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[alsa-devel] [PATCH] opti93x: fix sound ouput for Opti930
by Krzysztof Helt 09 Jun '08

09 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes silenced output from the Opti930. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The bug was introduced by me in the patch which added QS1000 wavetable support: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h… Simply do not change digital input bits on Opti930. diff -urp linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c --- linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-07 23:59:27.218950875 +0200 @@ -491,16 +491,9 @@ static int __devinit snd_opti9xx_configu break; #else /* OPTi93X */ - case OPTi9XX_HW_82C930: case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | - (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. */ @@ -510,6 +503,13 @@ static int __devinit snd_opti9xx_configu * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); + case OPTi9XX_HW_82C930: /* FALL THROUGH */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | + (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), + 0x34); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); break; #endif /* OPTi93X */ --------------------------------------------------------------- Sprawdz jak zdobyc zdrowy usmiech! Kliknij >> http://link.interia.pl/f1e26

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[alsa-devel] [PATCH] Fix "alc262_sony_unsol[]" hda_verb array
by idak 09 Jun '08

09 Jun '08

I think that hda_verb array must have "terminator (empty array)". But alc262_sony_unsol[] does not have it. And it causes gcc-4.3's buggy behavior with snd_hda_sequence_write(). Signed-off-by: Akio Idehara <zbe64533(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) --- a/sound/pci/hda/patch_realtek.c 2008-06-08 19:23:57.000000000 +0900 +++ b/sound/pci/hda/patch_realtek.c 2008-06-08 19:42:19.000000000 +0900 @@ -8839,6 +8839,7 @@ static struct hda_verb alc262_sony_unsol {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} }; /* mute/unmute internal speaker according to the hp jack and mute state */

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[alsa-devel] [PATCH] Add Toshiba dynabook SS RX1 support
by Akio Idehara 09 Jun '08

09 Jun '08

I have Toshiba dynabook SS RX1 and this patch adds that support. Signed-off-by: Akio Idehara <zbe64533(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) --- a/sound/pci/hda/patch_realtek.c 2008-06-08 19:23:57.000000000 +0900 +++ b/sound/pci/hda/patch_realtek.c 2008-06-08 20:55:34.000000000 +0900 @@ -9718,6 +9718,8 @@ static struct snd_pci_quirk alc262_cfg_t SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),

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[alsa-devel] A little problem with main gitweb page
by Rafał Mużyło 08 Jun '08

08 Jun '08

Well, it would be nice if http://git.alsa-project.org/ listed alsa-plugins git too.

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[alsa-devel] Problems regarding alsa-info.sh
by Travis Place 08 Jun '08

08 Jun '08

Since the move from HG->GIT there is now a new download location for alsa-info.sh . There are 2 current problems with this though.. 1) The new URL is very ugly(and hard to post on wikis etc) The new link is http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=utils/alsa-in… 2) The current script version at HG repo (which is where people currently get it from, is an outdated version, that checks the HG repo for updates to the script. We need to put the new version of the script into HG (if its still accessible), with the updated 'update' URL, or, somehow redirect from the old URL to the new URL. Please let me know what should be done about this, and how to go about fixing it. Attached is the updated alsa-info.sh with the new GIT URLS (update and changelog) in place.

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[alsa-devel] Wrong period size interval with hda-intel makes snd_pcm_set_params() fail
by Christian Morales Vega 08 Jun '08

08 Jun '08

I looked deeper into the problem and looks like a hda-intel problem. This code fails with "ALSA lib pcm.c:7193:(snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" #include <alsa/asoundlib.h> int main(){ snd_pcm_t *handle; if(snd_pcm_open(&handle, "default:1", SND_PCM_STREAM_PLAYBACK, 0) < 0) { exit(1); } if(snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 24000, 1, 100000) < 0) { exit(2); } } The "Invalid argument" comes from pcm_params.c:145. The test "!snd_interval_single(i)" fails since the interval for SND_PCM_HW_PARAM_PERIOD_SIZE ends being min=511 and max=513. I understand that this interval is set at the driver, correct? I don't know where exactly, so some help in finding it or a patch would be welcome. I'm using alsa-driver-kmp-default-1.0.16.20080608, libasound2-1.0.16.hg20080510 and alsa-1.0.16.hg20080510 RPM packages from multimedia:audio openSUSE Build Service repository (from Takashi Iwai) and an AD1988 (hda-intel) codec. A XUbuntu user also has the same problem with an hda-intel codec (that I suppose uses stable 1.0.16). With a SB Live! in the same computer there are no problems.

