From aff42309d39e9485532a169e6128006990056fe6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 21 Mar 2012 13:22:40 +0000 Subject: [PATCH] --- yaml --- r: 297833 b: refs/heads/master c: e16605855d58803fe0608417150c7a618b4f8243 h: refs/heads/master i: 297831: ac74fedd6c273e76ed042f4ee5472dfbb455f94a v: v3 --- [refs] | 2 +- .../sound/alsa/ALSA-Configuration.txt | 8 +- .../sound/alsa/HD-Audio-Models.txt | 79 +- trunk/Documentation/sound/alsa/HD-Audio.txt | 7 +- trunk/arch/arm/mach-omap1/Kconfig | 3 + trunk/arch/arm/mach-omap1/Makefile | 4 +- trunk/arch/arm/mach-omap1/devices.c | 9 + trunk/arch/arm/mach-omap1/mcbsp.c | 14 +- trunk/arch/arm/mach-omap2/Makefile | 4 +- trunk/arch/arm/mach-omap2/board-omap4panda.c | 2 +- trunk/arch/arm/mach-omap2/devices.c | 22 + trunk/arch/arm/mach-omap2/mcbsp.c | 54 +- trunk/arch/arm/plat-omap/Kconfig | 8 + trunk/arch/arm/plat-omap/Makefile | 2 + trunk/arch/arm/plat-omap/include/plat/mcbsp.h | 333 +++ .../soc/omap => arch/arm/plat-omap}/mcbsp.c | 549 ++++- trunk/include/sound/control.h | 5 - trunk/include/sound/jack.h | 3 - trunk/include/sound/pcm.h | 5 +- trunk/include/sound/version.h | 2 +- trunk/include/sound/ymfpci.h | 2 - trunk/sound/aoa/codecs/onyx.c | 13 +- trunk/sound/aoa/codecs/tas.c | 13 +- trunk/sound/core/control.c | 2 +- trunk/sound/core/init.c | 169 +- trunk/sound/core/jack.c | 4 +- trunk/sound/core/misc.c | 2 +- trunk/sound/core/pcm_lib.c | 3 +- trunk/sound/core/pcm_native.c | 15 +- trunk/sound/core/vmaster.c | 46 +- trunk/sound/pci/asihpi/hpi_internal.h | 2 +- trunk/sound/pci/asihpi/hpios.c | 2 +- trunk/sound/pci/au88x0/au88x0.h | 13 +- trunk/sound/pci/au88x0/au88x0_core.c | 20 +- trunk/sound/pci/au88x0/au88x0_pcm.c | 127 +- trunk/sound/pci/azt3328.c | 3 +- trunk/sound/pci/ctxfi/ctvmem.c | 2 +- trunk/sound/pci/hda/alc260_quirks.c | 968 +++++++++ trunk/sound/pci/hda/alc880_quirks.c | 1707 +++++++++++++++ trunk/sound/pci/hda/alc882_quirks.c | 866 ++++++++ trunk/sound/pci/hda/alc_quirks.c | 480 +++++ trunk/sound/pci/hda/hda_codec.c | 204 +- trunk/sound/pci/hda/hda_codec.h | 4 - trunk/sound/pci/hda/hda_eld.c | 4 +- trunk/sound/pci/hda/hda_intel.c | 52 +- trunk/sound/pci/hda/hda_jack.c | 16 - trunk/sound/pci/hda/hda_jack.h | 13 +- trunk/sound/pci/hda/hda_local.h | 30 +- trunk/sound/pci/hda/patch_analog.c | 72 +- trunk/sound/pci/hda/patch_cirrus.c | 4 +- trunk/sound/pci/hda/patch_conexant.c | 146 +- trunk/sound/pci/hda/patch_hdmi.c | 2 - trunk/sound/pci/hda/patch_realtek.c | 1879 ++++++----------- trunk/sound/pci/hda/patch_sigmatel.c | 205 +- trunk/sound/pci/hda/patch_via.c | 48 +- trunk/sound/pci/ice1712/ice1724.c | 23 - trunk/sound/pci/rme9652/hdspm.c | 1 - trunk/sound/pci/ymfpci/ymfpci_main.c | 9 - trunk/sound/soc/codecs/ak4642.c | 31 +- trunk/sound/soc/codecs/wm8962.c | 2 +- trunk/sound/soc/codecs/wm8994.c | 2 +- trunk/sound/soc/imx/imx-ssi.c | 2 +- trunk/sound/soc/omap/Kconfig | 2 + trunk/sound/soc/omap/Makefile | 2 +- trunk/sound/soc/omap/am3517evm.c | 2 +- trunk/sound/soc/omap/ams-delta.c | 2 +- trunk/sound/soc/omap/igep0020.c | 2 +- trunk/sound/soc/omap/mcbsp.h | 346 --- trunk/sound/soc/omap/n810.c | 2 +- trunk/sound/soc/omap/omap-mcbsp.c | 321 ++- trunk/sound/soc/omap/omap-mcbsp.h | 2 +- trunk/sound/soc/omap/omap-pcm.h | 2 - trunk/sound/soc/omap/omap3beagle.c | 2 +- trunk/sound/soc/omap/omap3evm.c | 2 +- trunk/sound/soc/omap/omap3pandora.c | 4 +- trunk/sound/soc/omap/osk5912.c | 2 +- trunk/sound/soc/omap/overo.c | 2 +- trunk/sound/soc/omap/rx51.c | 4 +- trunk/sound/soc/omap/sdp3430.c | 4 +- trunk/sound/soc/omap/zoom2.c | 4 +- trunk/sound/soc/samsung/neo1973_wm8753.c | 4 +- trunk/sound/soc/soc-dapm.c | 12 +- trunk/sound/spi/at73c213.c | 12 +- trunk/sound/usb/6fire/chip.c | 3 +- trunk/sound/usb/6fire/chip.h | 1 + trunk/sound/usb/6fire/comm.c | 1 + trunk/sound/usb/6fire/comm.h | 1 + trunk/sound/usb/6fire/common.h | 1 + trunk/sound/usb/6fire/control.c | 341 +-- trunk/sound/usb/6fire/control.h | 7 +- trunk/sound/usb/6fire/firmware.c | 1 + trunk/sound/usb/6fire/midi.c | 1 + trunk/sound/usb/6fire/midi.h | 1 + trunk/sound/usb/6fire/pcm.c | 1 + trunk/sound/usb/6fire/pcm.h | 1 + trunk/sound/usb/Kconfig | 1 - trunk/sound/usb/caiaq/audio.c | 5 +- trunk/sound/usb/card.h | 1 - trunk/sound/usb/format.c | 4 +- trunk/sound/usb/pcm.c | 6 +- trunk/sound/usb/quirks.c | 6 +- trunk/sound/usb/usx2y/usbusx2yaudio.c | 4 +- trunk/sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 103 files changed, 6218 insertions(+), 3229 deletions(-) rename trunk/{sound/soc/omap => arch/arm/plat-omap}/mcbsp.c (66%) create mode 100644 trunk/sound/pci/hda/alc260_quirks.c create mode 100644 trunk/sound/pci/hda/alc880_quirks.c create mode 100644 trunk/sound/pci/hda/alc882_quirks.c create mode 100644 trunk/sound/pci/hda/alc_quirks.c delete mode 100644 trunk/sound/soc/omap/mcbsp.h diff --git a/[refs] b/[refs] index ac3a9d08bf57..607aef9da677 100644 --- a/[refs] +++ b/[refs] @@ -1,2 +1,2 @@ --- -refs/heads/master: a938fb1ef600c885f80e59b840a60894d16efb77 +refs/heads/master: e16605855d58803fe0608417150c7a618b4f8243 diff --git a/trunk/Documentation/sound/alsa/ALSA-Configuration.txt b/trunk/Documentation/sound/alsa/ALSA-Configuration.txt index 9af64c508ab4..936699e4f04b 100644 --- a/trunk/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/trunk/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,8 +860,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, - 3 = VIACOMBO, 4 = COMBO) + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the @@ -926,11 +925,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPIB register is more accurate than the position buffer.) - position_fix=3 is specific to VIA devices. The position - of the capture stream is checked from both LPIB and POSBUF - values. position_fix=4 is a combination mode, using LPIB - for playback and POSBUF for capture. - NB: If you get many "azx_get_response timeout" messages at loading, it's likely a problem of interrupts (e.g. ACPI irq routing). Try to boot with options like "pci=noacpi". Also, you diff --git a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt index d97d992ced14..c8c54544abc5 100644 --- a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt @@ -8,10 +8,37 @@ ALC880 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out 6stack 6-jack in back, 2-jack in front 6stack-digout 6-jack with a SPDIF out + w810 3-jack + z71v 3-jack (HP shared SPDIF) + asus 3-jack (ASUS Mobo) + asus-w1v ASUS W1V + asus-dig ASUS with SPDIF out + asus-dig2 ASUS with SPDIF out (using GPIO2) + uniwill 3-jack + fujitsu Fujitsu Laptops (Pi1536) + F1734 2-jack + lg LG laptop (m1 express dual) + lg-lw LG LW20/LW25 laptop + tcl TCL