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[alsa-devel] [PATCH] CA0106 on MSI K8N Diamond PLUS Motherboard
by Travis Place 25 Aug '08

25 Aug '08

Previously i posted a patch for a new ssid for the onboard ca0106 on the K8N Diamond motherboard. Now, it is apparently really called a 'K8N Diamond PLUS', and also requires the gpio_type to be changed too. Attached is a patch for this. Changelog: Correct a previous patch for the ca0106 onboard the MSI K8N Diamond PLUS motherboard. Confirmed to have Line/Mic/Aux working for input, and sound output working as expected. Signed-off-by: Travis Place <wishie(a)wishie.net>

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[alsa-devel] [PATCH 001/002] snd-powermac: mixers for PowerMac G4 AGP
by Risto Suominen 25 Aug '08

25 Aug '08

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[alsa-devel] [PATCH] wss_lib: do not mess mixer settings during probe
by Krzysztof Helt 25 Aug '08

25 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss_dout function which does not mess shadowed register values during chip probing. Otherwise, user ends up with stupid mixer settings after driver loading. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Especially, the recording settings are messed up like left channel recording from different source than right channel. diff -urp linux-ref/sound/isa/wss/wss_lib.c linux-2.6/sound/isa/wss/wss_lib.c --- linux-ref/sound/isa/wss/wss_lib.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/wss/wss_lib.c 2008-08-25 21:13:14.000000000 +0200 @@ -1162,9 +1162,9 @@ static int snd_ad1848_probe(struct snd_w spin_lock_irqsave(&chip->reg_lock, flags); /* set CS423x MODE 1 */ - snd_wss_out(chip, CS4231_MISC_INFO, 0); + snd_wss_dout(chip, CS4231_MISC_INFO, 0); - snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */ + snd_wss_dout(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */ r = snd_wss_in(chip, CS4231_RIGHT_INPUT); if (r != 0x45) { /* RMGE always high on AD1847 */ @@ -1174,7 +1174,7 @@ static int snd_ad1848_probe(struct snd_w } hardware = WSS_HW_AD1847; } else { - snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); + snd_wss_dout(chip, CS4231_LEFT_INPUT, 0xaa); r = snd_wss_in(chip, CS4231_LEFT_INPUT); /* L/RMGE always low on AT2320 */ if ((r | CS4231_ENABLE_MIC_GAIN) != 0xaa) { @@ -1199,7 +1199,7 @@ static int snd_ad1848_probe(struct snd_w r = snd_wss_in(chip, CS4231_MISC_INFO); /* set CS423x MODE 2 */ - snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + snd_wss_dout(chip, CS4231_MISC_INFO, CS4231_MODE2); for (i = 0; i < 16; i++) { if (snd_wss_in(chip, i) != snd_wss_in(chip, 16 + i)) { /* we have more than 16 registers: check ID */ @@ -1221,7 +1221,7 @@ static int snd_ad1848_probe(struct snd_w else chip->hardware = WSS_HW_AD1848; out_mode: - snd_wss_out(chip, CS4231_MISC_INFO, 0); + snd_wss_dout(chip, CS4231_MISC_INFO, 0); out: spin_unlock_irqrestore(&chip->reg_lock, flags); return err; ---------------------------------------------------------------------- Zobacz galerie - tak wygladaja wampiry! >> link http://link.interia.pl/f1eee

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[alsa-devel] [ALSA] remove unused #include <version.h>
by Huang Weiyi 25 Aug '08

25 Aug '08

The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/mips/au1x00.c sound/soc/at32/playpaq_wm8510.c sound/soc/at91/eti_b1_wm8731.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index ee0741f..fbef38a 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -38,7 +38,6 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <linux/slab.h> -#include <linux/version.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 3f32621..fd2b2f3 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -22,7 +22,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/errno.h> #include <linux/clk.h> diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index b081e83..b81d6b2 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -22,7 +22,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h>

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[alsa-devel] alsa-utils,alsa-plugins v1.0.18rc2
by Jaroslav Kysela 25 Aug '08

25 Aug '08

Hello all, I released alsa-utils and alsa-plugins v1.0.18rc2 packages. The changelog is available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.18rc1_v1.0.18rc2 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [ALSA PATCH] alsa-git merge request
by Jaroslav Kysela 25 Aug '08