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[alsa-devel] Previous patch to alsa-info.sh
by Travis Place 07 Jun '08

07 Jun '08

I sent of a patch for alsa-info.sh about a week or two ago now, and Im starting to think it lost amongst all the other posts. Unforunately my main system died, and is off getting warrantied, so i cant repost. Can someone look through the list and find the patch (it added a check to see if pastebin.ca was alive before posting to it, and falling back to --no-upload if it was down. Upped the version to 0.4.45 too i think) Thanks, Travis Place

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[alsa-devel] [PATCH][ASoC V2] Update MPC8610 drivers to pick the right DMA channels
by Timur Tabi 06 Jun '08

06 Jun '08

On the Freescale MPC8610 SOC, SSI1 is hard-wired to DMA channels 0 and 1, and SSI2 is hard-wired to DMA channels 2 and 3. Fix fsl_ssi.c to reflect that. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9d141ec..4f66e84 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -790,13 +790,13 @@ static int fsl_ssi_probe(struct of_device *ofdev, /* * Get the DMA information. If it's an older device tree (i.e. without * an "fsl,playback-dma" property), then we assume that SSI1 uses DMA1 - * Channels 0 and 1, and SSI2 uses DMA2 Channels 0 and 1. + * Channels 0 and 1, and SSI2 uses DMA1 Channels 2 and 3. */ dma_ph = of_get_property(np, "fsl,playback-dma", NULL); if (!dma_ph) { dev_warn(&ofdev->dev, "please update your device tree\n"); - dma_np[0] = find_dma_node(ssi_info->id, 0); - dma_np[1] = find_dma_node(ssi_info->id, 1); + dma_np[0] = find_dma_node(0, ssi_info->id ? 2 : 0); + dma_np[1] = find_dma_node(0, ssi_info->id ? 3 : 1); } else { dma_np[0] = of_find_node_by_phandle(*dma_ph); dma_ph = of_get_property(np, "fsl,capture-dma", NULL); -- 1.5.5

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[alsa-devel] [PATCH][ASoC V2] Update snd_soc_card_get_codec() to take the codec number
by Timur Tabi 06 Jun '08