S700 + clevo Clevo laptops (m520G, m665n) + medion Medion Rim 2150 + test for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) ALC260 ====== - N/A + fujitsu Fujitsu S7020 + acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V + favorit100 Maxdata Favorit 100XS + basic fixed pin assignment (old default model) + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) ALC262 ====== @@ -43,7 +70,55 @@ ALC680 ALC882/883/885/888/889 ====================== - N/A + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack digital with SPDIF I/O + arima Arima W820Di1 + targa Targa T8, MSI-1049 T8 + asus-a7j ASUS A7J + asus-a7m ASUS A7M + macpro MacPro support + mb5 Macbook 5,1 + macmini3 Macmini 3,1 + mba21 Macbook Air 2,1 + mbp3 Macbook Pro rev3 + imac24 iMac 24'' with jack detection + imac91 iMac 9,1 + w2jc ASUS W2JC + 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) + alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883) + 3stack-6ch 3-jack 6-channel + 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O + 6stack-dig-demo 6-jack digital for Intel demo board + acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + acer-aspire Acer Aspire 9810 + acer-aspire-4930g Acer Aspire 4930G + acer-aspire-6530g Acer Aspire 6530G + acer-aspire-7730g Acer Aspire 7730G + acer-aspire-8930g Acer Aspire 8930G + medion Medion Laptops + targa-dig Targa/MSI + targa-2ch-dig Targa/MSI with 2-channel + targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) + laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + lenovo-sky Lenovo Sky + haier-w66 Haier W66 + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) + 6stack-dell Dell machines with 6stack (Inspiron 530) + mitac Mitac 8252D + clevo-m540r Clevo M540R (6ch + digital) + clevo-m720 Clevo M720 laptop series + fujitsu-pi2515 Fujitsu AMILO Pi2515 + fujitsu-xa3530 Fujitsu AMILO XA3530 + 3stack-6ch-intel Intel DG33* boards + intel-alc889a Intel IbexPeak with ALC889A + intel-x58 Intel DX58 with ALC889 + asus-p5q ASUS P5Q-EM boards + mb31 MacBook 3,1 + sony-vaio-tt Sony VAIO TT + auto auto-config reading BIOS (default) ALC861/660 ========== diff --git a/trunk/Documentation/sound/alsa/HD-Audio.txt b/trunk/Documentation/sound/alsa/HD-Audio.txt index 7813c06a5c71..91fee3b45fb8 100644 --- a/trunk/Documentation/sound/alsa/HD-Audio.txt +++ b/trunk/Documentation/sound/alsa/HD-Audio.txt @@ -59,12 +59,7 @@ a case, you can change the default method via `position_fix` option. `position_fix=1` means to use LPIB method explicitly. `position_fix=2` means to use the position-buffer. `position_fix=3` means to use a combination of both methods, needed -for some VIA controllers. The capture stream position is corrected -by comparing both LPIB and position-buffer values. -`position_fix=4` is another combination available for all controllers, -and uses LPIB for the playback and the position-buffer for the capture -streams. -0 is the default value for all other +for some VIA and ATI controllers. 0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might help. diff --git a/trunk/arch/arm/mach-omap1/Kconfig b/trunk/arch/arm/mach-omap1/Kconfig index 922ab0dc2bcd..4f8d66f044e7 100644 --- a/trunk/arch/arm/mach-omap1/Kconfig +++ b/trunk/arch/arm/mach-omap1/Kconfig @@ -37,6 +37,7 @@ comment "OMAP Board Type" config MACH_OMAP_INNOVATOR bool "TI Innovator" depends on ARCH_OMAP1 && (ARCH_OMAP15XX || ARCH_OMAP16XX) + select OMAP_MCBSP help TI OMAP 1510 or 1610 Innovator board support. Say Y here if you have such a board. @@ -44,6 +45,7 @@ config MACH_OMAP_INNOVATOR config MACH_OMAP_H2 bool "TI H2 Support" depends on ARCH_OMAP1 && ARCH_OMAP16XX + select OMAP_MCBSP help TI OMAP 1610/1611B H2 board support. Say Y here if you have such a board. @@ -70,6 +72,7 @@ config MACH_HERALD config MACH_OMAP_OSK bool "TI OSK Support" depends on ARCH_OMAP1 && ARCH_OMAP16XX + select OMAP_MCBSP help TI OMAP 5912 OSK (OMAP Starter Kit) board support. Say Y here if you have such a board. diff --git a/trunk/arch/arm/mach-omap1/Makefile b/trunk/arch/arm/mach-omap1/Makefile index 9923f92b5450..11c85cd2731a 100644 --- a/trunk/arch/arm/mach-omap1/Makefile +++ b/trunk/arch/arm/mach-omap1/Makefile @@ -6,9 +6,7 @@ obj-y := io.o id.o sram.o time.o irq.o mux.o flash.o serial.o devices.o dma.o obj-y += clock.o clock_data.o opp_data.o reset.o pm_bus.o timer.o -ifneq ($(CONFIG_SND_OMAP_SOC_MCBSP),) -obj-y += mcbsp.o -endif +obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o obj-$(CONFIG_OMAP_32K_TIMER) += timer32k.o diff --git a/trunk/arch/arm/mach-omap1/devices.c b/trunk/arch/arm/mach-omap1/devices.c index 187b2fe132e9..1d76a63c0983 100644 --- a/trunk/arch/arm/mach-omap1/devices.c +++ b/trunk/arch/arm/mach-omap1/devices.c @@ -28,6 +28,7 @@ #include #include #include +#include #include "clock.h" @@ -249,8 +250,16 @@ static struct platform_device omap_pcm = { .id = -1, }; +OMAP_MCBSP_PLATFORM_DEVICE(1); +OMAP_MCBSP_PLATFORM_DEVICE(2); +OMAP_MCBSP_PLATFORM_DEVICE(3); + static void omap_init_audio(void) { + platform_device_register(&omap_mcbsp1); + platform_device_register(&omap_mcbsp2); + if (!cpu_is_omap7xx()) + platform_device_register(&omap_mcbsp3); platform_device_register(&omap_pcm); } diff --git a/trunk/arch/arm/mach-omap1/mcbsp.c b/trunk/arch/arm/mach-omap1/mcbsp.c index 3e8410a99990..91f9abbd3250 100644 --- a/trunk/arch/arm/mach-omap1/mcbsp.c +++ b/trunk/arch/arm/mach-omap1/mcbsp.c @@ -419,6 +419,18 @@ static int __init omap1_mcbsp_init(void) if (!cpu_class_is_omap1()) return -ENODEV; + if (cpu_is_omap7xx()) + omap_mcbsp_count = OMAP7XX_MCBSP_COUNT; + else if (cpu_is_omap15xx()) + omap_mcbsp_count = OMAP15XX_MCBSP_COUNT; + else if (cpu_is_omap16xx()) + omap_mcbsp_count = OMAP16XX_MCBSP_COUNT; + + mcbsp_ptr = kzalloc(omap_mcbsp_count * sizeof(struct omap_mcbsp *), + GFP_KERNEL); + if (!mcbsp_ptr) + return -ENOMEM; + if (cpu_is_omap7xx()) omap_mcbsp_register_board_cfg(omap7xx_mcbsp_res_0, OMAP7XX_MCBSP_RES_SZ, @@ -437,7 +449,7 @@ static int __init omap1_mcbsp_init(void) omap16xx_mcbsp_pdata, OMAP16XX_MCBSP_COUNT); - return 0; + return omap_mcbsp_init(); } arch_initcall(omap1_mcbsp_init); diff --git a/trunk/arch/arm/mach-omap2/Makefile b/trunk/arch/arm/mach-omap2/Makefile index 06326a6e460d..bd76394ccaf8 100644 --- a/trunk/arch/arm/mach-omap2/Makefile +++ b/trunk/arch/arm/mach-omap2/Makefile @@ -17,9 +17,7 @@ obj-$(CONFIG_ARCH_OMAP2) += $(omap-2-3-common) $(hwmod-common) obj-$(CONFIG_ARCH_OMAP3) += $(omap-2-3-common) $(hwmod-common) $(secure-common) obj-$(CONFIG_ARCH_OMAP4) += prm44xx.o $(hwmod-common) $(secure-common) -ifneq ($(CONFIG_SND_OMAP_SOC_MCBSP),) -obj-y += mcbsp.o -endif +obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o obj-$(CONFIG_TWL4030_CORE) += omap_twl.o diff --git a/trunk/arch/arm/mach-omap2/board-omap4panda.c b/trunk/arch/arm/mach-omap2/board-omap4panda.c index e4415917693f..1af564b0487e 100644 --- a/trunk/arch/arm/mach-omap2/board-omap4panda.c +++ b/trunk/arch/arm/mach-omap2/board-omap4panda.c @@ -530,7 +530,7 @@ void omap4_panda_display_init(void) static void omap4_panda_init_rev(void) { - if (cpu_is_omap443x()) { + if (cpu_is_omap4430()) { /* PandaBoard 4430 */ /* ASoC audio configuration */ panda_abe_audio_data.card_name = "PandaBoard"; diff --git a/trunk/arch/arm/mach-omap2/devices.c b/trunk/arch/arm/mach-omap2/devices.c index e9fae652cd04..283d11eae693 100644 --- a/trunk/arch/arm/mach-omap2/devices.c +++ b/trunk/arch/arm/mach-omap2/devices.c @@ -26,6 +26,7 @@ #include #include +#include #include #include #include @@ -303,8 +304,29 @@ static struct platform_device omap_pcm = { .id = -1, }; +/* + * OMAP2420 has 2 McBSP ports + * OMAP2430 has 5 McBSP ports + * OMAP3 has 5 McBSP ports + * OMAP4 has 4 McBSP ports + */ +OMAP_MCBSP_PLATFORM_DEVICE(1); +OMAP_MCBSP_PLATFORM_DEVICE(2); +OMAP_MCBSP_PLATFORM_DEVICE(3); +OMAP_MCBSP_PLATFORM_DEVICE(4); +OMAP_MCBSP_PLATFORM_DEVICE(5); + static void omap_init_audio(void) { + platform_device_register(&omap_mcbsp1); + platform_device_register(&omap_mcbsp2); + if (cpu_is_omap243x() || cpu_is_omap34xx() || cpu_is_omap44xx()) { + platform_device_register(&omap_mcbsp3); + platform_device_register(&omap_mcbsp4); + } + if (cpu_is_omap243x() || cpu_is_omap34xx()) + platform_device_register(&omap_mcbsp5); + platform_device_register(&omap_pcm); } diff --git a/trunk/arch/arm/mach-omap2/mcbsp.c b/trunk/arch/arm/mach-omap2/mcbsp.c index ecc039e794db..fb4bcf81a183 100644 --- a/trunk/arch/arm/mach-omap2/mcbsp.c +++ b/trunk/arch/arm/mach-omap2/mcbsp.c @@ -34,7 +34,7 @@ #include "cm2xxx_3xxx.h" #include "cm-regbits-34xx.h" -/* McBSP1 internal signal muxing function for OMAP2/3 */ +/* McBSP internal signal muxing function */ static int omap2_mcbsp1_mux_rx_clk(struct device *dev, const char *signal, const char *src) { @@ -65,42 +65,6 @@ static int omap2_mcbsp1_mux_rx_clk(struct device *dev, const char *signal, return 0; } -/* McBSP4 internal signal muxing function for OMAP4 */ -#define OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX (1 << 31) -#define OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX (1 << 30) -static int omap4_mcbsp4_mux_rx_clk(struct device *dev, const char *signal, - const char *src) -{ - u32 v; - - /* - * In CONTROL_MCBSPLP register only bit 30 (CLKR mux), and bit 31 (FSR - * mux) is used */ - v = omap4_ctrl_pad_readl(OMAP4_CTRL_MODULE_PAD_CORE_CONTROL_MCBSPLP); - - if (!strcmp(signal, "clkr")) { - if (!strcmp(src, "clkr")) - v &= ~OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX; - else if (!strcmp(src, "clkx")) - v |= OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX; - else - return -EINVAL; - } else if (!strcmp(signal, "fsr")) { - if (!strcmp(src, "fsr")) - v &= ~OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX; - else if (!strcmp(src, "fsx")) - v |= OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX; - else - return -EINVAL; - } else { - return -EINVAL; - } - - omap4_ctrl_pad_writel(v, OMAP4_CTRL_MODULE_PAD_CORE_CONTROL_MCBSPLP); - - return 0; -} - /* McBSP CLKS source switching function */ static int omap2_mcbsp_set_clk_src(struct device *dev, struct clk *clk, const char *src) @@ -182,15 +146,9 @@ static int omap_init_mcbsp(struct omap_hwmod *oh, void *unused) pdata->has_ccr = true; } pdata->set_clk_src = omap2_mcbsp_set_clk_src; - - /* On OMAP2/3 the McBSP1 port has 6 pin configuration */ - if (id == 1 && oh->class->rev < MCBSP_CONFIG_TYPE4) + if (id == 1) pdata->mux_signal = omap2_mcbsp1_mux_rx_clk; - /* On OMAP4 the McBSP4 port has 6 pin configuration */ - if (id == 4 && oh->class->rev == MCBSP_CONFIG_TYPE4) - pdata->mux_signal = omap4_mcbsp4_mux_rx_clk; - if (oh->class->rev == MCBSP_CONFIG_TYPE3) { if (id == 2) /* The FIFO has 1024 + 256 locations */ @@ -222,6 +180,7 @@ static int omap_init_mcbsp(struct omap_hwmod *oh, void *unused) name, oh->name); return PTR_ERR(pdev); } + omap_mcbsp_count++; return 0; } @@ -229,6 +188,11 @@ static int __init omap2_mcbsp_init(void) { omap_hwmod_for_each_by_class("mcbsp", omap_init_mcbsp, NULL); - return 0; + mcbsp_ptr = kzalloc(omap_mcbsp_count * sizeof(struct omap_mcbsp *), + GFP_KERNEL); + if (!mcbsp_ptr) + return -ENOMEM; + + return omap_mcbsp_init(); } arch_initcall(omap2_mcbsp_init); diff --git a/trunk/arch/arm/plat-omap/Kconfig b/trunk/arch/arm/plat-omap/Kconfig index 8f81503a4df7..aa59f4247dc5 100644 --- a/trunk/arch/arm/plat-omap/Kconfig +++ b/trunk/arch/arm/plat-omap/Kconfig @@ -110,6 +110,14 @@ config OMAP_MUX_WARNINGS to change the pin multiplexing setup. When there are no warnings printed, it's safe to deselect OMAP_MUX for your product. +config OMAP_MCBSP + bool "McBSP support" + depends on ARCH_OMAP + default y + help + Say Y here if you want support for the OMAP Multichannel + Buffered Serial Port. + config OMAP_MBOX_FWK tristate "Mailbox framework support" depends on ARCH_OMAP diff --git a/trunk/arch/arm/plat-omap/Makefile b/trunk/arch/arm/plat-omap/Makefile index c0fe2757b695..9a584614e7e6 100644 --- a/trunk/arch/arm/plat-omap/Makefile +++ b/trunk/arch/arm/plat-omap/Makefile @@ -17,6 +17,8 @@ obj-$(CONFIG_ARCH_OMAP2) += omap_device.o obj-$(CONFIG_ARCH_OMAP3) += omap_device.o obj-$(CONFIG_ARCH_OMAP4) += omap_device.o +obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o + obj-$(CONFIG_OMAP_DM_TIMER) += dmtimer.o obj-$(CONFIG_OMAP_DEBUG_DEVICES) += debug-devices.o obj-$(CONFIG_OMAP_DEBUG_LEDS) += debug-leds.o diff --git a/trunk/arch/arm/plat-omap/include/plat/mcbsp.h b/trunk/arch/arm/plat-omap/include/plat/mcbsp.h index 18814127809a..8fa74e2c9d6e 100644 --- a/trunk/arch/arm/plat-omap/include/plat/mcbsp.h +++ b/trunk/arch/arm/plat-omap/include/plat/mcbsp.h @@ -27,10 +27,271 @@ #include #include +/* macro for building platform_device for McBSP ports */ +#define OMAP_MCBSP_PLATFORM_DEVICE(port_nr) \ +static struct platform_device omap_mcbsp##port_nr = { \ + .name = "omap-mcbsp-dai", \ + .id = port_nr - 1, \ +} + #define MCBSP_CONFIG_TYPE2 0x2 #define MCBSP_CONFIG_TYPE3 0x3 #define MCBSP_CONFIG_TYPE4 0x4 +/* McBSP register numbers. Register address offset = num * reg_step */ +enum { + /* Common registers */ + OMAP_MCBSP_REG_SPCR2 = 