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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 04 Dec '09

04 Dec '09

The following changes since commit a649d1fcc9bd2299cb06b6594fabb429fa50f174: Daniel Mack (1): ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Very last minute but these would qualify as bug fixes during the release cycle so I feel comfortable merging them now. Manuel Lauss (2): ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path sound/soc/au1x/dbdma2.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-)

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[alsa-devel] What's in sound git tree for 2.6.33
by Takashi Iwai 04 Dec '09

04 Dec '09

Hi, here is the annual report of sound subsystem for 2.6.33. For 2.6.33, sound bits aren't crazy huge like 2.6.31 or 2.6.32. HD-audio stuff is getting stabilized, but still lots of changes. There are many small fixes here and there, including (not surprisingly) even ISA drivers. Large amounts of changes are found for the support of new VIA HD-audio codecs, Intel HDMI codecs, and a few new drivers and board-supports like Infrasonic Quartet support. In ALSA core side, there are a couple of important fixes. The all "PC Speaker" and "PC Beep" mixer elements are renamed to "Beep". This was a confusion for a long time, and hopefully more understandable now. And, a long-standing PCM mmap problem on PPC and MIPS non-coherent architectures is fixed -- or better to say, Oops is worked around. We are still looking for a saner solution, but it'll be likely an issue for 2.6.34. The biggest change in HD-audio is that we enabled MSI as default now. So far, there is no regression report by this. But, if you see any IRQ issues newly on the new kernel, please report. For ASoC, this has been another fairly quiet release, too. Aside from the addition of virtual mux support to DAPM and some further preparatory work for multi-CODEC cards the majority of changes have been driver updates, including: - New drivers for ADS117x, AK4671, TLV320DAC33, TPA6130A2, WM8711 and WM8727. - Substantial improvements to DMA performance and reliability on MPC5200 and DaVinci. - Capture support for the FSI port on SH. Due to cross tree merge issues some updates to the TWL4030 MFD code are also included, enabling sharing of the CODEC hardware by support for the VIBRA interface of the devices. The tree is at: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus The short log and diffstat are below. Takashi === Akinobu Mita (2): ALSA: ice1712: Use bitrev8 ALSA: sound: usbmidi: Use hweight16 Aleksey Kunitskiy (1): ALSA: ice1724 - Patch for suspend/resume for ESI Juli@ Alexey Fisher (1): ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT Anuj Aggarwal (3): ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal ASoC: Adding OMAP3517 / AM3517 EVM support in ASOC ASoC: Modifying Kconfig/Makefile for AM3517 EVM Barry Song (4): ASoC: Provide API for reordering channels ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 ASoC: Blackfin I2S: use dai state rather than local counter ASoC: move setting ac97 platformdata earlier than ac97 read/write Ben Dooks (1): ASoC: S3C: Remove <plat/audio.h> Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Clemens Ladisch (23): sound: oxygen: do not try to restore nonexistent EEPROM sound: oxygen: fix for PI7C9X110 compatibility sound: virtuoso: split virtuoso.c sound: virtuoso: fix Xonar Essence ST support sound: oxygen: better defaults for upmixing control sound: oxygen: add stereo upmixing to center/LFE channels sound: oxygen: more hardware documentation sound: oxygen: cache codec registers sound: virtuoso: add headphone impedance control sound: oxygen: allow custom MCLK rates sound: virtuoso: add PCM1796 oversampling control sound: oxygen: add digital filter control sound: oxygen: add high-pass filter control sound: oxygen: fix input monitor control names control: remove snd_konctrol_volatile::owner_pid field control: use reference-counted pid sound: rawmidi: fix opened substreams count sound: pcm: record a substream's owner process sound: rawmidi: record a substream's owner process sound: usb-audio: add Roland UA-1G support sound: usb-audio: allow switching altsetting on Roland USB MIDI devices sound: usb: make the USB MIDI module more independent sound: usxxx: cleanup chip field Dan Carpenter (1): ALSA: remove unnecessary null check Daniel J Blueman (1): mfd: twl4030: fix ELF section mismatch... Daniel Mack (3): ASoC: CS4270: export de-emphasis filter as ALSA control ALSA: ARM: add Raumfeld audio support ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API Daniel T Chen (2): ALSA: intel8x0: Mute External Amplifier by default for Gateway 4525GZ ALSA: hda: Fix max PCM level to 0 dB for Fujitsu-Siemens laptops using CX20549 (Venice) Eero Nurkkala (2): ASoC: remove io_mutex ASoC: refactor snd_soc_update_bits() Einar Rünkaru (1): ALSA: hda - Make Dell Vostro 1015n mic and