25 Aug '08

Linus, please pull from: git pull git://git.alsa-project.org/alsa-kernel.git for-linus gitweb interface: http://git.alsa-project.org/?p=alsa-kernel.git;a=shortlog;h=for-linus The GNU patch is available at: ftp://ftp.alsa-project.org/pub/kernel-patches/alsa-git-for-linus-2008-07-14… The following files will be updated: Documentation/sound/alsa/ALSA-Configuration.txt | 17 +- .../sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 4 +- include/asm-mips/mach-au1x00/au1xxx_psc.h | 8 + include/sound/ad1843.h | 46 + include/sound/control.h | 3 - include/sound/core.h | 8 +- include/sound/cs4231-regs.h | 8 + include/sound/cs4231.h | 3 + include/sound/emu10k1.h | 1 + include/sound/seq_kernel.h | 2 +- include/sound/soc-dapm.h | 42 +- include/sound/soc.h | 175 ++- include/sound/uda1341.h | 2 - include/sound/version.h | 4 +- sound/Kconfig | 34 +- sound/aoa/Kconfig | 11 +- sound/aoa/codecs/Kconfig | 4 - sound/aoa/fabrics/Kconfig | 1 - sound/aoa/soundbus/Kconfig | 1 - sound/arm/Kconfig | 21 +- sound/arm/sa11xx-uda1341.c | 2 - sound/core/Kconfig | 29 +- sound/core/control.c | 7 +- sound/core/init.c | 67 +- sound/core/memalloc.c | 62 - sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_device.c | 6 +- sound/core/sound.c | 8 +- sound/core/timer.c | 6 +- sound/drivers/Kconfig | 91 +- sound/drivers/vx/vx_hwdep.c | 2 +- sound/i2c/cs8427.c | 6 +- sound/i2c/l3/uda1341.c | 2 - sound/isa/Kconfig | 61 +- sound/isa/cs423x/cs4231_lib.c | 118 ++- sound/isa/opti9xx/opti92x-ad1848.c | 1126 +------------- sound/isa/sb/Makefile | 2 - sound/isa/wavefront/wavefront_synth.c | 2 +- sound/mips/Kconfig | 27 +- sound/mips/Makefile | 4 + sound/mips/ad1843.c | 561 +++++++ sound/mips/hal2.c | 947 ++++++++++++ sound/mips/hal2.h | 245 +++ sound/mips/sgio2audio.c | 1006 ++++++++++++ sound/oss/Kconfig | 49 +- sound/oss/dmasound/dmasound_core.c | 7 +- sound/oss/dmasound/dmasound_paula.c | 2 +- sound/oss/dmasound/dmasound_q40.c | 2 +- sound/oss/msnd.c | 2 - sound/oss/msnd.h | 2 - sound/oss/msnd_classic.h | 2 - sound/oss/msnd_pinnacle.c | 5 - sound/oss/msnd_pinnacle.h | 2 - sound/parisc/Kconfig | 13 +- sound/pci/Kconfig | 104 +-- sound/pci/Makefile | 2 +- sound/pci/ac97/Makefile | 12 +- sound/pci/ac97/ac97_codec.c | 11 +- sound/pci/ac97/ac97_patch.c | 81 +- sound/pci/{ac97 => }/ak4531_codec.c | 34 +- sound/pci/au88x0/au88x0_game.c | 2 - sound/pci/azt3328.c | 1235 ++++++++++----- sound/pci/azt3328.h | 207 +++- sound/pci/ca0106/ca0106_main.c | 5 + sound/pci/emu10k1/emu10k1_main.c | 1 + sound/pci/emu10k1/emumixer.c | 13 +- sound/pci/emu10k1/memory.c | 69 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 306 +++- sound/pci/hda/hda_proc.c | 5 +- sound/pci/hda/patch_analog.c | 38 +- sound/pci/hda/patch_conexant.c | 33 +- sound/pci/hda/patch_realtek.c | 548 +++++++- sound/pci/hda/patch_sigmatel.c | 71 +- sound/pci/ice1712/envy24ht.h | 10 +- sound/pci/ice1712/ice1712.h | 2 + sound/pci/ice1712/ice1724.c | 213 ++- sound/pci/maestro3.c | 42 +- sound/pci/nm256/nm256.c | 4 +- sound/pci/oxygen/hifier.c | 33 +- sound/pci/oxygen/oxygen.c | 76 +- sound/pci/oxygen/oxygen.h | 14 + sound/pci/oxygen/oxygen_io.c | 22 +- sound/pci/oxygen/oxygen_lib.c | 106 ++- sound/pci/oxygen/oxygen_pcm.c | 53 +- sound/pci/oxygen/virtuoso.c | 252 ++-- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/pcxhr/pcxhr_core.c | 18 +- sound/pci/trident/trident_main.c | 5 +- sound/pci/trident/trident_memory.c | 178 --- sound/pci/via82xx.c | 6 + sound/pci/ymfpci/ymfpci_main.c | 2 + sound/pcmcia/Kconfig | 15 +- sound/pcmcia/vx/vxp_ops.c | 2 +- sound/ppc/Kconfig | 26 +- sound/ppc/daca.c | 2 - sound/ppc/tumbler.c | 2 - sound/sh/Kconfig | 16 +- sound/soc/Kconfig | 19 +- sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 + sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 ++++++ sound/soc/at32/at32-pcm.h | 79 + sound/soc/at32/at32-ssc.c | 849 ++++++++++ sound/soc/at32/at32-ssc.h | 59 + sound/soc/at32/playpaq_wm8510.c | 522 +++++++ sound/soc/at91/Kconfig | 2 +- sound/soc/at91/at91-pcm.c | 6 +- sound/soc/at91/at91-ssc.c | 12 +- sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 53 +- sound/soc/au1x/Kconfig | 32 + sound/soc/au1x/Makefile | 13 + sound/soc/au1x/dbdma2.c | 421 +++++ sound/soc/au1x/psc-ac97.c | 387 +++++ sound/soc/au1x/psc-i2s.c | 414 +++++ sound/soc/au1x/psc.h | 53 + sound/soc/au1x/sample-ac97.c | 144 ++ sound/soc/codecs/Kconfig | 22 +- sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ac97.c | 31 +- sound/soc/codecs/ac97.h | 2 +- sound/soc/codecs/ak4535.c | 696 +++++++++ sound/soc/codecs/ak4535.h | 46 + sound/soc/codecs/cs4270.c | 8 +- sound/soc/codecs/cs4270.h | 2 +- sound/soc/codecs/tlv320aic3x.c | 384 +++-- sound/soc/codecs/tlv320aic3x.h | 55 +- sound/soc/codecs/uda1380.c | 852 ++++++++++ sound/soc/codecs/uda1380.h | 89 ++ sound/soc/codecs/wm8510.c | 817 ++++++++++ sound/soc/codecs/wm8510.h | 103 ++ sound/soc/codecs/wm8731.c | 79 +- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8750.c | 87 +- sound/soc/codecs/wm8750.h | 2 +- sound/soc/codecs/wm8753.c | 183 ++-- sound/soc/codecs/wm8753.h | 2 +- sound/soc/codecs/wm8990.c | 1626 ++++++++++++++++++++ sound/soc/codecs/wm8990.h | 832 ++++++++++ sound/soc/codecs/wm9712.c | 53 +- sound/soc/codecs/wm9712.h | 2 +- sound/soc/codecs/wm9713.c | 79 +- sound/soc/codecs/wm9713.h | 2 +- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-evm.c | 40 +- sound/soc/davinci/davinci-i2s.c | 16 +- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/fsl/Kconfig | 6 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_dma.h | 2 +- sound/soc/fsl/fsl_ssi.c | 24 +- sound/soc/fsl/fsl_ssi.h | 4 +- sound/soc/fsl/mpc8610_hpcd.c | 72 +- sound/soc/omap/Kconfig | 4 - sound/soc/omap/n810.c | 106 +- sound/soc/omap/omap-mcbsp.c | 16 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/Kconfig | 11 +- sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/corgi.c | 70 +- sound/soc/pxa/em-x270.c | 102 ++ sound/soc/pxa/poodle.c | 50 +- sound/soc/pxa/pxa2xx-ac97.c | 18 +- sound/soc/pxa/pxa2xx-ac97.h | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 17 +- sound/soc/pxa/pxa2xx-i2s.h | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 91 +- sound/soc/pxa/tosa.c | 47 +- sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/s3c24xx/neo1973_wm8753.c | 237 ++-- sound/soc/s3c24xx/s3c2412-i2s.c | 15 +- sound/soc/s3c24xx/s3c2412-i2s.h | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 15 +- sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 25 +- sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 6 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 3 - sound/soc/sh/Kconfig | 5 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/sh7760-ac97.c | 4 +- sound/soc/sh/ssi.c | 8 +- sound/soc/soc-core.c | 443 ++++-- sound/soc/soc-dapm.c | 344 ++++- sound/sparc/Kconfig | 17 +- sound/sparc/dbri.c | 2 +- sound/spi/Kconfig | 13 +- sound/usb/Kconfig | 16 +- sound/usb/caiaq/caiaq-audio.c | 1 + sound/usb/caiaq/caiaq-device.c | 12 +- sound/usb/caiaq/caiaq-device.h | 1 + sound/usb/usbaudio.c | 4 - sound/usb/usbquirks.h | 38 + 201 files changed, 16473 insertions(+), 3893 deletions(-) rename sound/pci/{ac97 => }/ak4531_codec.c (96%) Adrian Bunk (6): [ALSA] remove CVS keywords sound: sound/oss/: remove CVS keywords [ALSA] remove SND_GUS_SYNTH [ALSA] usbaudio.c: remove #ifndef CONFIG_USB_EHCI_SPLIT_ISO code ALSA: make sparc/dbri.c:snd_dbri_proc() static sound: sound/oss/dmasound/: cleanups Akio Idehara (1): [ALSA] hda - Add Toshiba dynabook SS RX1 support Andreas Mohr (2): [ALSA] PCI168 snd-azt3328 Linux driver: another huge update ALSA: PCI168 snd-azt3328: some more fixups Andy Green (1): ALSA: ASoC: Don't block system resume Chris Mennie (1): [ALSA] usb-audio - Support for Roland SonicCell sound module Clemens Ladisch (12): [ALSA] oxygen: fix version in MODULE_LICENSE [ALSA] oxygen: add symbol for I/O space size [ALSA] oxygen: save register writes [ALSA] oxygen: simplify DAC volume initialization [ALSA] oxygen: separate out hardware initialization code [ALSA] virtuoso: add xonar_enable_output() [ALSA] oxygen: add PM support [ALSA] oxygen: add symbols for buffer/period size constraints [ALSA] virtuoso: restrict period time to less than 10 s [ALSA] ice1724: fix MIDI ALSA: usb-audio: fix Yamaha KX quirk ALSA: usb-audio: add some Yamaha USB MIDI