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[alsa-devel] save alsa buffer
by arif setiawan 10 Jul '09

10 Jul '09

Hi, I am trying to save content of alsa buffer to file for playing it in Matlab.I choose Matlab for prototyping. I run full duplex ALSA based on code from http://www.bel.fi/~alankila/blog/2005/08/21/Full Duplex ALSA.html<http://www.bel.fi/%7Ealankila/blog/2005/08/21/Full%20Duplex%20ALSA.html> I run alsa on Beagleboard. I want to save buffer content to a file, but it failed with error: Segmentation fault when reach fclose(fp). I want to know what kind of data that ALSA processes so I can implement my LPC vocoder. Thank you. Best regards, Arif

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[alsa-devel] runtime->boundary
by Harsha, Priya 10 Jul '09

10 Jul '09

Hi, Can anyone help explain what is runtime->boundary and what is the value that is set for this? It seems to be a pretty huge value when I try and print it. Also in snd_pcm_indiret_capture_transfer, I see this code snippet; If (diff < runtime->boundary /2) diff += runtime->boundary what is the reason for the same? If its for handling ring buffer wrapping, should I be setting my runtime->boundary to rumtime->buffer_size in the driver code? Thanks, Harsha

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[alsa-devel] No incoming sound indigo iox
by Susan Cragin 10 Jul '09

10 Jul '09

I just purchased the indigoaudio iox. Outgoing shows up on Volume Control, incoming does not. In Audacity, incoming shows up as: ALSA: Indigo IOx (hw:1,0) but recording does not work. Here's the catch. My problem may be due to my kernel. I am running Ubuntu-Studio and have the 29.5-2-rt experimental kernel, which is the 5-1 recompiled with alsadriver support for 30 kernels. So you may just tell me to go away until the kernel's better, but I don't think the kernel is the problem. FYI winecfg says: fixme:mixer:ALSA_MixerInit No master control found on Indigo IOx, disabling mixer

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[alsa-devel] targa and targa-2ch fix for 2.6.31
by okias 09 Jul '09

09 Jul '09

Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id 64a8be74357477558183b43156c5536b642de134 Signed-off-by: David Heidelberger <d.okias(a)gmail.com> --- --- sound/pci/hda/patch_realtek.c 2009-07-09 17:36:37.593323363 +0200 +++ sound/pci/hda/patch_realtek.c 2009-07-09 17:57:31.014574807 +0200 @@ -6895,9 +6895,6 @@ static struct hda_verb alc882_targa_verb {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -7217,7 +7214,8 @@ static struct alc_config_preset alc882_p }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs, + alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -9212,7 +9210,8 @@ static struct alc_config_preset alc883_p }, [ALC883_TARGA_DIG] = { .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9225,7 +9224,8 @@ static struct alc_config_preset alc883_p }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_targa_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, -- Jabber/XMPP: okias(a)isgeek.info SIP VoIP: sip:17474537254@proxy01.sipphone.com

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[alsa-devel] [PATCH] snd-hda-intel: fix beep tone calculation for IDT/STAC codecs
by Paul Vojta 09 Jul '09

09 Jul '09

In the beep tone calculation for IDT/STAC codecs, lower numbers correspond to higher frequencies and vice versa. The current code has this backwards, resulting in beep frequencies which are way too high (and sound bad on tinny laptop speakers, resulting in complaints). Signed-off-by: Paul Vojta <vojta(a)math.berkeley.edu> diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 08fe659..d559a7a 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -51,19 +51,20 @@ static void snd_hda_generate_beep(struct work_struct *work) * The tone frequency of beep generator on IDT/STAC codecs is * defined from the 8bit tone parameter, in Hz, * freq = 48000 * (257 - tone) / 1024 - * that is from 12kHz to 93.75kHz in step of 46.875 hz + * that is from 12kHz to 93.75Hz in steps of 46.875 Hz */ static int beep_linear_tone(struct hda_beep *beep, int hz) { hz *= 1000; /* fixed point */ - hz = hz - DIGBEEP_HZ_MIN; + hz = hz - DIGBEEP_HZ_MIN + + DIGBEEP_HZ_STEP / 2; /* round to nearest step */ if (hz < 0) hz = 0; /* turn off PC beep*/ else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) - hz = 0xff; + hz = 1; /* max frequency */ else { hz /= DIGBEEP_HZ_STEP; - hz++; + hz = 255 - hz; } return hz; }

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[alsa-devel] [PATCH] [resend] cmi8330: find OPL3 port automatically
by Ondrej Zary 09 Jul '09