4, + OMAP_MCBSP_REG_SPCR1, + OMAP_MCBSP_REG_RCR2, + OMAP_MCBSP_REG_RCR1, + OMAP_MCBSP_REG_XCR2, + OMAP_MCBSP_REG_XCR1, + OMAP_MCBSP_REG_SRGR2, + OMAP_MCBSP_REG_SRGR1, + OMAP_MCBSP_REG_MCR2, + OMAP_MCBSP_REG_MCR1, + OMAP_MCBSP_REG_RCERA, + OMAP_MCBSP_REG_RCERB, + OMAP_MCBSP_REG_XCERA, + OMAP_MCBSP_REG_XCERB, + OMAP_MCBSP_REG_PCR0, + OMAP_MCBSP_REG_RCERC, + OMAP_MCBSP_REG_RCERD, + OMAP_MCBSP_REG_XCERC, + OMAP_MCBSP_REG_XCERD, + OMAP_MCBSP_REG_RCERE, + OMAP_MCBSP_REG_RCERF, + OMAP_MCBSP_REG_XCERE, + OMAP_MCBSP_REG_XCERF, + OMAP_MCBSP_REG_RCERG, + OMAP_MCBSP_REG_RCERH, + OMAP_MCBSP_REG_XCERG, + OMAP_MCBSP_REG_XCERH, + + /* OMAP1-OMAP2420 registers */ + OMAP_MCBSP_REG_DRR2 = 0, + OMAP_MCBSP_REG_DRR1, + OMAP_MCBSP_REG_DXR2, + OMAP_MCBSP_REG_DXR1, + + /* OMAP2430 and onwards */ + OMAP_MCBSP_REG_DRR = 0, + OMAP_MCBSP_REG_DXR = 2, + OMAP_MCBSP_REG_SYSCON = 35, + OMAP_MCBSP_REG_THRSH2, + OMAP_MCBSP_REG_THRSH1, + OMAP_MCBSP_REG_IRQST = 40, + OMAP_MCBSP_REG_IRQEN, + OMAP_MCBSP_REG_WAKEUPEN, + OMAP_MCBSP_REG_XCCR, + OMAP_MCBSP_REG_RCCR, + OMAP_MCBSP_REG_XBUFFSTAT, + OMAP_MCBSP_REG_RBUFFSTAT, + OMAP_MCBSP_REG_SSELCR, +}; + +/* OMAP3 sidetone control registers */ +#define OMAP_ST_REG_REV 0x00 +#define OMAP_ST_REG_SYSCONFIG 0x10 +#define OMAP_ST_REG_IRQSTATUS 0x18 +#define OMAP_ST_REG_IRQENABLE 0x1C +#define OMAP_ST_REG_SGAINCR 0x24 +#define OMAP_ST_REG_SFIRCR 0x28 +#define OMAP_ST_REG_SSELCR 0x2C + +/************************** McBSP SPCR1 bit definitions ***********************/ +#define RRST 0x0001 +#define RRDY 0x0002 +#define RFULL 0x0004 +#define RSYNC_ERR 0x0008 +#define RINTM(value) ((value)<<4) /* bits 4:5 */ +#define ABIS 0x0040 +#define DXENA 0x0080 +#define CLKSTP(value) ((value)<<11) /* bits 11:12 */ +#define RJUST(value) ((value)<<13) /* bits 13:14 */ +#define ALB 0x8000 +#define DLB 0x8000 + +/************************** McBSP SPCR2 bit definitions ***********************/ +#define XRST 0x0001 +#define XRDY 0x0002 +#define XEMPTY 0x0004 +#define XSYNC_ERR 0x0008 +#define XINTM(value) ((value)<<4) /* bits 4:5 */ +#define GRST 0x0040 +#define FRST 0x0080 +#define SOFT 0x0100 +#define FREE 0x0200 + +/************************** McBSP PCR bit definitions *************************/ +#define CLKRP 0x0001 +#define CLKXP 0x0002 +#define FSRP 0x0004 +#define FSXP 0x0008 +#define DR_STAT 0x0010 +#define DX_STAT 0x0020 +#define CLKS_STAT 0x0040 +#define SCLKME 0x0080 +#define CLKRM 0x0100 +#define CLKXM 0x0200 +#define FSRM 0x0400 +#define FSXM 0x0800 +#define RIOEN 0x1000 +#define XIOEN 0x2000 +#define IDLE_EN 0x4000 + +/************************** McBSP RCR1 bit definitions ************************/ +#define RWDLEN1(value) ((value)<<5) /* Bits 5:7 */ +#define RFRLEN1(value) ((value)<<8) /* Bits 8:14 */ + +/************************** McBSP XCR1 bit definitions ************************/ +#define XWDLEN1(value) ((value)<<5) /* Bits 5:7 */ +#define XFRLEN1(value) ((value)<<8) /* Bits 8:14 */ + +/*************************** McBSP RCR2 bit definitions ***********************/ +#define RDATDLY(value) (value) /* Bits 0:1 */ +#define RFIG 0x0004 +#define RCOMPAND(value) ((value)<<3) /* Bits 3:4 */ +#define RWDLEN2(value) ((value)<<5) /* Bits 5:7 */ +#define RFRLEN2(value) ((value)<<8) /* Bits 8:14 */ +#define RPHASE 0x8000 + +/*************************** McBSP XCR2 bit definitions ***********************/ +#define XDATDLY(value) (value) /* Bits 0:1 */ +#define XFIG 0x0004 +#define XCOMPAND(value) ((value)<<3) /* Bits 3:4 */ +#define XWDLEN2(value) ((value)<<5) /* Bits 5:7 */ +#define XFRLEN2(value) ((value)<<8) /* Bits 8:14 */ +#define XPHASE 0x8000 + +/************************* McBSP SRGR1 bit definitions ************************/ +#define CLKGDV(value) (value) /* Bits 0:7 */ +#define FWID(value) ((value)<<8) /* Bits 8:15 */ + +/************************* McBSP SRGR2 bit definitions ************************/ +#define FPER(value) (value) /* Bits 0:11 */ +#define FSGM 0x1000 +#define CLKSM 0x2000 +#define CLKSP 0x4000 +#define GSYNC 0x8000 + +/************************* McBSP MCR1 bit definitions *************************/ +#define RMCM 0x0001 +#define RCBLK(value) ((value)<<2) /* Bits 2:4 */ +#define RPABLK(value) ((value)<<5) /* Bits 5:6 */ +#define RPBBLK(value) ((value)<<7) /* Bits 7:8 */ + +/************************* McBSP MCR2 bit definitions *************************/ +#define XMCM(value) (value) /* Bits 0:1 */ +#define XCBLK(value) ((value)<<2) /* Bits 2:4 */ +#define XPABLK(value) ((value)<<5) /* Bits 5:6 */ +#define XPBBLK(value) ((value)<<7) /* Bits 7:8 */ + +/*********************** McBSP XCCR bit definitions *************************/ +#define EXTCLKGATE 0x8000 +#define PPCONNECT 0x4000 +#define DXENDLY(value) ((value)<<12) /* Bits 12:13 */ +#define XFULL_CYCLE 0x0800 +#define DILB 0x0020 +#define XDMAEN 0x0008 +#define XDISABLE 0x0001 + +/********************** McBSP RCCR bit definitions *************************/ +#define RFULL_CYCLE 0x0800 +#define RDMAEN 0x0008 +#define RDISABLE 0x0001 + +/********************** McBSP SYSCONFIG bit definitions ********************/ +#define CLOCKACTIVITY(value) ((value)<<8) +#define SIDLEMODE(value) ((value)<<3) +#define ENAWAKEUP 0x0004 +#define SOFTRST 0x0002 + +/********************** McBSP SSELCR bit definitions ***********************/ +#define SIDETONEEN 0x0400 + +/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ +#define ST_AUTOIDLE 0x0001 + +/********************** McBSP Sidetone SGAINCR bit definitions *************/ +#define ST_CH1GAIN(value) ((value<<16)) /* Bits 16:31 */ +#define ST_CH0GAIN(value) (value) /* Bits 0:15 */ + +/********************** McBSP Sidetone SFIRCR bit definitions **************/ +#define ST_FIRCOEFF(value) (value) /* Bits 0:15 */ + +/********************** McBSP Sidetone SSELCR bit definitions **************/ +#define ST_COEFFWRDONE 0x0004 +#define ST_COEFFWREN 0x0002 +#define ST_SIDETONEEN 0x0001 + +/********************** McBSP DMA operating modes **************************/ +#define MCBSP_DMA_MODE_ELEMENT 0 +#define MCBSP_DMA_MODE_THRESHOLD 1 +#define MCBSP_DMA_MODE_FRAME 2 + +/********************** McBSP WAKEUPEN bit definitions *********************/ +#define XEMPTYEOFEN 0x4000 +#define XRDYEN 0x0400 +#define XEOFEN 0x0200 +#define XFSXEN 0x0100 +#define XSYNCERREN 0x0080 +#define RRDYEN 0x0008 +#define REOFEN 0x0004 +#define RFSREN 0x0002 +#define RSYNCERREN 0x0001 + +/* CLKR signal muxing options */ +#define CLKR_SRC_CLKR 0 +#define CLKR_SRC_CLKX 1 + +/* FSR signal muxing options */ +#define FSR_SRC_FSR 0 +#define FSR_SRC_FSX 1 + +/* McBSP functional clock sources */ +#define MCBSP_CLKS_PRCM_SRC 0 +#define MCBSP_CLKS_PAD_SRC 1 + +/* we don't do multichannel for now */ +struct omap_mcbsp_reg_cfg { + u16 spcr2; + u16 spcr1; + u16 rcr2; + u16 rcr1; + u16 xcr2; + u16 xcr1; + u16 srgr2; + u16 srgr1; + u16 