06 Jun '08

Update function snd_soc_card_get_codec() and all the drivers that call it to take a codec number in addition to a codec name. This is necessary for codec drivers that register multiple instances of themselves that differ only by codec number. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- I have only tested this code on the MPC8610, so please test this on other platforms before committing. include/sound/soc-card.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- sound/soc/imx/imx32ads-wm8350.c | 2 +- sound/soc/pxa/amesom_tlv320.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/h5000.c | 2 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/mainstone_wm8753.c | 2 +- sound/soc/pxa/mainstone_wm9712.c | 2 +- sound/soc/pxa/mainstone_wm9713.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/soc-core.c | 5 +++-- 17 files changed, 23 insertions(+), 22 deletions(-) diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index 9b8600d..290f649 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -44,7 +44,7 @@ struct snd_ac97_bus_ops *snd_soc_card_get_ac97_ops( struct snd_soc_card *soc_card, const char *dai_id); struct snd_soc_codec *snd_soc_card_get_codec(struct snd_soc_card *soc_card, - const char *codec_id); + const char *codec_name, int codec_num); struct snd_soc_platform * snd_soc_card_get_platform( struct snd_soc_card *soc_card, const char *platform_id); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index bcee769..a70fffe 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -242,7 +242,7 @@ static int eti_b1_init(struct snd_soc_card *soc_card) struct at91_ssc_periph *ssc; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 45c3311..8813291 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -38,7 +38,6 @@ * MPC8610 HPCD. Some of the data is taken from the device tree. */ struct mpc8610_hpcd_data { - char codec_name[32]; struct snd_soc_card soc_card; struct ccsr_guts __iomem *guts; unsigned int num_configs; @@ -47,8 +46,6 @@ struct mpc8610_hpcd_data { char pcm_name[32]; char ssi_name[32]; char codec_name[32]; - char platform_name[32]; - char cpu_dai_name[32]; } names[MAX_SSI]; }; @@ -99,7 +96,8 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) for (i = 0; i < soc_card_data->num_configs; i++) { codec = snd_soc_card_get_codec(soc_card, - soc_card_data->names[i].codec_name); + soc_card_data->names[i].codec_name, + soc_card_data->configs[i].codec_num); if (!codec) { dev_err(soc_card->dev, "could not find codec\n"); @@ -111,7 +109,9 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) */ ret = snd_soc_card_init_codec(codec, soc_card); if (ret < 0) { - dev_err(soc_card->dev, "could not initialize codec\n"); + dev_err(soc_card->dev, + "could not initialize codec %s-%u\n", + codec->name, codec->num); continue; } } diff --git a/sound/soc/imx/imx32ads-wm8350.c b/sound/soc/imx/imx32ads-wm8350.c index feb83cc..94de723 100644 --- a/sound/soc/imx/imx32ads-wm8350.c +++ b/sound/soc/imx/imx32ads-wm8350.c @@ -407,7 +407,7 @@ int imx32_audio_init(struct snd_soc_card *soc_card) if (pcm_runtime == NULL) return -ENODEV; - codec = snd_soc_card_get_codec(soc_card, wm8350_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8350_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/amesom_tlv320.c b/sound/soc/pxa/amesom_tlv320.c index 9be4db0..c2ba22a 100644 --- a/sound/soc/pxa/amesom_tlv320.c +++ b/sound/soc/pxa/amesom_tlv320.c @@ -134,7 +134,7 @@ static int amesom_tlv320_init(struct snd_soc_card *soc_card) { struct snd_soc_codec *codec; - codec = snd_soc_card_get_codec(soc_card, tlv320_codec_id); + codec = snd_soc_card_get_codec(soc_card, tlv320_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index c24d76b..9f30f12 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -313,7 +313,7 @@ static int corgi_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 987106a..9159d03 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -50,7 +50,7 @@ static int e800_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/h5000.c b/sound/soc/pxa/h5000.c index 54ff23c..fa340f4 100644 --- a/sound/soc/pxa/h5000.c +++ b/sound/soc/pxa/h5000.c @@ -272,7 +272,7 @@ static int h5000_ak4535_init (struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, ak4535_codec_id); + codec = snd_soc_card_get_codec(soc_card, ak4535_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index ad6dea7..0d7b145 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -436,7 +436,7 @@ static int magician_uda1380_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, uda1380_codec_id); + codec = snd_soc_card_get_codec(soc_card, uda1380_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/mainstone_wm8753.c b/sound/soc/pxa/mainstone_wm8753.c index f7ab6fb..2bef9e9 100644 --- a/sound/soc/pxa/mainstone_wm8753.c +++ b/sound/soc/pxa/mainstone_wm8753.c @@ -328,7 +328,7 @@ static int mainstone_wm8753_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8753_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8753_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/mainstone_wm9712.c b/sound/soc/pxa/mainstone_wm9712.c index b87362b..f2ded1d 100644 --- a/sound/soc/pxa/mainstone_wm9712.c +++ b/sound/soc/pxa/mainstone_wm9712.c @@ -81,7 +81,7 @@ static int mainstone_wm9712_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/mainstone_wm9713.c b/sound/soc/pxa/mainstone_wm9713.c index 9492b3e..f9723e1 100644 --- a/sound/soc/pxa/mainstone_wm9713.c +++ b/sound/soc/pxa/mainstone_wm9713.c @@ -163,7 +163,7 @@ static int mainstone_wm9713_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9713_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9713_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4ce0ce4..126ba5b 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -264,7 +264,7 @@ static int poodle_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 497b3b2..393adda 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -315,7 +315,7 @@ static int spitz_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8750_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8750_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 77688e2..18e8e4b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -211,7 +211,7 @@ static int tosa_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index a7e8ab7..4209fc2 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -93,7 +93,7 @@ static int zylonite_init(struct snd_soc_card *card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(card, wm9713_codec_id); + codec = snd_soc_card_get_codec(card, wm9713_codec_id, 0); if (codec == NULL) { printk(KERN_ERR "Unable to obtain WM9713 codec\n"); return -ENODEV; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 520469b..a88e3a0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2172,12 +2172,13 @@ EXPORT_SYMBOL_GPL(snd_soc_card_free); * Get a codec from a codec ID. */ struct snd_soc_codec *snd_soc_card_get_codec(struct snd_soc_card *soc_card, - const char *codec_id) + const char *codec_name, int codec_num) { struct soc_pcm_config *config; list_for_each_entry(config, &soc_card->config_list, list) { - if (config->codec && !strcmp(config->codec->name, codec_id)) + if (config->codec && !strcmp(config->codec->name, codec_name) + && config->codec->num == codec_num) return config->codec; } return NULL; -- 1.5.5

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[alsa-devel] [PATCH] ASoC: Fix default mono mixer configuration for WM8510
by Mark Brown 06 Jun '08

06 Jun '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6f54cb0..152e6f2 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -71,7 +71,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0003, 0x0010, 0x0000, 0x0000, 0x0000, 0x0002, 0x0001, 0x0000, 0x0000, 0x0000, 0x0039, 0x0000, - 0x0000, + 0x0001, }; /* -- 1.5.5.3

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