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[alsa-devel] [PATCH] ALSA: hda - add support for Intel DX58SO board
by Wu Fengguang 12 Jan '09

12 Jan '09

The Intel DX58SO board works fine with model ALC883_3ST_6ch_INTEL. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) --- mm.orig/sound/pci/hda/patch_realtek.c +++ mm/sound/pci/hda/patch_realtek.c @@ -8522,6 +8522,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} };

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Re: [alsa-devel] [PATCH] wss-lib: move AD1845 frequency setting into wss-lib
by Krzysztof Helt 11 Jan '09

11 Jan '09

On Sun, 11 Jan 2009 08:28:13 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > The Soundscape Elite does work, but after a reboot > I noticed that there is an oddness in that there's > a little "pop" before playing sounds with aplay and > a series of short spike pops before the sound plays > with xmms, which uses the old oss interface. > This is imperfection of early codecs. Sequences for changing sound format and rate are not yet perfect enough to avoid these "pops". Newer codecs mutes sound during such changes to avoid pops e.g. Soundscape VIVO with AD1845 does not produce "pops" before playing sounds. This is something of low priority and hard to master (guess) so do not expect solution soon. Regards, Krzysztof ---------------------------------------------------------------------- Szef przynudza? Zagraj sobie! Kliknij >> http://link.interia.pl/f200e

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[alsa-devel] The Dark Art of Sound on Linux
by A. C. Censi 11 Jan '09

11 Jan '09

This is of interest: http://glyph.twistedmatrix.com/2009/01/dark-art-of-sound-on-linux.html -- A. C. Censi accensi [em] gmail [ponto] com accensi [em] montreal [ponto] com [ponto] br

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[alsa-devel] [PATCH] wss-lib: move AD1845 frequency setting into wss-lib
by Krzysztof Helt 11 Jan '09

11 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is required to allow the sscape driver to autodetect installed codec. Also, do not create a timer if detected codec has no hardware timer (e.g. AD1848). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> Cc: Rene Herman --- Ted, please test if the patch helps to make your Soundscape Elite card working. If not the patch is required anyway as your last feedback showed that frequency setting is incorrect in the sscape driver if the codec is not the AD1845. sound/isa/sscape.c | 113 +++------------------------------------------- sound/isa/wss/wss_lib.c | 40 +++++++++++++++++ 2 files changed, 48 insertions(+), 105 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 48a16d8..bc44916 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -129,9 +129,6 @@ enum GA_REG { #define DMA_8BIT 0x80 -#define AD1845_FREQ_SEL_MSB 0x16 -#define AD1845_FREQ_SEL_LSB 0x17 - enum card_type { SSCAPE, SSCAPE_PNP, @@ -955,82 +952,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l /* - * Override for the CS4231 playback format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_playback_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the PLAYBACK_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* - * Override for the CS4231 capture format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_capture_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the CAPTURE_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* * Create an AD1845 PCM subdevice on the SoundScape. The AD1845 * is very much like a CS4231, with a few extra bits. We will * try to support at least some of the extra bits by overriding @@ -1055,11 +976,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -#define AD1845_FREQ_SEL_ENABLE 0x08 - -#define AD1845_PWR_DOWN_CTRL 0x1b -#define AD1845_CRYS_CLOCK_SEL 0x1d - /* * It turns out that the PLAYBACK_ENABLE bit is set * by the lowlevel driver ... @@ -1074,7 +990,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ if (sscape->type != SSCAPE_VIVO) { - int val; /* * The input clock frequency on the SoundScape must * be 14.31818 MHz, because we must set this register @@ -1082,22 +997,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); + snd_wss_out(chip, AD1845_CLOCK, 0x20); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); - /* - * More custom configuration: - * a) select "mode 2" and provide a current drive of 8mA - * b) enable frequency selection (for capture/playback) - */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, - CS4231_MODE2 | 0x10); - val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL); - snd_wss_out(chip, AD1845_PWR_DOWN_CTRL, - val | AD1845_FREQ_SEL_ENABLE); - spin_unlock_irqrestore(&chip->reg_lock, flags); } err = snd_wss_pcm(chip, 0, &pcm); @@ -1113,11 +1016,13 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "for AD1845 chip\n"); goto _error; } - err = snd_wss_timer(chip, 0, NULL); - if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " - "for AD1845 chip\n"); - goto _error; + if (chip->hardware != WSS_HW_AD1848) { + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No timer device " + "for AD1845 chip\n"); + goto _error; + } } if (sscape->type != SSCAPE_VIVO) { @@ -1128,8 +1033,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "MIDI mixer control\n"); goto _error; } - chip->set_playback_format = ad1845_playback_format; - chip->set_capture_format = ad1845_capture_format; } strcpy(card->driver, "SoundScape"); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 3d6c5f2..13299ae 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -646,6 +646,24 @@ static void snd_wss_playback_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the playback stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (pdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -690,6 +708,24 @@ static void snd_wss_capture_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the capture stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_REC_FORMAT, (cdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -1314,6 +1350,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } + /* enable fine grained frequency selection */ + if (chip->hardware == WSS_HW_AD1845) + chip->image[AD1845_PWR_DOWN] = 8; + ptr = (unsigned char *) &chip->image; regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32; snd_wss_mce_down(chip); -- 1.5.2.2 ---------------------------------------------------------------------- Promocja w Speak Up. 3 miesiace angielskiego gratis. Sprawdz teraz i wypelnij formularz! >> http://link.interia.pl/f2019