quirks Daniel Jacobowitz (1): [ALSA] ac97 - Add virtual master control to VT1616/VT1617A codec. Daniel Mack (3): [ALSA] soc - tlv320aic3x - revisit clock setup [ALSA] soc - tlv320aic3x - add GPIO support [ALSA] snd_usb_caiaq: add support for 'Session I/O' interface David Howells (2): ALSA: Fix a const pointer usage warning in the Digigram VX soundcard driver ALSA: Fix a const to non-const assignment in the Digigram VXpocket sound driver Dmitry Baryshkov (2): ALSA: tosa: fix compilation with new DAPM API ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration Geoffrey Wossum (1): [ALSA] Revised AT32 ASoC Patch Graeme Gregory (1): [ALSA] soc - DAPM - add hook to read state of DAPM widget Harvey Harrison (2): [ALSA] i2c: cs8427.c use put_unaligned helper alsa: add annotations to bitwise type snd_pcm_hw_param_t Jarkko Nikula (8): [ALSA] ASoC: Convert N810 machine driver to use gpiolib ALSA: ASoC: Cover also Nokia N810 WiMAX Edition in N810 machine driver ALSA: ASoC: Add support for generic DAPM register modifier widget ALSA: ASoC: TLV320AIC3X: Use register modifier widget for mic bias ALSA: ASoC: TLV320AIC3X: Modify only interface related bits in aic3x_set_dai_fmt ALSA: ASoC: TLV320AIC3X: Add support for digital microphone input ALSA: ASoC: Add digital mic configuration to N810 machine driver ALSA: ASoC: TLV320AIC3X: Add mixer control for ADC highpass filter Jaroslav Kysela (8): ALSA: Release v1.0.17rc1 ALSA: Release 1.0.17rc2 ALSA: Remove duplicate MODULE_AUTHOR/DESCRIPTION/LICENCE from snd-ens1370.ko ALSA: emu10k1 - fix possible memory leak in memory allocation routines [ALSA] Revert "alsa: add annotations to bitwise type snd_pcm_hw_param_t" ALSA: Release v1.0.17rc3 ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model ALSA: Release v1.0.17 Jiang zhe (1): [ALSA] hda - support intel DG33 motherboards Johannes Berg (1): ALSA: remove CONFIG_KMOD from sound Juergen Beisert (1): [ALSA] ac97: add support for wm9711 master left inv switch Kailang Yang (6): [ALSA] hda - Add support of Teradici controller [ALSA] hda - Add ICH9 controller support (8086:2911) [ALSA] hda - Add ALC663 support [ALSA] hda - Fix vref pincap check in alc882 auto-detection [ALSA] hda - show correct codec chip in PCM stream names [ALSA] hda - Fix EAPD and COEF setups for realtek codecs Krzysztof Helt (3): [ALSA] opti93x: fix sound ouput for Opti930 ALSA: opti93x: add support for Opti93x codec in cs4231-lib ALSA: opti93x: use cs4231 lib Liam Girdwood (16): ALSA: asoc: core - refactored DAPM pin control API. ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: omap - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: pxa - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: sh - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. ALSA: asoc: machines - add Digital Audio Interface (DAI) control functions. ALSA: asoc: n810 - fix build error. ALSA: asoc: at32 - DAI struct merge and enable_pin() change. ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform. Manuel Lauss (2): ALSA: ASoC: AC97 codec PM ALSA: ASoC: Au12x0/Au1550 PSC Audio support Mark Brown (30): [ALSA] soc - DAPM - Add bulk control registration [ALSA] soc - DAPM - Bulk route registration [ALSA] soc - tlv320aic3x - Convert to use bulk registration APIs [ALSA] soc - Zaurus - Convert to bulk DAPM registration APIs [ALSA] soc - eti_b1_wm8731 - Convert to use bulk DAPM control registration [ALSA] soc - neo1973_wm8753 - Convert to bulk DAPM registration APIs [ALSA] soc - davinci-evm - Update for bulk DAPM registration APIs [ALSA] soc - n810 - Update for bulk DAPM registration APIs [ALSA] ASoC: Remove in-code changelogs [ALSA] ASoC: Make CPU and codec DAI operations have same type [ALSA] ASoC: Clarify API for bias configuration [ALSA] ASoC: core checkpatch cleanups [ALSA] soc - Convert Wolfson codec drivers to use bulk DAPM registration [ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type [ALSA] ASoC: Add WM8510 driver [ALSA] ASoC: Add WM8990 driver [ALSA] ASoC: Fix default mono mixer configuration for WM8510 ALSA: ASoC: Fix register cache sizes for Wolfson codecs ALSA: ASoC: Fix register cache size for UDA1380 ALSA: ASoC: Tweak tlv320aicx reg_cache_size ALSA: ASoC: Add missing includes ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove ALSA: ASoC: Fix WM9713 voice PCM slave mode configuration ALSA: ASoC: Check for exact register match in wm97xx_reset() ALSA: ASoC: Replace custom debug macros with pr_ equivalents ALSA: ASoC: Advertise 16000Hz rate for WM9713 PCM interface ALSA: ASoC: Make pop/click debug wait times dynamically configurable ALSA: ASoC: Fix warning from strict_strtoul() ALSA: ASoC: Switch DAPM to use of standard DEBUG macro ALSA: ASoC: Dump DAPM state for non-stream changes Matthew Ranostay (3): ALSA: hda: Add support for 92HD73xxx codecs ALSA: hda - removed redundant gpio_mask ALSA: hda: 92hd71bxx PC Beep Mike Montour (3): ALSA: ASoC: Add TLV information to remaining WM8753 controls ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type ALSA: ASoC: Add TLV information to the LM4857 controls on the GTA01 Mike Rapoport (2): [ALSA] Add EM-X270 ASoC driver ALSA: ac97 - fix patch_ucb1400 for proper resume Milton Miller (1): ALSA: correct kcalloc usage Philipp Zabel (1): [ALSA] ASoC: Add UDA1380 driver Pierre Ossman (1): ALSA: trident - pause s/pdif output Randy Dunlap (1): ALSA: ASoC: fix PM=n build Richard Chan (1): ALSA: Add Yamaha KX49 (USB MIDI controller) to usbquirks.h Richard Purdie (1): ALSA: ASoC: Add AK4535 driver Stephen Rothwell (1): [ALSA] sound: fix export symbol typo Takashi Iwai (44): [ALSA] Remove unneeded ugly hack for i386 in memalloc.c [ALSA] hda - Fix DMA position inaccuracy sound: Convert to menuconfig [ALSA] Fix AC97 kconfig items [ALSA] Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE [ALSA] Improve the slots option handling [ALSA] Clean up sound/pci/ac97/Makefile [ALSA] Make ak4531 local to ens1370 driver [ALSA] trident - clean up obsolete synth codes sound: Clean up sound/oss/Kconfig [ALSA] trivial clean up of sound/isa/sb/Makefile [ALSA] emu10k1 - simplify page allocation for synth [ALSA] emu10k1 - Fix inverted Analog/Digital mixer switch on Audigy2 [ALSA] hda - Fix PLL gating control on Realtek codecs [ALSA] hda - increase max_codecs of ICH to 4 [ALSA] ac97 - Fix power_save option value as time-out ALSA: emu10k1 - Fix page allocation with GFP_DMA ALSA: hda - Add bdl_pos_adj option ALSA: hda - remove position_fix=3 ALSA: hda - bdl_pos_adj option to each instance ALSA: hda - Fix bdl_pos_adj value for ATI SB chipsets ALSA: hda - Add description of bdl_pos_adj option ALSA: hda - Add a warning if pending IRQ is found ALSA: hda - Fix stac9205_cfg_tbl ALSA: hda - Remove unused mutex ALSA: make snd_ctl_elem_read() and snd_ctl_elem_write() static ALSA: hda - bdl_pos_adj=32 as default ALSA: hda - use upper_32_bits() ALSA: emu10k1 - simplify the last fix ALSA: hda - Fix wrong volumes in AD1988 auto-probe mode ALSA: ca0106 - Add entry for another MSI K8N Diamond MB ALSA: hda - Fix digital converter proc output ALSA: hda - Add MacBook 3.1 support ALSA: ymfpci - fix initial volume for 44.1kHz output ALSA: hda - disable amp override on non-HP machines ALSA: Fix AC97 power down ALSA: ALSA: hda - Fix ALC883 medion model ALSA: hda - Add missing Thinkpad Z60m support ALSA: ALSA: hda - Fix speaker output on Toshiba P105 ALSA: hda - Add support for Lenovo 3000 N200 ALSA: wavefront - add const ALSA: hda - Fix internal mic vref pin setup ALSA: hda - Fix missing init for unsol events on micsense model ALSA: hda - Fix FSC V5505 model Thomas Bogendoerfer (2): ALSA: ALSA driver for SGI HAL2 audio device ALSA: ALSA driver for SGI O2 audio board Tim Niemeyer (1): [ALSA] soc - Patch to add debug messages to the neo1973_wm8753 (GTA01) sound driver Timur Tabi (1): ALSA: Fix register programming in Freescale MPC8610 HPCD sound driver Travis Place (2): ALSA: hda - Added model selection for iMac 24" ALSA: hda - Added SSID for 'Fujitsu Siemens Amilo M1451G' laptop Ville Syrjälä (1): [ALSA] maestro3: Fix hw volume on HP OmniBook Walter Sheets (1): ALSA: via82xx - Add VIA audio device #1841 to ac97_quirk list Werner Almesberger (1): [ALSA] soc - Fix s3c24xx-i2s LR sync while timer ticks are disabled ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