mcr2; + u16 mcr1; + u16 pcr0; + u16 rcerc; + u16 rcerd; + u16 xcerc; + u16 xcerd; + u16 rcere; + u16 rcerf; + u16 xcere; + u16 xcerf; + u16 rcerg; + u16 rcerh; + u16 xcerg; + u16 xcerh; + u16 xccr; + u16 rccr; +}; + +typedef enum { + OMAP_MCBSP_WORD_8 = 0, + OMAP_MCBSP_WORD_12, + OMAP_MCBSP_WORD_16, + OMAP_MCBSP_WORD_20, + OMAP_MCBSP_WORD_24, + OMAP_MCBSP_WORD_32, +} omap_mcbsp_word_length; + /* Platform specific configuration */ struct omap_mcbsp_ops { void (*request)(unsigned int); @@ -51,6 +312,43 @@ struct omap_mcbsp_platform_data { int (*mux_signal)(struct device *dev, const char *signal, const char *src); }; +struct omap_mcbsp_st_data { + void __iomem *io_base_st; + bool running; + bool enabled; + s16 taps[128]; /* Sidetone filter coefficients */ + int nr_taps; /* Number of filter coefficients in use */ + s16 ch0gain; + s16 ch1gain; +}; + +struct omap_mcbsp { + struct device *dev; + unsigned long phys_base; + unsigned long phys_dma_base; + void __iomem *io_base; + u8 id; + u8 free; + + int rx_irq; + int tx_irq; + + /* DMA stuff */ + u8 dma_rx_sync; + u8 dma_tx_sync; + + /* Protect the field .free, while checking if the mcbsp is in use */ + spinlock_t lock; + struct omap_mcbsp_platform_data *pdata; + struct clk *fclk; + struct omap_mcbsp_st_data *st_data; + int dma_op_mode; + u16 max_tx_thres; + u16 max_rx_thres; + void *reg_cache; + int reg_cache_size; +}; + /** * omap_mcbsp_dev_attr - OMAP McBSP device attributes for omap_hwmod * @sidetone: name of the sidetone device @@ -59,4 +357,39 @@ struct omap_mcbsp_dev_attr { const char *sidetone; }; +extern struct omap_mcbsp **mcbsp_ptr; +extern int omap_mcbsp_count; + +int omap_mcbsp_init(void); +void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config); +void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold); +void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold); +u16 omap_mcbsp_get_max_tx_threshold(unsigned int id); +u16 omap_mcbsp_get_max_rx_threshold(unsigned int id); +u16 omap_mcbsp_get_fifo_size(unsigned int id); +u16 omap_mcbsp_get_tx_delay(unsigned int id); +u16 omap_mcbsp_get_rx_delay(unsigned int id); +int omap_mcbsp_get_dma_op_mode(unsigned int id); +int omap_mcbsp_request(unsigned int id); +void omap_mcbsp_free(unsigned int id); +void omap_mcbsp_start(unsigned int id, int tx, int rx); +void omap_mcbsp_stop(unsigned int id, int tx, int rx); + +/* McBSP functional clock source changing function */ +extern int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id); + +/* McBSP signal muxing API */ +void omap2_mcbsp1_mux_clkr_src(u8 mux); +void omap2_mcbsp1_mux_fsr_src(u8 mux); + +int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream); +int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream); + +/* Sidetone specific API */ +int omap_st_set_chgain(unsigned int id, int channel, s16 chgain); +int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain); +int omap_st_enable(unsigned int id); +int omap_st_disable(unsigned int id); +int omap_st_is_enabled(unsigned int id); + #endif diff --git a/trunk/sound/soc/omap/mcbsp.c b/trunk/arch/arm/plat-omap/mcbsp.c similarity index 66% rename from trunk/sound/soc/omap/mcbsp.c rename to trunk/arch/arm/plat-omap/mcbsp.c index e5f44440d1b9..4b15cd7926d7 100644 --- a/trunk/sound/soc/omap/mcbsp.c +++ b/trunk/arch/arm/plat-omap/mcbsp.c @@ -1,11 +1,9 @@ /* - * sound/soc/omap/mcbsp.c + * linux/arch/arm/plat-omap/mcbsp.c * * Copyright (C) 2004 Nokia Corporation * Author: Samuel Ortiz * - * Contact: Jarkko Nikula - * Peter Ujfalusi * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -26,8 +24,13 @@ #include #include +#include -#include "mcbsp.h" +struct omap_mcbsp **mcbsp_ptr; +int omap_mcbsp_count; + +#define omap_mcbsp_check_valid_id(id) (id < omap_mcbsp_count) +#define id_to_mcbsp_ptr(id) mcbsp_ptr[id]; static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) { @@ -77,8 +80,10 @@ static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) #define MCBSP_ST_WRITE(mcbsp, reg, val) \ omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) -static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) +static void omap_mcbsp_dump_reg(u8 id) { + struct omap_mcbsp *mcbsp = id_to_mcbsp_ptr(id); + dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", MCBSP_READ(mcbsp, DRR2)); @@ -151,9 +156,16 @@ static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) * You either call this function or set the McBSP registers * by yourself before calling omap_mcbsp_start(). */ -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config) +void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) { + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", mcbsp->id, mcbsp->phys_base); @@ -173,10 +185,33 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, MCBSP_WRITE(mcbsp, XCCR, config->xccr); MCBSP_WRITE(mcbsp, RCCR, config->rccr); } - /* Enable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); } +EXPORT_SYMBOL(omap_mcbsp_config); + +/** + * omap_mcbsp_dma_params - returns the dma channel number + * @id - mcbsp id + * @stream - indicates the direction of data flow (rx or tx) + * + * Returns the dma channel number for the rx channel or tx channel + * based on the value of @stream for the requested mcbsp given by @id + */ +int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + if (stream) + return mcbsp->dma_rx_sync; + else + return mcbsp->dma_tx_sync; +} +EXPORT_SYMBOL(omap_mcbsp_dma_ch_params); /** * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register @@ -187,11 +222,17 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, * to be used by DMA for transferring/receiving data based on the value of * @stream for the requested mcbsp given by @id */ -static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, - unsigned int stream) +int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream) { + struct omap_mcbsp *mcbsp; int data_reg; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->pdata->reg_size == 2) { if (stream) data_reg = OMAP_MCBSP_REG_DRR1; @@ -206,6 +247,7 @@ static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; } +EXPORT_SYMBOL(omap_mcbsp_dma_reg_params); static void omap_st_on(struct omap_mcbsp *mcbsp) { @@ -274,11 +316,20 @@ static void omap_st_chgain(struct omap_mcbsp *mcbsp) ST_CH1GAIN(st_data->ch1gain)); } -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) +int omap_st_set_chgain(unsigned int id, int channel, s16 chgain) { - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; int ret = 0; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + if (!st_data) return -ENOENT; @@ -296,12 +347,22 @@ int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) return