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[alsa-devel] [patch] Indigo DJx
by carmen 11 Jan '09

11 Jan '09

the old Indigo DJ driver worked after a few tweaks... tested w/ jack, cmus, mplayer mapping/virtual-outs seem good.. this is a diff on alsa-kernel.git. copied the old firmware and amazed it works, although something seems wonky with the samplerate registers. going to email echo about an updated firmware .. no noise bursts or ear destruction tho!

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Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 11 Jan '09

11 Jan '09

At Sat, 10 Jan 2009 09:34:57 -0500, Kyler Laird wrote: > > On Fri, Jan 09, 2009 at 05:14:45PM +0100, Takashi Iwai wrote: > > > Could you try test/cf74-fix branch on sound git tree? > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/cf74-fix > > That didn't work for me. I see: > HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ > 22 > HDA Intel 0000:00:1b.0: setting latency timer to 64 > input: HDA Intel at 0xe8640000 irq 22 Line In at Ext Rear Jack > as /class/input/input7 > input: HDA Intel at 0xe8640000 irq 22 Mic at Ext Front Jack as > /class/input/input8 > input: HDA Intel at 0xe8640000 irq 22 Mic at Ext Rear Jack as > /class/input/input9 > input: HDA Intel at 0xe8640000 irq 22 Speaker at Ext Rear Jack > as /class/input/input10 > input: HDA Intel at 0xe8640000 irq 22 HP Out at Ext Front Jack > as /class/input/input11 These are all OK. The new messages regarding input device are for jack-sensing stuff. Could you run alsa-info.sh on the latest kernel, too? Also, to be sure, try to put a printk in the quirk for CF-74 (disabling hp_detect) to confirm it's activated. thanks, Takashi

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[alsa-devel] [PATCH 0/2] ASoC: MioA701 board
by Robert Jarzmik 09 Jan '09

09 Jan '09

Hi, These 2 patches have been already reviewed in April 2008 and have been siting in Mark's asoc-v2 tree since then. They have been ported back to asoc-v1 for mainline submission. Please consider them for review if need be and scheduling for the next queue. The serie was build upon Takashi's topic/asoc branch, commit "ASoC: Clean up kerneldoc warnings" of id ac11a2b35cc25c77d28218aaf60e7f7c6c7ee5d3. Cheers. -- Robert Robert Jarzmik (2): Add initial support of Mitac mioa701 device SoC. Add initial support of Mitac mioa701 master volume. sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_masterctrl.c | 243 +++++++++++++++ sound/soc/pxa/mioa701_wm9713.c | 578 ++++++++++++++++++++++++++++++++++++ 4 files changed, 832 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/mioa701_masterctrl.c create mode 100644 sound/soc/pxa/mioa701_wm9713.c

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Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 09 Jan '09