6 12

[alsa-devel] repost - snd_via82xx input channels problem
by Vaclav Peroutka 24 Aug '08

24 Aug '08

thank you for your answer. Finally I found, that backporting is not so necessary for me as I needed just three drivers - bt87x, via82xx and ice1724. Those drivers do not use newer functions I mentioned before, so the compilation has successfully ended. But I noticed that even with the newset driver the alc655 line-in inputs do not work well. Before ( alsa 1.0.9 ) I had the right channel always connected to the Mic input. Now (Alsa 2008-08-22 ) in the right channel there is a copy of the left channel. Thus I have monophonic input instead of stereo. I just downloaded the datasheet of ALC655 codec. I will read it hopefully tomorrow. But can you just advice me where to look to try to fix the problem ? It should be somewhere in snd_via82xx module ? Or somewhere else ? Inputs in snd_ice1724 (my second sound card) work properly. But I have never worked with kernel modules or kernel programming. I am just interested how to fix the problem I have. Thank you in advance, Vaclav

1 0

[alsa-devel] [PATCH 002/002] snd-powermac: HP detection for 1st iMac G3 SL
by Risto Suominen 23 Aug '08

23 Aug '08

1 0

[PATCH] ca0106 playback 44100Hz support to SPDIF and playback format & rate constraints
by Ben Stanley 23 Aug '08