speaker switching work Enric Balletbò i Serra (1): ASoC: Add support for IGEP v2 Graeme Gregory (3): ASoC: pxa-ssp increase max_channels to 8 ASoC: ADS117x ADC driver ASoC: omap-mcbsp - add support for upto 16 channels. Grant Likely (6): ASoC/mpc5200: Track DMA position by period number instead of bytes ASoC/mpc5200: get rid of the appl_ptr tracking nonsense ASoC/mpc5200: Improve printk debug output for trigger ASoC/mpc5200: add to_psc_dma_stream() helper ASoC/mpc5200: fix enable/disable of AC97 slots ASoC: mpc5200: remove duplicate identical IRQ handler Igor Grinberg (1): ASoC: finally enable support for eXeda and CM-X300 Janusz Krzysztofik (2): ASoC: Amstrad Delta minor cleanups ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1 Jari Vanhala (1): ASoC: TWL4030: Vibra motor stop fix when it is driven with audio Jarkko Nikula (1): ASoC: Pandora: Pass SRG input clock frequency to the OMAP McBSP DAI Jaroslav Kysela (10): ALSA: rename "PC Speaker" and "PC Beep" controls to "Beep" ALSA: rename "PC Speaker" controls to "Speaker" ALSA: hda - proc - show which I/O NID is associated to PCM device ALSA: hda_intel: Digital PC Beep - change behaviour for input layer ALSA: hda_intel: Digital PC Beep - delay input device unregistration ALSA: hda: beep - add missing cancel_delayed_work ALSA: hda - add beep_mode module parameter ALSA: hda - proc - introduce Control: lines to show mixer<->NID assignment ALSA: hda - proc - add support for dynamic controls to mixer<->NID mapping ALSA: hda - move snd_hda_pcm_type_name from hda_codec.h to hda_local.h Jassi (2): ASoC: S3C I2S LRCLK polarity option. ASoC: Return correct codec clock in s3c64xx-i2s Jassi Brar (7): ASoC: Fixed arguments passed to SMDK64xx set_pll ASoC: Rename s3c24xx_pcm prefix to s3c_dma ASoC: Rename 's3c24xx-pcm' driver to 's3c-dma' ARM: S3C64XX: Define PCM Controller base registers ARM: S3C64XX: Added platform data header ARM: S3C64XX: Defined PCM controller platform devices ASoC: Added the CPU driver for PCM controllers Joonyoung Shim (4): ASoC: Factor out I2C 8 bit address 8 bit data I/O ASoC: AK4671: add ak4671 codec driver ASoC: Add jack_status_check callback function for GPIO jacks ASoC: Rename controls with a / in wm_hubs Julia Lawall (1): ASoC: Move dereference after NULL test Kailang Yang (1): ALSA: hda - Add ALC661/259, ALC892/888VD support Krzysztof Helt (24): ALSA: sscape: add supoort for SPEA Media FX/Reveal SC-600 ALSA: sscape: convert to firmware loader framework ALSA: sscape: remove MIDI instances counting with limit ULONG_MAX ALSA: sscape: force AD1848 codec mode on old Soundscape ALSA: sscape: coding style fixes ALSA: wss: convert CS4231 mixer to dB scale ALSA: wss: reuse CS4231 controls for AD1848 ALSA: snd_dma_pointer workaround for chipsets with buggy DMA sound: remove OSS Ensoniq SoundScape driver ALSA: es18xx: remove snd_card pointer from snd_es18xx structure ALSA: es18xx: remove snd_audiodrive structure ALSA: cs4236: detect chip in one pass ALSA: es18xx: code improvements ALSA: cs4236: update control names ALSA: opti-miro: use variables directly in the probe function ALSA: cs4236: add dB scale for all volume controls ALSA: opti-miro: remove snd_card pointer from snd_miro structure ALSA: opti-miro: make miro.h header available outside the alsa directory ALSA: opti-miro: expose ACI mixer to outside drivers ALSA: opti-miro: fix OOPS if hardware is not detected media/radio: New driver for the radio FM module on Miro PCM20 sound card ALSA: opti-miro: separate comon probing code ALSA: opti-miro: add PnP detection ALSA: opti9xx: remove snd_opti9xx fields Kuninori Morimoto (4): ASoC: sh: FSI: Remove DMA support ASoC: sh: FSI: Add capture support sh: ms7724se: Add runtime PM support for FSI ASoC: sh: fsi: Add runtime PM support Logan Li (1): ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Lydia Wang (30): ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro ALSA: HDA VIA: Change get_codec_type argument to hda_codec type ALSA: HDA VIA: Add VT1708B-CE codec support. ALSA: HDA VIA: Limit VT1702 AA-Path max volume ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type ALSA: HDA VIA: Add low current mode for power saving. ALSA: HDA VIA: Remove unused argument of via_new_analog_input ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls ALSA: HDA VIA: Rewrite via_independent_hp_put ALSA: HDA VIA: Add smart5.1 function. ALSA: HDA VIA: When changing input source, update power state. ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. ALSA: HDA VIA: Refresh front playback mute in via_hp_automute. ALSA: HDA VIA: Add Jack detect feature for VT1708. ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function. ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls. ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls. ALSA: HDA VIA: Replace MIC_BOOST_VOLUME. ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb ALSA: HDA VIA: Add VT1718S support. ALSA: HDA VIA: Add VT1828S and VT2020 support. ALSA: HDA VIA: Add VT1716S support. ALSA: HDA VIA: Add VT2002P support. ALSA: HDA VIA: Add VT1812 support. ALSA: HDA VIA: rename vt1708_control_templates[]. ALSA: HDA VIA: Change PW4 connect select default to to MW0. ALSA: HDA VIA: comments: update copyright, changeset, etc. ALSA: HDA VIA: Only cosmetic changes ALSA: VIA HDA: Add support for VT1818S. Manuel Lauss (5): ASoC: au1x: psc-ac97: verify correct codec register was read ASoC: au1x: psc-ac97: reorganize timeouts ASoC: au1x: convert to platform drivers. ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path Marin Mitov (1): ASoC: Use DMA_BIT_MASK(32) instead of deprecated DMA_32BIT_MASK Mark Brown (26): ASoC: Update WM8711 to driver model registration method ASoC: Factor out WM8711 cache I/O ASoC: Add SPI support to WM8711 ASoC: WM8711 minor cleanups ASoC: Add TLV information to WM8711 ASoC: Add source argument to PLL configuration ASoC: Allow per-route connectedness checks for supplies ASoC: Add S3C64xx IIS CDCLK source selection ASoC: Convert soc-cache to use C99 style initialisers for the table ASoC: Clean up WM8974 PLL configuration ASoC: Factor out analogue platform data from WM8993 ASoC: Simplify code for DAPM widget updates ASoC: Push DAPM enumeration register change test out ASoC: Add virtual enumeration support for DAPM muxes ASoC: Minor fixups to tpa6130a2 driver ASoC: Remove snd_soc_suspend_device() ASoC: Add regulator support for WM8731 ASoC: Minor SMDK64xx WM8580 cleanups ASoC: Staticise wm8727 driver structure ASoC: Move sysfs and debugfs functions to head of soc-core.c ASoC: Factor out snd_soc_init_card() ASoC: Update ads117x to current APIs ASoC: Add bit clock rate calculator utility functions ASoC: Remove redundant snd_soc_dapm_new_widgets() calls ASoC: Add BCLK calculation utility for TDM mode too S3C64XX: Staticise platform data for PCM devices Miguel Aguilar (1): ASoC: Davinci: Add audio codec support for DM365 EVM Mike Arthur (1): ASoC: Add WM8711 CODEC driver Mike Rapoport (1): ASoC: OMAP: enable Overo driver for CM-T35 Neil Jones (1): ASoC: Add support for the WM8727 DAC. Nicolas Ferre (1): ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek Ondrej Zary (2): ALSA: tea575x-tuner: fix mute snd-fm801: autodetect SF64-PCR (tuner-only) card Pavel Hofman (7): ALSA: ak4114 - fix errors in output selector bits ALSA: ak4620 support, codec regs listed in proc ALSA: ak4113 support ALSA: ice1724 - adding GPIO routines for mask and direction ALSA: ice1724 - pro-rate-locking makes sense only for internal clock mode ALSA: ice1724 - Support for multiple external clock types ALSA: ice1724 - Infrasonic Quartet support Peter Ujfalusi (21): ASoC: add support for multiple cards/codecs in debugfs ASoC: Improve the debugfs hierarchy ASoC: TPA6130A2 amplifier driver ASoC: TPA6130A2: Make tpa6130a2_power as static ASoC: Codec driver for Texas Instruments tlv320dac33 codec ASoC: TWL4030: Only update the needed bits in *set_dai_sysclk ASoC: tlv320dac33: typo fix in the header MFD: twl4030: add twl4030_codec MFD as a new child to the core OMAP: Platform support for twl4030_codec MFD ASoC: TWL4030: use the twl4030-codec.h for register descriptions ASoC: TWL4030: Driver registration via twl4030_codec MFD ASoC: TWL4030: Remove bypass tracking ASoC: TWL4030: Change codec_muted to apll_enabled ASoC: TWL4030: Change APLL powering sequence ASoC: TWL4030: Add APLL supply for the capture path ASoC: TWL4030: Make sure, that the codec is powered on startup MFD: TWL4030: Add audio_mclk to the codec platform data OMAP: Configure audio_mclk for twl4030-codec MFD MFD: twl4030-codec: APLL_INFREQ handling in the MFD driver ASoC: TWL4030: Do not modify the APLL_CTL register ASoC: tlv320dac33: Change RT wq to singlethread wq Rafael Ignacio Zurita (1): ALSA: sh: add SuperH DAC audio driver for ALSA V4 Roel Kluin (3): ALSA: Cleanup redundant tests on unsigned sound: OSS: fix error return in dma_ioctl() sound: OSS: keep index within bounds of midi_devs[] Russell King (1): ALSA: AACI cleanup Stas Sergeev (1): ALSA: snd-pcsp: add nopcm mode Stephen Rothwell (1): sound: use semicolons to end statements Takashi Iwai (41): ALSA: hda - Enable MSI as default ASoC: Fix build errors of wm8711.c with SPI ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c ALSA: sscape - Remove sscap_ioctl.h from include/sound/Kbuild ALSA: hda - Clean up name string creation in patch_realtek.c ALSA: sscape - Remove invalid __devinitdata to module parameters ALSA: hda - Add a proper ifdef to a debug code ALSA: hda - Switch to polling mode before disabling MSI ALSA: hda - Reset pins of IDT/STAC codecs at free ASoC: Remove dead code and labels ALSA: hda - Add reboot notifier to each codec ALSA: hda - Add missing export for snd_hda_bus_reboot_notify ALSA: hda - Add power on/off counter ALSA: hda - Don't access invalid substream in proc file ALSA: hda - Fix build error without CONFIG_SND_HDA_HWDEP=y ALSA: hda - Get rid of magic