ret; } +EXPORT_SYMBOL(omap_st_set_chgain); -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) +int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain) { - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; int ret = 0; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + if (!st_data) return -ENOENT; @@ -316,12 +377,13 @@ int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) return ret; } +EXPORT_SYMBOL(omap_st_get_chgain); static int omap_st_start(struct omap_mcbsp *mcbsp) { struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - if (st_data->enabled && !st_data->running) { + if (st_data && st_data->enabled && !st_data->running) { omap_st_fir_write(mcbsp, st_data->taps); omap_st_chgain(mcbsp); @@ -334,9 +396,18 @@ static int omap_st_start(struct omap_mcbsp *mcbsp) return 0; } -int omap_st_enable(struct omap_mcbsp *mcbsp) +int omap_st_enable(unsigned int id) { - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; if (!st_data) return -ENODEV; @@ -348,12 +419,13 @@ int omap_st_enable(struct omap_mcbsp *mcbsp) return 0; } +EXPORT_SYMBOL(omap_st_enable); static int omap_st_stop(struct omap_mcbsp *mcbsp) { struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - if (st_data->running) { + if (st_data && st_data->running) { if (!mcbsp->free) { omap_st_off(mcbsp); st_data->running = 0; @@ -363,11 +435,20 @@ static int omap_st_stop(struct omap_mcbsp *mcbsp) return 0; } -int omap_st_disable(struct omap_mcbsp *mcbsp) +int omap_st_disable(unsigned int id) { - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; int ret = 0; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + if (!st_data) return -ENODEV; @@ -378,52 +459,136 @@ int omap_st_disable(struct omap_mcbsp *mcbsp) return ret; } +EXPORT_SYMBOL(omap_st_disable); -int omap_st_is_enabled(struct omap_mcbsp *mcbsp) +int omap_st_is_enabled(unsigned int id) { - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; if (!st_data) return -ENODEV; + return st_data->enabled; } +EXPORT_SYMBOL(omap_st_is_enabled); /* * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. * The threshold parameter is 1 based, and it is converted (threshold - 1) * for the THRSH2 register. */ -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold) { + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return; if (threshold && threshold <= mcbsp->max_tx_thres) MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); } +EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold); /* * omap_mcbsp_set_rx_threshold configures the receive threshold in words. * The threshold parameter is 1 based, and it is converted (threshold - 1) * for the THRSH1 register. */ -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold) { + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return; if (threshold && threshold <= mcbsp->max_rx_thres) MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); } +EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold); + +/* + * omap_mcbsp_get_max_tx_thres just return the current configured + * maximum threshold for transmission + */ +u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->max_tx_thres; +} +EXPORT_SYMBOL(omap_mcbsp_get_max_tx_threshold); + +/* + * omap_mcbsp_get_max_rx_thres just return the current configured + * maximum threshold for reception + */ +u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->max_rx_thres; +} +EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold); + +u16 omap_mcbsp_get_fifo_size(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->pdata->buffer_size; +} +EXPORT_SYMBOL(omap_mcbsp_get_fifo_size); /* * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO */ -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) +u16 omap_mcbsp_get_tx_delay(unsigned int id) { + struct omap_mcbsp *mcbsp; u16 buffstat; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return 0; @@ -433,15 +598,22 @@ u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) /* Number of slots are different in McBSP ports */ return mcbsp->pdata->buffer_size - buffstat; } +EXPORT_SYMBOL(omap_mcbsp_get_tx_delay); /* * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO * to reach the threshold value (when the DMA will be triggered to read it) */ -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) +u16 omap_mcbsp_get_rx_delay(unsigned int id) { + struct omap_mcbsp *mcbsp; u16 buffstat, threshold; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return 0; @@ -456,12 +628,41 @@ u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) else return threshold - buffstat; } +EXPORT_SYMBOL(omap_mcbsp_get_rx_delay); + +/* + * omap_mcbsp_get_dma_op_mode just return the current configured + * operating mode for the mcbsp channel + */ +int omap_mcbsp_get_dma_op_mode(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + int dma_op_mode; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%u)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + dma_op_mode = mcbsp->dma_op_mode; + + return dma_op_mode; +} +EXPORT_SYMBOL(omap_mcbsp_get_dma_op_mode); -int omap_mcbsp_request(struct omap_mcbsp *mcbsp) +int omap_mcbsp_request(unsigned int id) { + struct omap_mcbsp *mcbsp; void *reg_cache; int err; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); if (!reg_cache) { return -ENOMEM; @@ -480,7 +681,13 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) spin_unlock(&mcbsp->lock); if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) - mcbsp->pdata->ops->request(mcbsp->id - 1); + mcbsp->pdata->ops->request(id); + + pm_runtime_get_sync(mcbsp->dev); + + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); /* * Make sure that transmitter, receiver and sample-rate generator are @@ -515,12 +722,14 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) free_irq(mcbsp->tx_irq, (void *)mcbsp); err_clk_disable: if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); + mcbsp->pdata->ops->free(id); /* Disable wakeup behavior */ if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + pm_runtime_put_sync(mcbsp->dev); + spin_lock(&mcbsp->lock); mcbsp->free = true; mcbsp->reg_cache = NULL; @@ -530,34 +739,34 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) return err; } +EXPORT_SYMBOL(omap_mcbsp_request); -void omap_mcbsp_free(struct omap_mcbsp *mcbsp) +void omap_mcbsp_free(unsigned int id) { + struct omap_mcbsp *mcbsp; void *reg_cache; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); + mcbsp->pdata->ops->free(id); /* Disable wakeup behavior */ if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + pm_runtime_put_sync(mcbsp->dev); + if (mcbsp->rx_irq) free_irq(mcbsp->rx_irq, (void *)mcbsp); free_irq(mcbsp->tx_irq, (void *)mcbsp); reg_cache = mcbsp->reg_cache; - /* - * Select CLKS source from internal source unconditionally before - * marking the McBSP port as free. - * If the external clock source via MCBSP_CLKS pin has been selected the - * system will refuse to enter idle if the CLKS pin source is not reset - * back to internal source. - */ - if (!cpu_class_is_omap1()) - omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); - spin_lock(&mcbsp->lock); if (mcbsp->free) dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); @@ -569,17 +778,25 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) if (reg_cache) kfree(reg_cache); } +EXPORT_SYMBOL(omap_mcbsp_free); /* * Here we start the McBSP, by enabling transmitter, receiver or both. * If no transmitter or receiver is active prior calling, then sample-rate * generator and frame sync are started. */ -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) +void omap_mcbsp_start(unsigned int id, int tx, int rx) { + struct omap_mcbsp *mcbsp; int enable_srg = 0; u16 w; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -629,14 +846,23 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) } /* Dump McBSP Regs */ - omap_mcbsp_dump_reg(mcbsp); + omap_mcbsp_dump_reg(id); } +EXPORT_SYMBOL(omap_mcbsp_start); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) +void omap_mcbsp_stop(unsigned int id, int tx, int rx) { + struct omap_mcbsp *mcbsp; int idle; u16 w; + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + + mcbsp = id_to_mcbsp_ptr(id); + /* Reset transmitter */ tx &= 1; if (mcbsp->pdata->has_ccr) { @@ -669,11 +895,19 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); } +EXPORT_SYMBOL(omap_mcbsp_stop); -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) +int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id) { + struct omap_mcbsp *mcbsp; const char *src; + if (!omap_mcbsp_check_valid_id(id)) { + pr_err("%s: Invalid id (%d)\n", __func__, id + 1); + return -EINVAL; + } + mcbsp = id_to_mcbsp_ptr(id); + if (fck_src_id == MCBSP_CLKS_PAD_SRC) src = "clks_ext"; else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) @@ -686,37 +920,43 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) else return -EINVAL; } +EXPORT_SYMBOL(omap2_mcbsp_set_clks_src); -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) +void omap2_mcbsp1_mux_clkr_src(u8 mux) { - const char *signal, *src; - - if (mcbsp->pdata->mux_signal) - return -EINVAL; + struct omap_mcbsp *mcbsp; + const char *src; - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; + if (mux == CLKR_SRC_CLKR) src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; + else if (mux == CLKR_SRC_CLKX) src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; + else + return; + + mcbsp = id_to_mcbsp_ptr(0); + if (mcbsp->pdata->mux_signal) + mcbsp->pdata->mux_signal(mcbsp->dev, "clkr", src); +} +EXPORT_SYMBOL(omap2_mcbsp1_mux_clkr_src); + +void omap2_mcbsp1_mux_fsr_src(u8 mux) +{ + struct omap_mcbsp *mcbsp; + const char *src; + + if (mux == FSR_SRC_FSR) src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; + else if (mux == FSR_SRC_FSX) src = "fsx"; - break; - default: - return -EINVAL; - } + else + return; - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); + mcbsp = id_to_mcbsp_ptr(0); + if (mcbsp->pdata->mux_signal) + mcbsp->pdata->mux_signal(mcbsp->dev, "fsr", src); } +EXPORT_SYMBOL(omap2_mcbsp1_mux_fsr_src); #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) @@ -892,56 +1132,97 @@ static int __devinit omap_st_add(struct omap_mcbsp *mcbsp, struct omap_mcbsp_st_data *st_data; int err; - st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); - if (!st_data) - return -ENOMEM; + st_data = kzalloc(sizeof(*mcbsp->st_data), GFP_KERNEL); + if (!st_data) { + err = -ENOMEM; + goto err1; + } - st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, - resource_size(res)); - if (!st_data->io_base_st) - return -ENOMEM; + st_data->io_base_st = ioremap(res->start, resource_size(res)); + if (!st_data->io_base_st) { + err = -ENOMEM; + goto err2; + } err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); if (err) - return err; + goto err3; mcbsp->st_data = st_data; return 0; + +err3: + iounmap(st_data->io_base_st); +err2: + kfree(st_data); +err1: + return err; + +} + +static void __devexit omap_st_remove(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); + iounmap(st_data->io_base_st); + kfree(st_data); } /* * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. * 730 has only 2 McBSP, and both of them are MPU peripherals. */ -int __devinit omap_mcbsp_init(struct platform_device *pdev) +static int __devinit omap_mcbsp_probe(struct platform_device *pdev) { - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct omap_mcbsp_platform_data *pdata = pdev->dev.platform_data; + struct omap_mcbsp *mcbsp; + int id = pdev->id - 1; struct resource *res; int ret = 0; + if (!pdata) { + dev_err(&pdev->dev, "McBSP device initialized without" + "platform data\n"); + ret = -EINVAL; + goto exit; + } + + dev_dbg(&pdev->dev, "Initializing OMAP McBSP (%d).\n", pdev->id); + + if (id >= omap_mcbsp_count) { + dev_err(&pdev->dev, "Invalid McBSP device id (%d)\n", id); + ret = -EINVAL; + goto exit; + } + + mcbsp = kzalloc(sizeof(struct omap_mcbsp), GFP_KERNEL); + if (!mcbsp) { + ret = -ENOMEM; + goto exit; + } + spin_lock_init(&mcbsp->lock); + mcbsp->id = id + 1; mcbsp->free = true; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!res) { res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { - dev_err(mcbsp->dev, "invalid memory resource\n"); - return -ENOMEM; + dev_err(&pdev->dev, "%s:mcbsp%d has invalid memory" + "resource\n", __func__, pdev->id); + ret = -ENOMEM; + goto exit; } } - if (!devm_request_mem_region(&pdev->dev, res->start, resource_size(res), - dev_name(&pdev->dev))) { - dev_err(mcbsp->dev, "memory region already claimed\n"); - return -ENODEV; - } - mcbsp->phys_base = res->start; mcbsp->reg_cache_size = resource_size(res); - mcbsp->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!mcbsp->io_base) - return -ENOMEM; + mcbsp->io_base = ioremap(res->start, resource_size(res)); + if (!mcbsp->io_base) { + ret = -ENOMEM; + goto err_ioremap; + } res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); if (!res) @@ -953,38 +1234,40 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); /* From OMAP4 there will be a single irq line */ - if (mcbsp->tx_irq == -ENXIO) { + if (mcbsp->tx_irq == -ENXIO) mcbsp->tx_irq = platform_get_irq(pdev, 0); - mcbsp->rx_irq = 0; - } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; + dev_err(&pdev->dev, "%s:mcbsp%d has invalid rx DMA channel\n", + __func__, pdev->id); + ret = -ENODEV; + goto err_res; } - /* RX DMA request number, and port address configuration */ - mcbsp->dma_data[1].name = "Audio Capture"; - mcbsp->dma_data[1].dma_req = res->start; - mcbsp->dma_data[1].