23 Aug '08

Squashed commit of the following: commit 8bfbd000b4b602b517b5ab1edc5610ef39dced1e Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sat Aug 23 23:19:57 2008 +1000 Correct style and log messages according to Takashi's review comments. commit eb973ebecfac64b6f0e60df90eef8885ccc64aaf Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:56:15 2008 +1000 Annotate bug in comments. commit df725ab6af87e1c94d1d41e31e3865437b1999e0 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:44:12 2008 +1000 Checkpatch.pl round 4. commit c277bfa1773595b98c5447a7a085f21ccead110f Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:39:39 2008 +1000 Checkpatch.pl changes round 3. commit a3f9b1faae250aa0786ef68c3c53c3915e42972b Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Wed Aug 20 23:18:36 2008 +1000 Second round of checkpatch.pl changes. commit 3f45e4895fc8419a4a90eddf29af4f8547aa105d Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sun Aug 17 00:06:47 2008 +1000 Satisfy checkpatch.pl pass 1 commit 88424052a25359aa7ffa840e0515a7a93278f931 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sat Aug 16 22:32:25 2008 +1000 44100Hz playback support to SPDIF and playback format & rate constraints commit e0795c3bc756f07986eaae7c4ab54a78d3fecc80 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Sat Aug 16 22:25:38 2008 +1000 Remove use of snd_assert. commit 097b03e9ff07432c45a9865bd02208b20b09beb3 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Thu Jul 24 02:59:38 2008 +1000 * fix cosmetic bug commit 75769f154ea53b4215de37d314351c4e45bd2543 Author: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Thu Jul 3 20:47:31 2008 +1000 44100Hz playback support to SPDIF and playback format & rate constraints --- pci/ca0106/ca0106.h | 7 +- pci/ca0106/ca0106_main.c | 265 ++++++++++++++++++++++++++++++++++++++-------- 2 files changed, 228 insertions(+), 44 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..376fa9c 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.22 + * Version: 0.0.23 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -49,6 +49,8 @@ * Implement support for Line-in capture on SB Live 24bit. * 0.0.22 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.23 + * Add support for playback sampling rate and format constraints. * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -644,6 +646,8 @@ #include "ca_midi.h" +#define DRVNAME "snd-ca0106" + struct snd_ca0106; struct snd_ca0106_channel { @@ -659,6 +663,7 @@ struct snd_ca0106_pcm { struct snd_pcm_substream *substream; int channel_id; unsigned short running; + unsigned short hw_reserved; }; struct snd_ca0106_details { diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 2f8b28a..b06f7a9 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.25 + * Version: 0.0.26 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -83,9 +83,14 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * 0.0.25 * Powerdown SPI DAC channels when not in use + * 0.0.26 Ben Stanley + * Added support for output at 44100Hz rate (SPDIF only). + * Implemented constraints system for output rate and format. * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. + * Some programs fail to produce sound output (tested on SPDIF). See + * http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 * -- * * TODO: @@ -145,6 +150,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/ac97_codec.h> #include <sound/info.h> @@ -284,9 +290,9 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000), - .rate_min = 48000, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, @@ -318,6 +324,110 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; +static unsigned int all_spdif_playback_rates[] = + {44100, 48000, 96000, 192000}; + +static int hw_rule_playback_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int chi, any_44100 = 0, any_non_44100 = 0, mask; + struct snd_ca0106_channel *chp; + struct snd_pcm_runtime *runtime; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (chip->spdif_enable) { + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printdd("snd_hw_rule_playback_rate: ch=%d, " + "rate=%d.\n", chi, runtime->rate); + any_44100 += runtime->rate == 44100; + any_non_44100 += runtime->rate != 44100; + } + if (snd_BUG_ON(any_44100 && any_non_44100)) + return -EINVAL; + /* Compute the mask applied to all_spdif_playback_rates */ + if (any_44100) + mask = 0x1; + else if (any_non_44100) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC (FIXME Why?) */ + mask = 0xE; + } + snd_printdd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + any_44100, any_non_44100, mask, chip->spdif_enable); + return snd_interval_list(hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE), + ARRAY_SIZE(all_spdif_playback_rates), + all_spdif_playback_rates, mask); +} + +static int hw_rule_playback_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int chi, any_S16 = 0, any_S32 = 0; + struct snd_ca0106_channel *chp; + struct snd_pcm_runtime *runtime; + struct snd_mask fmt, *f = hw_param_mask( + params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + for (chi = 0; chi < 4; ++chi) { + chp = &(chip->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (!chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtime = chp->epcm->substream->runtime; + snd_printdd("snd_hw_rule_playback_format: ch=%d, format=%d.