digits for subdev hack ALSA: hda - Add description of beep_mode in ALSA-Configuration.txt ALSA: hda - Update / add kerneldoc comments to exported functions ALSA: hda - Fix beep_mode option value ALSA: hda - Fix build errors with CONFIG_SND_HDA_INPUT_BEEP=n ALSA: hda - Disable default quirk for Sony VAIO with ALC262 codec ALSA: hda - Replace the rest of jack-detections with snd_hda_jack_detect() ALSA: opti-miro: Fix missing semicolon ALSA: hda - Fix detection of dual headphones ALSA: hda - Fix mute-LED sync on HP laptops with IDT92HD83xxx codecs ALSA: hda - Change quirk for Acer Aspire 5930G ALSA: hda - Fix input and jack Kconfig depenencies Revert "ALSA: hda - Change quirk for Acer Aspire 5930G" ALSA: pcm - Use dma_mmap_coherent() if available ALSA: pcm - define snd_pcm_default_page_ops() ALSA: pcm - fix page conversion on non-coherent MIPS arch ALSA: pcm - fix page conversion on non-coherent PPC arch ALSA: Remove old DMA-mmap code from arm/devdma.c ALSA: hda - Don't trigger pin-sense for STAC/IDT codecs ASoC: Revert missing reset_err in wm97*.c ALSA: hda - Add a position_fix quirk for MSI Wind U115 ALSA: hda - Fix Cxt5047 test mode ALSA: hda - Add a pin-fix for FSC Amilo Pi1505 ALSA: hda - Add position_fix quirk for HP dv3 ALSA: aaci - Clean up duplicate code ALSA: hda - Fix memory leaks in the previous patch Timothy Knoll (1): sound: Kconfig typo fix Tobias Hansen (2): ALSA: snd-usb-us122l: add support for US-144 ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd Troy Kisky (5): ASoC: DaVinci: remove requirement that dma_params is 1st in structure ASoC: DaVinci: i2s, reduce underruns by combining into 1 element ASoC: DaVinci: pcm, rename variables in prep for ping/pong ASoC: DaVinci: pcm, fix underrun by using sram ASoC: DaVinci: use edma_pause, edma_resume Wu Fengguang (21): ALSA: hda - select IbexPeak handler for Calpella ALSA: hda - vectorize get_empty_pcm_device() ALSA: hda - allow up to 4 HDMI devices ALSA: hda - convert intelhdmi global references to local parameters ALSA: hda - remove intelhdmi dependency on multiout ALSA: hda - use pcm prepare/cleanup callbacks for intelhdmi ALSA: hda - reorder intelhdmi prepare/cleanup callbacks ALSA: hda - vectorize intelhdmi ALSA: hda - get intelhdmi max channels from widget caps ALSA: hda - auto parse intelhdmi cvt/pin configurations ALSA: hda - remove static intelhdmi configurations ALSA: intelhdmi - fix audio infoframe fill size ALSA: intelhdmi - fix channel mapping slot mask ALSA: intelhdmi - export monitor-presence and ELD-valid status ALSA: hda - introduce snd_hda_jack_detect() and snd_hda_pin_sense() ALSA: intelhdmi - probe for monitor/eld presence at module init time ALSA: intelhdmi - separate out infoframe checksum routine ALSA: intelhdmi - sticky infoframe ALSA: intelhdmi - sticky stream id and format ALSA: intelhdmi - sticky channel count ALSA: hda - show EPSS capability in proc Wu Zhangjin (1): ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency akpm(a)linux-foundation.org (1): sound/pci/hda/patch_via.c: work around gcc-4.0.2 ICE jassi brar (1): ASoC: Support WM8580 based audio subsystem on SMDK64xx machines Łukasz Wojniłowicz (1): ALSA: hda - 4930g mute lfe and side when pluging in headphones --- Documentation/sound/alsa/ALSA-Configuration.txt | 13 +- Documentation/sound/alsa/ControlNames.txt | 3 +- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + arch/arm/mach-davinci/include/mach/asp.h | 8 + arch/arm/mach-omap2/board-3430sdp.c | 10 + arch/arm/mach-omap2/board-omap3beagle.c | 10 + arch/arm/mach-omap2/board-omap3evm.c | 10 + arch/arm/mach-omap2/board-omap3pandora.c | 10 + arch/arm/mach-omap2/board-overo.c | 10 + arch/arm/mach-omap2/board-zoom2.c | 10 + arch/arm/mach-s3c6400/include/mach/map.h | 2 + arch/arm/plat-s3c/include/plat/audio.h | 48 +- arch/arm/plat-s3c/include/plat/devs.h | 3 + arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 2 + arch/arm/plat-s3c64xx/dev-audio.c | 101 +- arch/sh/boards/mach-hp6xx/setup.c | 55 + arch/sh/boards/mach-se/7724/setup.c | 3 + arch/sh/include/mach-common/mach/hp6xx.h | 4 + drivers/media/radio/Kconfig | 18 + drivers/media/radio/Makefile | 1 + drivers/media/radio/radio-miropcm20.c | 270 ++ drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 1 + drivers/mfd/twl4030-codec.c | 276 ++ drivers/mfd/twl4030-core.c | 16 +- include/linux/i2c/twl4030.h | 19 + include/linux/mfd/twl4030-codec.h | 272 ++ include/sound/Kbuild | 1 - sound/isa/opti9xx/miro.h => include/sound/aci.h | 23 +- include/sound/ak4113.h | 321 ++ include/sound/ak4114.h | 12 +- include/sound/ak4xxx-adda.h | 5 +- include/sound/control.h | 5 +- include/sound/cs4231-regs.h | 1 - include/sound/pcm.h | 3 + include/sound/rawmidi.h | 2 + include/sound/sh_dac_audio.h | 21 + include/sound/soc-dai.h | 14 +- include/sound/soc-dapm.h | 17 + include/sound/soc.h | 15 +- include/sound/sscape_ioctl.h | 21 - include/sound/tlv320dac33-plat.h | 20 + include/sound/tpa6130a2-plat.h | 30 + include/sound/wss.h | 1 - sound/Kconfig | 2 +- sound/arm/Makefile | 2 +- sound/arm/aaci.c | 35 +- sound/arm/devdma.c | 80 - sound/arm/devdma.h | 3 - sound/core/control.c | 9 +- sound/core/isadma.c | 10 +- sound/core/oss/mixer_oss.c | 4 +- sound/core/pcm.c | 6 +- sound/core/pcm_native.c | 83 +- sound/core/rawmidi.c | 17 +- sound/drivers/pcsp/pcsp.c | 32 +- sound/drivers/pcsp/pcsp.h | 2 +- sound/drivers/pcsp/pcsp_mixer.c | 35 +- sound/i2c/cs8427.c | 15 +- sound/i2c/other/Makefile | 3 +- sound/i2c/other/ak4113.c | 639 ++++ sound/i2c/other/ak4xxx-adda.c | 136 +- sound/i2c/other/tea575x-tuner.c | 2 +- sound/isa/Kconfig | 12 +- sound/isa/cmi8330.c | 4 +- sound/isa/cs423x/cs4236.c | 13 +- sound/isa/cs423x/cs4236_lib.c | 241 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 221 +- sound/isa/opti9xx/miro.c | 783 +++-- sound/isa/opti9xx/opti92x-ad1848.c | 110 +- sound/isa/sb/sb_mixer.c | 4 +- sound/isa/sscape.c | 727 ++--- sound/isa/wss/wss_lib.c | 105 +- sound/oss/Kconfig | 12 - sound/oss/Makefile | 1 - sound/oss/audio.c | 2 +- sound/oss/midi_synth.c | 2 +- sound/oss/mpu401.c | 2 +- sound/oss/sh_dac_audio.c | 3 - sound/oss/sscape.c | 1480 --------- sound/pci/Kconfig | 1 + sound/pci/ac97/ac97_codec.c | 6 +- sound/pci/ac97/ac97_patch.c | 12 +- sound/pci/azt3328.c | 4 +- sound/pci/ca0106/ca0106_mixer.c | 4 +- sound/pci/ca0106/ca0106_proc.c | 4 +- sound/pci/cmipci.c | 4 +- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +- sound/pci/emu10k1/emumixer.c | 4 +- sound/pci/emu10k1/emuproc.c | 4 +- sound/pci/emu10k1/io.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/fm801.c | 40 +- sound/pci/hda/Kconfig | 13 +- sound/pci/hda/hda_beep.c | 114 +- sound/pci/hda/hda_beep.h | 10 + sound/pci/hda/hda_codec.c | 607 +++- sound/pci/hda/hda_codec.h | 11 + sound/pci/hda/hda_eld.c | 20 +- sound/pci/hda/hda_generic.c | 17 +- sound/pci/hda/hda_hwdep.c | 38 + sound/pci/hda/hda_intel.c | 50 +- sound/pci/hda/hda_local.h | 69 +- sound/pci/hda/hda_proc.c | 70 + sound/pci/hda/patch_analog.c | 61 +- sound/pci/hda/patch_ca0110.c | 4 +- sound/pci/hda/patch_cirrus.c | 31 +- sound/pci/hda/patch_cmedia.c | 4 +- sound/pci/hda/patch_conexant.c | 189 +- sound/pci/hda/patch_intelhdmi.c | 488 +++- sound/pci/hda/patch_realtek.c | 440 ++-- sound/pci/hda/patch_sigmatel.c | 149 +- sound/pci/hda/patch_via.c | 3509 +++++++++++++++++++-- sound/pci/ice1712/Makefile | 2 +- sound/pci/ice1712/ice1712.c | 12 + sound/pci/ice1712/ice1712.h | 14 +- sound/pci/ice1712/ice1724.c | 103 +- sound/pci/ice1712/juli.c | 56 +- sound/pci/ice1712/quartet.c | 1130 +++++++ sound/pci/ice1712/quartet.h | 10 + sound/pci/intel8x0.c | 6 + sound/pci/oxygen/Makefile | 3 +- sound/pci/oxygen/cs2000.h | 83 + sound/pci/oxygen/hifier.c | 61 +- sound/pci/oxygen/oxygen.c | 248 ++- sound/pci/oxygen/oxygen.h | 5 + sound/pci/oxygen/oxygen_lib.c | 29 +- sound/pci/oxygen/oxygen_mixer.c | 52 +- sound/pci/oxygen/oxygen_pcm.c | 19 +- sound/pci/oxygen/virtuoso.c | 1105 +------- sound/pci/oxygen/xonar.h | 50 + sound/pci/oxygen/xonar_cs43xx.c | 434 +++ sound/pci/oxygen/xonar_hdmi.c | 128 + sound/pci/oxygen/xonar_lib.c | 132 + sound/pci/oxygen/xonar_pcm179x.c | 1115 +++++++ sound/ppc/awacs.c | 12 +- sound/ppc/burgundy.c | 8 +- sound/ppc/tumbler.c | 2 +- sound/sh/Kconfig | 8 + sound/sh/Makefile | 2 + sound/sh/sh_dac_audio.c | 453 +++ sound/soc/Makefile | 2 +- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/au1x/dbdma2.c | 115 +- sound/soc/au1x/psc-ac97.c | 243 +- sound/soc/au1x/psc-i2s.c | 189 +- sound/soc/au1x/psc.h | 7 +- sound/soc/blackfin/bf5xx-ad1836.c | 7 + sound/soc/blackfin/bf5xx-ad1938.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 15 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 45 +- sound/soc/blackfin/bf5xx-tdm.h | 11 + sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 12 + sound/soc/codecs/ac97.c | 3 - sound/soc/codecs/ad1836.c | 12 - sound/soc/codecs/ad1938.c | 12 - sound/soc/codecs/ad1980.c | 5 - sound/soc/codecs/ad73311.c | 8 - sound/soc/codecs/ads117x.c | 123 + sound/soc/codecs/ads117x.h | 13 + sound/soc/codecs/ak4104.c | 8 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4642.c | 9 - sound/soc/codecs/ak4671.c | 815 +++++ sound/soc/codecs/ak4671.h | 156 + sound/soc/codecs/cs4270.c | 28 +- sound/soc/codecs/cx20442.c | 12 - sound/soc/codecs/pcm3008.c | 9 - sound/soc/codecs/ssm2602.c | 9 - sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 11 +- sound/soc/codecs/tlv320aic26.c | 11 - sound/soc/codecs/tlv320aic3x.c | 11 - sound/soc/codecs/tlv320dac33.c | 1229 ++++++++ sound/soc/codecs/tlv320dac33.h | 267 ++ sound/soc/codecs/tpa6130a2.c | 463 +++ sound/soc/codecs/tpa6130a2.h | 61 + sound/soc/codecs/twl4030.c | 452 ++-- sound/soc/codecs/twl4030.h | 242 +-- sound/soc/codecs/uda134x.c | 9 - sound/soc/codecs/uda1380.c | 9 - sound/soc/codecs/wm8350.c | 32 +- sound/soc/codecs/wm8400.c | 32 +- sound/soc/codecs/wm8510.c | 14 +- sound/soc/codecs/wm8523.c | 26 - sound/soc/codecs/wm8580.c | 30 +- sound/soc/codecs/wm8711.c | 633 ++++ sound/soc/codecs/wm8711.h | 42 + sound/soc/codecs/wm8727.c | 135 + sound/soc/codecs/wm8727.h | 21 + sound/soc/codecs/wm8728.c | 10 - sound/soc/codecs/wm8731.c | 94 +- sound/soc/codecs/wm8750.c | 9 - sound/soc/codecs/wm8753.c | 49 +- sound/soc/codecs/wm8776.c | 43 - sound/soc/codecs/wm8900.c | 34 +- sound/soc/codecs/wm8903.c | 28 - sound/soc/codecs/wm8940.c | 28 +- sound/soc/codecs/wm8960.c | 30 +- sound/soc/codecs/wm8961.c | 27 - sound/soc/codecs/wm8971.c | 11 +- sound/soc/codecs/wm8974.c | 36 +- sound/soc/codecs/wm8988.c | 44 - sound/soc/codecs/wm8990.c | 14 +- sound/soc/codecs/wm8993.c | 49 +- sound/soc/codecs/wm9081.c | 27 - sound/soc/codecs/wm9705.c | 7 - sound/soc/codecs/wm9712.c | 7 - sound/soc/codecs/wm9713.c | 32 +- sound/soc/codecs/wm_hubs.c | 51 +- sound/soc/codecs/wm_hubs.h | 5 + sound/soc/davinci/Kconfig | 4 +- sound/soc/davinci/davinci-evm.c | 7 +- sound/soc/davinci/davinci-i2s.c | 85 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 5 - sound/soc/davinci/davinci-pcm.c | 571 +++- sound/soc/davinci/davinci-pcm.h | 2 + sound/soc/fsl/mpc5200_dma.c | 123 +- sound/soc/fsl/mpc5200_dma.h | 24 +- sound/soc/fsl/mpc5200_psc_ac97.c | 39 +- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/omap/Kconfig | 23 +- sound/soc/omap/Makefile | 4 + sound/soc/omap/am3517evm.c | 202 ++ sound/soc/omap/ams-delta.c | 4 +- sound/soc/omap/igep0020.c | 148 + sound/soc/omap/omap-mcbsp.c | 63 +- sound/soc/omap/omap3evm.c | 7 + sound/soc/omap/omap3pandora.c | 24 +- sound/soc/omap/overo.c | 4 +- sound/soc/pxa/Kconfig | 12 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 20 +- sound/soc/pxa/raumfeld.c | 335 ++ sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/Kconfig | 12 + sound/soc/s3c24xx/Makefile | 6 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 10 +- sound/soc/s3c24xx/neo1973_wm8753.c | 11 +- sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} | 88 +- sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} | 8 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 35 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +- sound/soc/s3c24xx/s3c-pcm.c | 552 ++++ sound/soc/s3c24xx/s3c-pcm.h | 123 + sound/soc/s3c24xx/s3c2412-i2s.c | 7 +- sound/soc/s3c24xx/s3c2443-ac97.c | 13 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 26 +- sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 268 ++ sound/soc/s6000/s6000-pcm.c | 4 +- sound/soc/sh/Kconfig | 1 - sound/soc/sh/fsi.c | 271 +- sound/soc/soc-cache.c | 46 +- sound/soc/soc-core.c | 566 ++-- sound/soc/soc-dapm.c | 135 +- sound/soc/soc-jack.c | 6 +- sound/soc/soc-utils.c | 74 + sound/usb/usbaudio.c | 38 +- sound/usb/usbaudio.h | 7 +- sound/usb/usbmidi.c | 208 +- sound/usb/usbmixer_maps.c | 23 + sound/usb/usbquirks.h | 23 + sound/usb/usx2y/us122l.c | 135 +- sound/usb/usx2y/us122l.h | 4 +- sound/usb/usx2y/usX2Yhwdep.c | 8 +- sound/usb/usx2y/usbusx2y.c | 28 +- sound/usb/usx2y/usbusx2y.h | 6 +- sound/usb/usx2y/usbusx2yaudio.c | 34 +- sound/usb/usx2y/usx2yhwdeppcm.c | 8 +- 285 files changed, 21108 insertions(+), 7537 deletions(-) create mode 100644 drivers/media/radio/radio-miropcm20.c create mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 include/linux/mfd/twl4030-codec.h rename sound/isa/opti9xx/miro.h => include/sound/aci.h (85%) create mode 100644 include/sound/ak4113.h create mode 100644 include/sound/sh_dac_audio.h delete mode 100644 include/sound/sscape_ioctl.h create mode 100644 include/sound/tlv320dac33-plat.h create mode 100644 include/sound/tpa6130a2-plat.h delete mode 100644 sound/arm/devdma.c delete mode 100644 sound/arm/devdma.h create mode 100644 sound/i2c/other/ak4113.c delete mode 100644 sound/oss/sscape.c create mode 100644 sound/pci/ice1712/quartet.c create mode 100644 sound/pci/ice1712/quartet.h create mode 100644 sound/pci/oxygen/cs2000.h create mode 100644 sound/pci/oxygen/xonar.h create mode 100644 sound/pci/oxygen/xonar_cs43xx.c create mode 100644 sound/pci/oxygen/xonar_hdmi.c create mode 100644 sound/pci/oxygen/xonar_lib.c create mode 100644 sound/pci/oxygen/xonar_pcm179x.c create mode 100644 sound/sh/sh_dac_audio.c create mode 100644 sound/soc/codecs/ads117x.c create mode 100644 sound/soc/codecs/ads117x.h create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h create mode 100644 sound/soc/codecs/tlv320dac33.c create mode 100644 sound/soc/codecs/tlv320dac33.h create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h create mode 100644 sound/soc/codecs/wm8727.c create mode 100644 sound/soc/codecs/wm8727.h create mode 100644 sound/soc/omap/am3517evm.c create mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/pxa/raumfeld.c rename sound/soc/s3c24xx/{s3c24xx-pcm.c => s3c-dma.c} (82%) rename sound/soc/s3c24xx/{s3c24xx-pcm.h => s3c-dma.h} (87%) create mode 100644 sound/soc/s3c24xx/s3c-pcm.c create mode 100644 sound/soc/s3c24xx/s3c-pcm.h create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c create mode 100644 sound/soc/soc-utils.c

