with the sound, and the quality of the sound clip it is so weak with slow playing in the sound. I make a little screen cast about this problem and you can see it at this link: http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html any way this is my alsa information: http://pastebin.ca/452453 finaly my device information : http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… I 'm sorry, but the English it's not my mother language. Thank you.

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[alsa-devel] Qustions on dmix - aplay - hw_ptr - appl_ptr
by Matthew Foster 23 Apr '07

23 Apr '07

I am a driver developer that is new to ALSA. I have read the tutorial as an example on how to write an ALSA driver. I appreciate the documentation you have provided online. I was wondering if you might have time to possibly answer a question. I am having trouble understanding things in the memory/buffer management area. I am not writing a driver for a PCI device, I am writing a driver that acts as an interface from the ALSA to a render device. So I essentially have a circular buffer I am transferring the ALSA data to. I set up the ALSA buffer like this: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), ALSA_BUF_SIZE, ALSA_BUF_SIZE)) < 0); And later on in my driver after open is called I have a buffer polling loop that does the period notification and constantly monitors my render (ouput) buffer. I have my asound.conf setup like this: pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } This is to enable the dmix plugin so I can run aplay more than once and have the two streams mixed in. The problem I am having is that it seems that when I run aplay like this: "aplay close.wav" I only see this pointer moving: substream->runtime->status->hw_ptrbir And if I run aplay like this: "aplay -D plughw:0,0 close.wav" I only see this pointer moving: substream->runtime->control->appl_ptr This seems odd to me and currently in my code I just check which pointer moves and use that to calculate how much data I need to write to my render device. This is so I can support both use cases. Another major problem is that when I play a short wav file 25KB or so and use the dmix "aplay close.wav" it randomly doesn't play at sometimes. Also if I run "aplay close.wav" and then very quickly run "aplay close.wav" again at the same prompt I get corruption of the data which produces screeching. In the case where I run aplay one right after the other I notice that the hw_ptr moves differently (less) than if I run it, wait, run again. I am also noticing also that when I do the aplay over and over quickly that these values change (get lower) in my snd_pcm_prepare callback: chip_data->buf_size = snd_pcm_lib_buffer_bytes(substream); chip_data->period_buf_size = snd_pcm_lib_period_bytes(substream); It seems like those value should remain constant, and I use those values to calculate later in my polling thread to notify of a period lapse. It's almost like the runtime is telling me the period has changed and the buffer size has changed. But this only happens in this case with a short wav file and running aplay over and over quickly. Is there a way to make sure the hw_ptr doesn't get affected by multiple instances of aplay trying to render out small clips. It seems like maybe I am setting up something incorrectly or maybe not blocking/ waiting correctly. I am at a loss in trying to figure this out, wondering if you would be so kind as to comment on my situation. I really appreciate your time in reading this email. Matt

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[alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, I downloaded and installed alsa-driver, alsa-lib and alsa-utils, but still no sound coming out of those speakers... I ran alsa-info.sh and the results are at http://pastebin.ca/451128 I am clueless as to what to do next, but I m ready to try anything. Any help would be greatly appreciated. Ben On 4/21/07, Tobin Davis < tdavis(a)dsl-only.net> wrote: > > Try downloading a current daily snapshot from ftp://ftp.suse.com/pub/projects/alsa/snapshot/ > . > > There was an issue with the ATI HD Audio detection that was fixed after > 14rc3 was released. > > Tobin > > On Sat, 2007-04-21 at 16:00 +1000, Ben DiDonc wrote: > > The output of alsa-info.sh is available at http://pastebin.ca/451007 > > > On 4/21/07, *Tobin Davis* < tdavis(a)dsl-only.net> wrote: > > Could you download an run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > , then post the resulting link? This will help determine what setup you > have. > > Tobin Davis > > > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. > Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I > just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) > and the only mixer settings available are those of the modem, named off-hook > and caller-id (see output of amixer, aplay -l , and the rest in attached > file output_feisty) I decided to upgrade alsa-driver and alsa-lib to > 1.0.14rc2. after downloading the package, I unloaded all sound related > modules (everything with "snd_" in it) in the right order and did: > ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; > sudo modprobe snd_hda_intel A minor improvement: the mixer now shows > controls for both the modem and the sound card (see > output_after_upgrade_to_rc2), but unfortunately, there is still no sound > coming out of those speakers (even though alsamixer shows control for PCM > and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it > a shot. This time again, only modem related settings appeared in alsamixer, > and no sound card stuff (see aplay and amixer in > output_after_upgrade_to_rc3), and no sound either. Does anyone know how to > get some sound out of these speakers (given that it used to work with > alsa1.0.12rc1...) ? Thanks for your help Ben > > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> > > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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Re: [alsa-devel] iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 22 Apr '07

22 Apr '07

[Nice analysis cut out..] At Tue, 03 Apr 2007 10:17:16 +0200, Dominique Dumont wrote: > > If the stream format is not valid when the digi converter is set up, > there's a chance that the digi converter goes belly up. > > I reckon that setting the stream *then* the digi converter would be safer. OK, now question is whether we need to turn off DIGI_CONV* once before setting it up again. Anyway, a test patch is below. It forces to set DIGI_CONV* verbs in prepare. Give it a try. thanks, Takashi diff -r 05ecca0fba92 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_codec.c Tue Apr 03 15:28:47 2007 +0200 @@ -1083,6 +1083,19 @@ static int snd_hda_spdif_default_put(str return change; } +static void setup_digi_convert(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned short val = codec->spdif_ctls; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | + AC_AMP_SET_OUTPUT | + ((val & 1) ? 0 : 0x80)); +} + static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1114,10 +1127,7 @@ static int snd_hda_spdif_out_switch_put( change = codec->spdif_ctls != val; if (change || codec->in_resume) { codec->spdif_ctls = val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | - AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); + setup_digi_convert(codec, nid); } mutex_unlock(&codec->spdif_mutex); return change; @@ -1901,6 +1911,21 @@ int snd_hda_multi_out_dig_open(struct hd return 0; } +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + mutex_lock(&codec->spdif_mutex); + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} + /* * release the digital out */ @@ -1949,6 +1974,8 @@ int snd_hda_multi_out_analog_prepare(str mout->dig_out_used = 0; snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); } + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); } mutex_unlock(&codec->spdif_mutex); diff -r 05ecca0fba92 pci/hda/hda_local.h --- a/pci/hda/hda_local.h Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_local.h Tue Apr 03 15:31:29 2007 +0200 @@ -148,6 +148,11 @@ struct hda_multi_out { int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout); +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, diff -r 05ecca0fba92 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/patch_realtek.c Tue Apr 03 15:32:23 2007 +0200 @@ -1916,6 +1916,17 @@ static int alc880_dig_playback_pcm_open( return snd_hda_multi_out_dig_open(codec, &spec->multiout); } +static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1984,7 +1995,8 @@ static struct hda_pcm_stream alc880_pcm_ /* NID is set in alc_build_pcms */ .ops = { .open = alc880_dig_playback_pcm_open, - .close = alc880_dig_playback_pcm_close + .close = alc880_dig_playback_pcm_close, + .prepare = alc880_dig_playback_pcm_prepare }, };

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[alsa-devel] Please ignore, test.
by Rene Herman 22 Apr '07

22 Apr '07

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