09 Jul '09

My CMI8329 had OPL3 port specified in SB16 resources. But now I found out that it was my modification of the card's PnP EEPROM a couple of years ago (can be done using C9SETROM.EXE utility). I did it because the OPL3 port was completely missing from PnP data. It seems to be hardwired to 0x388 on CMI8329. Find OPL3 port automatically by searching in WSS and SB16 resources. If not found, assume that it's hardwired to 0x388. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-test/sound/isa/cmi8330.c 2009-07-06 21:18:06.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-07-06 21:45:29.000000000 +0200 @@ -362,7 +362,7 @@ wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - if (acard->type == CMI8330) + if (pnp_port_start(pdev, 1)) fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ @@ -377,8 +377,13 @@ sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); - if (acard->type == CMI8329) - fmport[dev] = pnp_port_start(pdev, 1); + /* On CMI8239, the OPL3 port might be present in SB16 PnP resources */ + if (fmport[dev] == SNDRV_AUTO_PORT) { + if (pnp_port_start(pdev, 1)) + fmport[dev] = pnp_port_start(pdev, 1); + else + fmport[dev] = 0x388; /* Or hardwired */ + } /* allocate MPU-401 resources */ pdev = acard->mpu; -- Ondrej Zary

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 09 Jul '09

09 Jul '09

The following changes since commit 4ec5c9693b46ab34976511cd43bf75c3a0fc704d: Mark Brown (1): Merge branch 'for-2.6.31' into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Chaithrika U S (4): ASoC: Introduce platform driver model for dm644x, dm355 ASoC: Add mcasp support for DM646x ASoC: Add machine driver support for DM646x ASoC: Minor fixes to DaVinci I2S probe function Mark Brown (5): Merge branch 'davinci' into for-2.6.32 ASoC: Add TLV information for WM8731 ASoC: Correct WM8731 Mic Capture Switch control name ASoC: DaVinci I2S needs mach/asp.h ASoC: Add WM8993 CODEC driver Troy Kisky (1): ASoC: DaVinci: pcm, don't play 1st sound period twice include/sound/wm8993.h | 44 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8731.c | 17 +- sound/soc/codecs/wm8993.c | 2203 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8993.h | 2132 +++++++++++++++++++++++++++++++++++ sound/soc/davinci/Kconfig | 13 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-evm.c | 102 +- sound/soc/davinci/davinci-i2s.c | 123 ++- sound/soc/davinci/davinci-mcasp.c | 874 +++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 55 + sound/soc/davinci/davinci-pcm.c | 1 + sound/soc/davinci/davinci-pcm.h | 18 +- 14 files changed, 5473 insertions(+), 118 deletions(-) create mode 100644 include/sound/wm8993.h create mode 100644 sound/soc/codecs/wm8993.c create mode 100644 sound/soc/codecs/wm8993.h create mode 100644 sound/soc/davinci/davinci-mcasp.c create mode 100644 sound/soc/davinci/davinci-mcasp.h

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[alsa-devel] Choosing correct nid and xindex for hda_intel realtek patch for MacBook 5.1
by Chris Lasher 08 Jul '09

08 Jul '09

I'm trying to fix support for my MacBook 5.1. A patch by Kacper is getting me part of the way there but it seems my pin configuration is different than his model of MacBook. I'm trying to wire up my pins to his correctly. I am trying to figure out the bits of snd_kcontrol_new (in alsa-driver-1.0.19/alsa-kernel/pci/hda/patch_realtek.c) for the HDA_CODEC_VOLUME and HDA_BIND_MUTE. There is: string_description, nid , xindex, HDA_INPUT/HDA_OUTPUT I'm trying to figure out how to choose the correct values for nid and xindex. Actually, I'm really wondering how to identify the right xindex for a given nid, as I realize the nid's come from the Node #s from the output of /proc/asound/card0/codec#0. I also need some assistance with regards to the hda_verb settings for modifying the alc885_mb5_init_verbs section; if anyone can offer pointers there, I'd appreciate it. This is directly related to Bug #337314 in Launchpad https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/337314?comments=… and ALSA bug 4432: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4432 Kacper's patch is linked to in both bug reports. The output of my soundcard's codec#0 is attached to the Launchpad report https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/337314/comments/… The output from a soundcard that does seem to have full support by Kacper's patch is available at https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/337314/comments/… Any help would be much appreciated. Thanks, Chris

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[alsa-devel] [PATCH] ALSA: ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free
by Daniel Mack 08 Jul '09

08 Jul '09

Check for rtd->params->drcmr != NULL before accessing it. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 108b643..6205f37 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - if (rtd && rtd->params) + if (rtd && rtd->params && rtd->params->drcmr) *rtd->params->drcmr = 0; snd_pcm_set_runtime_buffer(substream, NULL); -- 1.6.3.1

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[alsa-devel] [PATCH] ASoC: DaVinci: pcm, don't play 1st sound period twice
by Troy Kisky 08 Jul '09

08 Jul '09

Update the dma link with correct data as soon as the master channel has copied it. Otherwise, the 1st period will play twice. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a059965..8fd0c3c 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -206,6 +206,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) /* Copy self-linked parameter RAM entry into master channel */ edma_read_slot(prtd->slave_lch, &temp); edma_write_slot(prtd->master_lch, &temp); + davinci_pcm_enqueue_dma(substream); return 0; } -- 1.5.6.3

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