1 0

[alsa-devel] Underruns in basic audio playback application
by Devin Heitmueller 04 Dec '09

04 Dec '09

Hello all, I am working on a very basic application that reads a capture device and outputs to a playback device. It's what you would traditionally accomplish with something like the following: arecord -D hw:1,0 -r 48000 -c 2 -f S16_LE | aplay - However, in this case, since the final result will be part of a TV watching application, I need to maintain less than 30ms of latency to preserve lipsync. The application I ended up with essentially creates two PCM streams (one for capture, one for playback), sets them up with the same parameters in terms of rate, format, channels, etc., and then has a loop of snd_pcm_readi() and snd_pcm_writei() calls. The application works, except I am getting a considerable amount of underruns on the playback device. Given my relative inexperience with alsa-lib, I suspect that I have misconfigured one of the parameters effecting buffering - buffer size, buffer time, period size, period time. Can anyone offer any constructive suggestions or tips on what these tuning parameters should be set to for a continuous 48000Khz, 2 channel stream of audio? Also, should the buffering configuration really be the same for both the capture and playback stream, or do I need to play some games such that one requires more/less buffering than the other? Thanks in advance, Devin -- Devin J. Heitmueller - Kernel Labs http://www.kernellabs.com

2 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for regulator framework
by Ilkka Koskinen 04 Dec '09

04 Dec '09

Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++------ 1 files changed, 79 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 9c8903d..5037454 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -469,6 +499,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -476,12 +508,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -959,6 +998,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + return 0; pcm_err: @@ -1039,7 +1081,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1130,6 +1172,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1149,6 +1209,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1177,6 +1241,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- 1.6.0.4

3 2

[alsa-devel] ALSA Control Questions
by Tobias Schneider 04 Dec '09

04 Dec '09

I've got some questions concerning the ALSA control interface... 1. What is a "simple" control? I would like to use the control interface to set more than just volume (e.g. I would like to set input gain, output attenuation, input channel selection, output channel selection, output channel activation...), so I am wondering if I should use SNDRV_CTL_ELEM_IFACE_CARD or SNDRV_CTL_ELEM_IFACE_MIXER, as they are obviously in context with "simple" control... 2. The more interesting question: as written in "Writing an alsa driver" callbacks are basically not atomic, but I am getting BUG: scheduling while atomic: amixer/0x00000001/430 Call Trace: .[..] in "put" callback?? So what is meant with "basically not atomic"?? What is allowed and what is not allowed, am I able to get hardware interrupts in this context?

