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[alsa-devel] OSS app works but aplay does not with SNDRV_PCM_FMTBIT_U32_LE
by Steve Strobel 21 Sep '07

21 Sep '07

I am developing an ALSA driver for the AD1938 codec on uClinux for Blackfin (I don't believe my problem is specific to that platform). When I set up the snd_pcm_hardware_t structure with .formats = SNDRV_PCM_FMTBIT_S32_LE, I can run both "aplay < /dev/urandom" and an OSS test application ("tone") that sends a sine wave, but the sine wave is "inside out" (see <http://www.link-comm.com/temp/inside_out.gif> for a screen shot) because the codec needs unsigned data. If I set up the snd_pcm_hardware_t structure with .formats = SNDRV_PCM_FMTBIT_U32_LE, the OSS test application sends a nice sine wave, but aplay fails with the following messages: root:~> aplay < /dev/urandom ALSA sound/blackfin/ad1836.c:1050: snd_ad1836_playback_open Playing raw data 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono snd_pcm_hw_refine() ACCESS = 00000000ffffffffffffffff -> 0000000000000008 FORMAT = 0000000000000400 -> 0000000000000000 snd_pcm_hw_refine() ACCESS = 333f8fc000a32c60c0333f8fc -> 333f8fc000a00000008 ALSA lib ALSA sound/blackfin/ad1836.c:1172: snd_ad1836_hw_free ../../../src/pcm/pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable aplay: set_params:882: Broken configuration for this PCM: no configurations available I am working my way through the ALSA source, but I have not yet figured out why aplay won't work with the unsigned codec configuration. Any suggestions? Steve P.S. A more complete log of the console output with both settings for those commands and most of the info from aadebug can be found at the following URLs: <http://www.link-comm.com/temp/SNDRV_PCM_FMTBIT_S32_LE.txt> <http://www.link-comm.com/temp/SNDRV_PCM_FMTBIT_U32_LE.txt> --- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

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[alsa-devel] Pending patches for 1.0.15
by Takashi Iwai 21 Sep '07

21 Sep '07

Hi, we're planning to release 1.0.15rc3 tomorrow. This should be the last candidate before 1.0.15 (hopefully). If you have pending "hotfix" patches, please submit as soon as possible. Thanks. Takashi

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[alsa-devel] [PATCH] alsa: Add the MCP79 support to hda_intel driver
by Peer Chen 21 Sep '07

21 Sep '07

Resend the patch for description modification and send to correct alsa maillist. Add the MCP79 support to hda driver. The patch base on kernel 2.6.23-rc7 Signed-off-by: Peer Chen <peerchen(a)gmail.com> --- --- linux-2.6.23-rc7/sound/pci/hda/hda_intel.c.orig 2007-09-20 15:40:51.000000000 -0400 +++ linux-2.6.23-rc7/sound/pci/hda/hda_intel.c 2007-09-21 13:50:33.000000000 -0400 @@ -1791,6 +1791,10 @@ static struct pci_device_id azx_ids[] = { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0ac0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); - -------------- Peer Chen 2007-09-21

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[alsa-devel] [PATCH] sun-cs4231: mce_down cleanup after ad1848
by Krzysztof Helt 20 Sep '07

20 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This simplifies and fixes waiting loops of the mce_down() function after Trent Piepho's patch for AD1848. It also makes busy_wait() function call not atomic. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-ref/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-ref/sound/sparc/cs4231.c 2007-09-19 23:31:30.603838619 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-19 23:23:24.552140109 +0200 @@ -392,62 +392,45 @@ static void snd_cs4231_mce_up(struct snd static void snd_cs4231_mce_down(struct snd_cs4231 *chip) { - unsigned long flags; - int timeout; + unsigned long flags, timeout; + int reg; - spin_lock_irqsave(&chip->lock, flags); snd_cs4231_busy_wait(chip); + spin_lock_irqsave(&chip->lock, flags); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) snd_printdd("mce_down [%p] - auto calibration time out (0)\n", CS4231U(chip, REGSEL)); #endif chip->mce_bit &= ~CS4231_MCE; - timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); - __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), + reg = __cs4231_readb(chip, CS4231U(chip, REGSEL)); + __cs4231_writeb(chip, chip->mce_bit | (reg & 0x1f), CS4231U(chip, REGSEL)); - if (timeout == 0x80) - snd_printdd("mce_down [%p]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0) { + if (reg == 0x80) + snd_printdd("mce_down [%p]: serious init problem " + "- codec still busy\n", chip->port); + if ((reg & CS4231_MCE) == 0) { spin_unlock_irqrestore(&chip->lock, flags); return; } - snd_cs4231_busy_wait(chip); - /* calibration process */ + /* + * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low. + */ - snd_cs4231_ready(chip); - snd_cs4231_ready(chip); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); + timeout = jiffies + msecs_to_jiffies(250); + do { spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - /* in 10ms increments, check condition, up to 250ms */ - timeout = 25; - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { - spin_unlock_irqrestore(&chip->lock, flags); - if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (2)\n"); - return; - } - msleep(10); - spin_lock_irqsave(&chip->lock, flags); - } - - /* in 10ms increments, check condition, up to 100ms */ - timeout = 10; - while (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) { - spin_unlock_irqrestore(&chip->lock, flags); - if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (3)\n"); - return; - } - msleep(10); + msleep(1); spin_lock_irqsave(&chip->lock, flags); - } + reg = snd_cs4231_in(chip, CS4231_TEST_INIT); + reg &= CS4231_CALIB_IN_PROGRESS; + } while (reg && time_before(jiffies, timeout)); spin_unlock_irqrestore(&chip->lock, flags); + + if (reg) + snd_printk(KERN_ERR + "mce_down - auto calibration time out (2)\n"); } static void snd_cs4231_advance_dma(struct cs4231_dma_control *dma_cont, ---------------------------------------------------------------------- Bedac w toalecie korzystala z ... >>> http://link.interia.pl/f1bd8