09 Jan '09

At Fri, 9 Jan 2009 10:21:21 -0500, Kyler Laird wrote: > > On Fri, Jan 09, 2009 at 08:51:58AM +0100, Takashi Iwai wrote: > > > Weird. I thought this path is active only when hp_detect != 0. > > Could you check whether it's really correctly patched, e.g. by the > > patch like below? > > That patch works. Thanks! > > > It seems applying my previous patch on 2.6.28 puts the code into a > > wrong place. > > I haven't succeeded in getting either of your recent patches to work > against 2.6.28-git13. I'll work on it more later. The previous patch was for 2.6.28 while the first one was for git. Maybe just because of a line difference. Could you try test/cf74-fix branch on sound git tree? git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/cf74-fix Takashi

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[alsa-devel] alsa-plugin: rate-libsamplerate, buffer shrinkage problem
by John L. Utz III 09 Jan '09

09 Jan '09

Hi; Our legacy product code is tormenting me again. :-( i have a 3rd party sound-fiddling library that only works with an 8k buffer. the company that wrote it is long gone. i cant change this. i have a enourmous array of content that is coded at 44.1K. i cant change this. i have to use a cheap soundblaster audigy card (ca0106) that only supports 48K. it sounds really good at 48K, so it seems like a winner. the samplerate plugin is almost perfect for helping me successfully use the 48K only card with our 44.1K content. *almost*. :-( if i use the samplerate plugin thusly in my CA0106.conf: CA0106.pcm.front.0 { @args [ CARD ] @args.CARD { type string } # type hw # card $CARD type rate slave { pcm "hw:0,0" rate 48000 } converter "samplerate" } I no longer get a 8192 byte buffer, i get a 7526 byte buffer. :-( snd_pcm_hw_params_set_buffer_size_near() fails if i try and set it to 8192, but it accepts 7526 or 15092. I *assume* that either the samplerate plugin or libsamplerate are consuming part of the buffer for the resample job. Not at all unreasonable. But it seems like i should be able to bump up the internal working buffer size so that i get my 8192 byte submittal buffer back. Does anybody have any suggestions? Is there a flag or config entry that i have missed? Or do i need to go hack something? That would be fine, but it would be nice to know where to look. :-) Here is a succinct example of the problem, first i call speaker-test without a rate argument and then with the rate argument: test-gntopxsg ~ # speaker-test -c2 -Dfront -t sin speaker-test 1.0.17 Playback device is front Stream parameters are 48000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 48000Hz (requested 48000Hz) Buffer size range from 32 to 16384 Period size range from 15 to 8193 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 test-gntopxsg ~ # speaker-test -c2 -Dfront -t sin -r44100 speaker-test 1.0.17 Playback device is front Stream parameters are 44100Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 44100Hz (requested 44100Hz) Buffer size range from 29 to 15052 Period size range from 14 to 7527 Using max buffer size 15052 Periods = 4 was set period_size = 3763 was set buffer_size = 15052 tnx in advance for any help you all can provide! johnu

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[alsa-devel] [PATCH 0/7] ASoc updates for -next
by Mark Brown 09 Jan '09

09 Jan '09

The following changes since commit 227b4dc6432d271eecd0ff0aefe6f0897ec47397: Mark Brown (1): ASoC: Fix SND_SOC_ALL_CODECS handling of dual SPI and I2C control buses are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Ian Molton (1): ASoC: dapm: Allow explictly named mixer controls Mark Brown (4): Merge branch 'asoc' of git://git.mnementh.co.uk/linux-2.6-im into for-tiwai ASoC: Constify pin names for DAPM pin status APIs ASoC: Add jack reporting interface ASoC: Implement WM8350 headphone jack detection Peter Ujfalusi (3): ASoC: Merge the soc_value_enum to soc_enum struct ASoC: TWL4030: Change the soc_value_enum back to soc_enum ASoC: TWL4030: Module unloading fix Documentation/sound/alsa/soc/dapm.txt | 3 + include/linux/mfd/wm8350/audio.h | 1 + include/sound/soc-dapm.h | 21 ++++- include/sound/soc.h | 47 ++++++++---- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/codecs/twl4030.c | 12 ++- sound/soc/codecs/wm8350.c | 116 +++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 8 ++ sound/soc/soc-core.c | 37 +-------- sound/soc/soc-dapm.c | 137 +++++++++++---------------------- sound/soc/soc-jack.c | 138 +++++++++++++++++++++++++++++++++ 12 files changed, 372 insertions(+), 151 deletions(-) create mode 100644 sound/soc/soc-jack.c

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