\n", + chi, runtime->format); + any_S16 += runtime->format == SNDRV_PCM_FORMAT_S16_LE; + any_S32 += runtime->format == SNDRV_PCM_FORMAT_S32_LE; + } + if (snd_BUG_ON(any_S16 && any_S32)) + return -EINVAL; + if (any_S16) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (any_S32) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + else { + /* No format yet chosen, so both formats are available. */ + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + } + result = snd_mask_refine(f, &fmt); + snd_printdd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + any_S16, any_S32, f->bits[0], fmt.bits[0], result); + return result; +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -508,10 +618,24 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + err = snd_pcm_hw_constraint_integer( + runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + err = snd_pcm_hw_constraint_step( + runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_playback_rate, (void *)chip, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_playback_format, (void *)chip, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -646,6 +770,9 @@ static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream /* hw_free callback */ static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + epcm->hw_reserved = 0; return snd_pcm_lib_free_pages(substream); } @@ -667,53 +794,103 @@ static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime, *runtimei; struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; + struct snd_ca0106_channel *chp; + int channel = epcm->channel_id, chi, any_44100 = 0, any_non_44100 = 0; u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; u32 hcfg_set = 0x00000000; u32 hcfg; - u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_mask = 0xFF0000; u32 reg40_set = 0; u32 reg40; - /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask; + u32 reg71_shift; u32 reg71_set = 0; u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); - /* Rate can be set per channel. */ - /* reg40 control host to fifo */ - /* reg71 controls DAC rate. */ - switch (runtime->rate) { - case 44100: - reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; - break; - case 48000: - reg40_set = 0; - reg71_set = 0; - break; - case 96000: - reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; - break; - case 192000: - reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; - break; - default: - reg40_set = 0; - reg71_set = 0; - break; + epcm->hw_reserved = 1; + /* FIXME CLEAN UP IF spdif_enable IS CHANGED WHILE CHANNELS ARE OPENED + * OR PREVENT THIS FROM HAPPENING. */ + if (emu->spdif_enable) + reg71_shift = 24; /* SPDIF Output Rate */ + else + reg71_shift = 16; /* I2S Output Rate */ + reg71_mask = 0x3 << reg71_shift; + + /*printk(KERN_DEBUG DRVNAME ": prepare_playback: " + "channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld,period_size=%ld, periods=%u, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1));*/ + /*printk("dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr,runtime->dma_area, table_base);*/ + /*printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr,emu->buffer.area, emu->buffer.bytes);*/ + /* We are forced to build the settings for all the channels. */ + for (chi = 0; chi < 4; ++chi) { + chp = &(emu->playback_channels[chi]); + if (!chp->use) + continue; + if (snd_BUG_ON(!chp->epcm)) + return -EINVAL; + if (chi != channel && !chp->epcm->hw_reserved) + continue; + if (snd_BUG_ON(!chp->epcm->substream)) + return -EINVAL; + if (snd_BUG_ON(!chp->epcm->substream->runtime)) + return -EINVAL; + runtimei = chp->epcm->substream->runtime; + any_44100 += runtimei->rate == 44100; + any_non_44100 += runtimei->rate != 44100; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* We only support 44100 to spdif, not to DAC. + (FIXME WHY?)*/ + if (emu->spdif_enable) { + /* When using 44100, *all* channels + must be set to that rate. */ + reg40_set |= 0x550000; + reg71_set |= 0x1 << reg71_shift; + break; + } else { + printk(KERN_ERR DRVNAME + "prepare_playback: " + "44100Hz is invalid for DAC.\n"); + } + case 48000: + /* reg40_set &= !(0x1 << (chi<<1)); */ + /* reg71_set &= !(0x1 << reg71_shift); */ + break; + case 96000: + reg40_set |= 0x20000 << (chi<<1); + reg71_set |= 0x2 << reg71_shift; + break; + case 192000: + reg40_set |= 0x30000 << (chi<<1); + reg71_set |= 0x3 << reg71_shift; + break; + default: + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %d.\n", + runtimei->rate); + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + snd_printdd(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + any_44100, any_non_44100, emu->spdif_enable); + /* Format is a global setting. */ + /* Only the first channel accessed can set this + (enforced by constraints). */ switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: hcfg_set = 0; @@ -1363,8 +1540,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); @@ -1378,7 +1555,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } chip->details = c; if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + printk(KERN_INFO DRVNAME + ": Sound card name=%s, subsystem=0x%x. " + "Forced to subsystem=0x%x\n", c->name, chip->serial, subsystem[dev]); } -- 1.5.4.3 --=-lsMZJ4IxY7skpnVaSzUT--

1 0

[alsa-devel] [PATCH] ALSA: ASoC V2: report error if DMA doesn't start in FSL MPC8610 sound drivers
by Timur Tabi 23 Aug '08

23 Aug '08

Return an error if the DMA's current pointer is not within the bounds of the DMA buffer. This can happen if the DMA controller is not programmed correctly, or if the SSI doesn't communicate with the DMA controller correctly. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2. sound/soc/fsl/fsl_dma.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index cb1e5ae..b4ae486 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -689,6 +689,15 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "DMA(%u,%u) pointer is out of range, halting stream\n", + dma_private->dma_info->controller_id, + dma_private->dma_info->channel_id); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* -- 1.5.5

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