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[alsa-devel] Realtek ALC660-VD jack sense
by Vladimir Pouzanov 20 Sep '07

20 Sep '07

Hello all, I wonder if it could be possible to somehow fix jack sense detection on Realtek ALC660-VD and speaker output. When I plug in headphones speakers are still working and there's no way to separately mute them.

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[alsa-devel] [PARTIAL PATCH] snd-hda-intel on Medion WIM2160
by Bernhard Rosenkraenzer 20 Sep '07

20 Sep '07

Hi, On Medion WIM2160 notebooks (soundchip PCI ID 8086:27d8/17c0:4071), snd-hda-intel loads and everything acts like it's working, but there's no sound. The master volume is muted and there's no way to unmute it. The patch below fixes it, but unfortunately, Medion MD2 uses the same PCI (and subvendor/subdevice) ID -- so the patch quite likely breaks Medion MD2 support. WIM2160's soundchip has revision ID 02 - if MD2 uses something else, maybe that can be used to identify them and do the right thing. (I don't have an MD2...). ttyl bero --- linux-2.6.22/sound/pci/hda/patch_realtek.c.wim2160snd~ 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22/sound/pci/hda/patch_realtek.c 2007-09-09 20:28:55.000000000 +0200 @@ -159,6 +159,7 @@ ALC883_ACER, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -5995,6 +5996,32 @@ { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6364,6 +6391,7 @@ [ALC883_ACER] = "acer", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -6401,6 +6429,7 @@ SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + //SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + SND_PCI_QUIRK(0x17c0, 0x4071, "Medion WIM2160", ALC883_MEDION_WIM2160), {} }; @@ -6530,6 +6559,19 @@ .unsol_event = alc883_medion_md2_unsol_event, .init_hook = alc883_medion_md2_automute, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_lenovo_nb0763_capture_source, + .unsol_event = alc883_medion_md2_unsol_event, + .init_hook = alc883_medion_md2_automute, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer },

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[alsa-devel] alsa driver #2234 SoundScape Elite not support
by Krzysztof Helt 20 Sep '07

20 Sep '07

Hi, I looked into the ALSA bug tracking system and found the bug that SoundScape Elite is not supported by the sscape driver. I don't know how to contact the reporter of the bug (tjallen) so I am posting here. I would like to ask the reporter to test the current driver which support VIVO cards. One of changes is delay to wait for codec to be ready (delay suggested by Rene Herman but the OSS code had long waiting loop in the probe function as well). The current driver could work on the SoundScape Elite. The driver will be released as part of alsa-driver-1.0.15-rc3 in few days. Regards, Krzysztof ---------------------------------------------------------------------- Gra - zostan agentem ABW! Zobacz >>> http://link.interia.pl/f1ba5

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[alsa-devel] resetting an audio device?
by william estrada 20 Sep '07

20 Sep '07

Hi guys, I got my alsa player code to work. Now I need to figure out how to reset the audio device? I don't want to close the device, just recycle it. Any pointers? Also, where are the documents for all of those helper subroutes? Like snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_any, etc. Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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Re: [alsa-devel] resetting an audio device?
by william estrada 20 Sep '07

20 Sep '07

I don't need item one, it works now. But I would like to see the better API docs. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum > Hi guys, > > I got my alsa player code to work. Now I need to figure out how to > reset the audio device? I don't want to close the device, just recycle > it. Any pointers? > > Also, where are the documents for all of those helper subroutes? Like > snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_any, etc. > > Thanks for your time.

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[alsa-devel] [PATCH] opti93x: opti931 + QS1000 wavetable
by Krzysztof Helt 19 Sep '07

19 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch adds support for a wavetable chip on the BTC 1817DW board. The QS1000 is connected through the digital input to the Opti931 chip. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is not a hot-fix and can be postponed after 1.0.15 (before Takashi asks). A cornerstone of this patch is to set the auto-source selection for the FM DAC. This way it does not affects boards using the FM OPL synthesizers which do not use the digital input. If any other wavetable is connected to the opti chip through the digital input, the format setting (MC_REG(21)) should be revised if it is correct for the new board. However, in the whole opti92x-ad1848 this register was never touched (so probably nobody ever used the digital input ). Regards, Krzysztof diff -urp linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.23/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-12 13:33:51.000000000 +0200 +++ linux-2.6.23/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-19 22:13:02.000000000 +0200 @@ -501,6 +501,16 @@ static int __devinit snd_opti9xx_configu (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), 0x34); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); + /* + * The BTC 1817DW has QS1000 wavetable which is connected + * to the serial digital input of the OPTI931. + */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(21), 0x82, 0xff); + /* + * This bit sets OPTI931 to automaticaly select FM + * or digital input signal. + */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); break; #endif /* OPTi93X */ ---------------------------------------------------------------------- Zaaresztuj ministra Koszmarka! http://link.interia.pl/f1ba6

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