3 7

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for regulator framework
by Ilkka Koskinen 04 Dec '09

04 Dec '09

Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 92 ++++++++++++++++++++++++++++++++++------ 1 files changed, 79 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3ca8934..c2510bb 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -470,6 +500,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -477,12 +509,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -960,6 +999,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(codec->dev, "failed to register card\n"); @@ -1048,7 +1090,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1138,6 +1180,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1157,6 +1217,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1185,6 +1249,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- 1.6.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: DaVinci: Update suspend/resume support for McASP driver
by Chaithrika U S 04 Dec '09

04 Dec '09

Add clock enable and disable calls to resume and suspend respectively. Also add a member to the audio device data structure which tracks the clock status. Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied. [1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/ 2009-November/016958.html Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 18 ++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0a302e1..a613bbb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -767,14 +767,27 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int ret = 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + /* Fall through */ + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(dev, substream->stream); break; @@ -866,6 +879,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } clk_enable(dev->clk); + dev->clk_active = 1; dev->base = (void __iomem *)IO_ADDRESS(mem->start); dev->op_mode = pdata->op_mode; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 582c924..e755b51 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -44,6 +44,7 @@ struct davinci_audio_dev { int sample_rate; struct clk *clk; unsigned int codec_fmt; + u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ad4d7f4..80c7fdf 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -49,7 +49,7 @@ static void print_buf_info(int slot, char *name) static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | -- 1.5.6

2 1

[alsa-devel] snd_hda_intel: MacBook Pro 5,1 / ALC889A issues
by Alexey Spiridonov 04 Dec '09

04 Dec '09

Hello, The hda_intel driver is only marginally usable on my laptop. I described the problems in detail here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/488103 In brief, one has to choose between two sets of problems: 1) model=mbp3: broken speakers or 2) model=mb5: working speakers, but noisy headphone output Neither option provides headphone jack sense. Neither option is able to access the headset-type microphone available under MacOS through 3-band jacks like this: http://img.tomshardware.com/us/2007/07/19/how_good_is_apples_wired_iphone_h… (MacOS automatically switches from the in-computer mic to the the headset mic -- I haven't been able to get any signal from it despite trying all permutations of alsamixer's capture settings) I have attached the output of alsa-info.sh. I'd be happy to help test patches, or other debugging support. Thanks a lot! ~a

1 0

[alsa-devel] syslog says snd-usb-audio bug?
by Achilles H 04 Dec '09

04 Dec '09

Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Dec 4 00:16:21 susbwoy pulseaudio[1719]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I am using Ubuntu 9.10 64bit with an Edirol UA-25 USB sound card. I installed OpenArena, and tried to open it. Screen went black - Could only get back to the desktop by killing X. This is what happened in the terminal when OpenArena Died at Initializing sound ------ Initializing Sound ------ bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Allocated 96 sources. OpenAL default capture device is 'ALSA Capture on default' AL lib: alsa.c:675: set access failed: Invalid argument AL lib: oss.c:307: Could not open /dev/dsp: No such file or directory Killed I have no issues with sound at all apart from trying to use OpenAL.Warsow(another great game) will also refuse to run with OpenAL, except it runs great in the default sound mode 'qf'. What is going on here? thanks

1 0

[alsa-devel] [PATCH] ASoC: Rename controls with a / in wm_hubs
by Joonyoung Shim 04 Dec '09

04 Dec '09

This renames from a character / to : of controls. A / occurs below error messages. ASoC: Failed to create IN2RP/VXRP debugfs file ASoC: Failed to create IN2LP/VXRN debugfs file Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- If there is the better a delimiter or other idea, please inform me it. sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 810a563..d73c305 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1LN"), SND_SOC_DAPM_INPUT("IN1LP"), SND_SOC_DAPM_INPUT("IN2LN"), -SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN2LP:VXRN"), SND_SOC_DAPM_INPUT("IN1RN"), SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), -SND_SOC_DAPM_INPUT("IN2RP/VXRP"), +SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), @@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, - { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" }, { "IN2L PGA", "IN2LN Switch", "IN2LN" }, - { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" }, { "IN2R PGA", "IN2RN Switch", "IN2RN" }, - { "Direct Voice", NULL, "IN2LP/VXRN" }, - { "Direct Voice", NULL, "IN2RP/VXRP" }, + { "Direct Voice", NULL, "IN2LP:VXRN" }, + { "Direct Voice", NULL, "IN2RP:VXRP" }, { "MIXINL", "IN1L Switch", "IN1L PGA" }, { "MIXINL", "IN2L Switch", "IN2L PGA" }, @@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Left Output Mixer", "Right Input Switch", "MIXINR" }, { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, - { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" }, { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Right Output Mixer", "Right Input Switch", "MIXINR" }, { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, - { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" }, { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, -- 1.6.3.3

1 0