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 1); + mcbsp->dma_rx_sync = res->start; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; + dev_err(&pdev->dev, "%s:mcbsp%d has invalid tx DMA channel\n", + __func__, pdev->id); + ret = -ENODEV; + goto err_res; } - /* TX DMA request number, and port address configuration */ - mcbsp->dma_data[0].name = "Audio Playback"; - mcbsp->dma_data[0].dma_req = res->start; - mcbsp->dma_data[0].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 0); + mcbsp->dma_tx_sync = res->start; mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); - dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); - return ret; + dev_err(&pdev->dev, "unable to get fck: %d\n", ret); + goto err_res; } + mcbsp->pdata = pdata; + mcbsp->dev = &pdev->dev; + mcbsp_ptr[id] = mcbsp; + platform_set_drvdata(pdev, mcbsp); + pm_runtime_enable(mcbsp->dev); + mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; if (mcbsp->pdata->buffer_size) { /* @@ -1024,17 +1307,55 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) err_st: if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); + sysfs_remove_group(&mcbsp->dev->kobj, + &additional_attr_group); err_thres: clk_put(mcbsp->fclk); +err_res: + iounmap(mcbsp->io_base); +err_ioremap: + kfree(mcbsp); +exit: return ret; } -void __devexit omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp) +static int __devexit omap_mcbsp_remove(struct platform_device *pdev) { - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - if (mcbsp->st_data) - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); + platform_set_drvdata(pdev, NULL); + if (mcbsp) { + + if (mcbsp->pdata && mcbsp->pdata->ops && + mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id); + + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, + &additional_attr_group); + + if (mcbsp->st_data) + omap_st_remove(mcbsp); + + clk_put(mcbsp->fclk); + + iounmap(mcbsp->io_base); + kfree(mcbsp); + } + + return 0; +} + +static struct platform_driver omap_mcbsp_driver = { + .probe = omap_mcbsp_probe, + .remove = __devexit_p(omap_mcbsp_remove), + .driver = { + .name = "omap-mcbsp", + }, +}; + +int __init omap_mcbsp_init(void) +{ + /* Register the McBSP driver */ + return platform_driver_register(&omap_mcbsp_driver); } diff --git a/trunk/include/sound/control.h b/trunk/include/sound/control.h index 8332e865c759..57815f6a02b9 100644 --- a/trunk/include/sound/control.h +++ b/trunk/include/sound/control.h @@ -227,11 +227,6 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master, return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE); } -int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl, - void (*hook)(void *private_data, int), - void *private_data); -void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kctl); - /* * Helper functions for jack-detection controls */ diff --git a/trunk/include/sound/jack.h b/trunk/include/sound/jack.h index 58916573db58..63c790742db4 100644 --- a/trunk/include/sound/jack.h +++ b/trunk/include/sound/jack.h @@ -53,9 +53,6 @@ enum snd_jack_types { SND_JACK_BTN_5 = 0x0200, }; -/* Keep in sync with definitions above */ -#define SND_JACK_SWITCH_TYPES 6 - struct snd_jack { struct input_dev *input_dev; int registered; diff --git a/trunk/include/sound/pcm.h b/trunk/include/sound/pcm.h index 0d1112815be3..1d58d7957390 100644 --- a/trunk/include/sound/pcm.h +++ b/trunk/include/sound/pcm.h @@ -264,7 +264,7 @@ struct snd_pcm_hw_constraint_ratdens { struct snd_pcm_hw_constraint_list { unsigned int count; - const unsigned int *list; + unsigned int *list; unsigned int mask; }; @@ -785,8 +785,7 @@ void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interva unsigned int k, struct snd_interval *c); void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, const struct snd_interval *b, struct snd_interval *c); -int snd_interval_list(struct snd_interval *i, unsigned int count, - const unsigned int *list, unsigned int mask); +int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask); int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp); diff --git a/trunk/include/sound/version.h b/trunk/include/sound/version.h index cc75024c1089..8fc5321e1ecc 100644 --- a/trunk/include/sound/version.h +++ b/trunk/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.25" +#define CONFIG_SND_VERSION "1.0.24" #define CONFIG_SND_DATE "" diff --git a/trunk/include/sound/ymfpci.h b/trunk/include/sound/ymfpci.h index 41199664666b..444cd6ba0ba7 100644 --- a/trunk/include/sound/ymfpci.h +++ b/trunk/include/sound/ymfpci.h @@ -366,8 +366,6 @@ struct snd_ymfpci { #ifdef CONFIG_PM u32 *saved_regs; u32 saved_ydsxgr_mode; - u16 saved_dsxg_legacy; - u16 saved_dsxg_elegacy; #endif }; diff --git a/trunk/sound/aoa/codecs/onyx.c b/trunk/sound/aoa/codecs/onyx.c index 270790d384e2..762af68c8996 100644 --- a/trunk/sound/aoa/codecs/onyx.c +++ b/trunk/sound/aoa/codecs/onyx.c @@ -1132,4 +1132,15 @@ static struct i2c_driver onyx_driver = { .id_table = onyx_i2c_id, }; -module_i2c_driver(onyx_driver); +static int __init onyx_init(void) +{ + return i2c_add_driver(&onyx_driver); +} + +static void __exit onyx_exit(void) +{ + i2c_del_driver(&onyx_driver); +} + +module_init(onyx_init); +module_exit(onyx_exit); diff --git a/trunk/sound/aoa/codecs/tas.c b/trunk/sound/aoa/codecs/tas.c index 8e63d1f35ce1..fd2188c3df2b 100644 --- a/trunk/sound/aoa/codecs/tas.c +++ b/trunk/sound/aoa/codecs/tas.c @@ -1026,4 +1026,15 @@ static struct i2c_driver tas_driver = { .id_table = tas_i2c_id, }; -module_i2c_driver(tas_driver); +static int __init tas_init(void) +{ + return i2c_add_driver(&tas_driver); +} + +static void __exit tas_exit(void) +{ + i2c_del_driver(&tas_driver); +} + +module_init(tas_init); +module_exit(tas_exit); diff --git a/trunk/sound/core/control.c b/trunk/sound/core/control.c index 2487a6bb1c54..819a5c579a39 100644 --- a/trunk/sound/core/control.c +++ b/trunk/sound/core/control.c @@ -1313,7 +1313,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, err = -EPERM; goto __kctl_end; } - err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv); + err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv); if (err > 0) { up_read(&card->controls_rwsem); snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_TLV, &kctl->id); diff --git a/trunk/sound/core/init.c b/trunk/sound/core/init.c index 068cf08d3ffb..3ac49b1b7cb8 100644 --- a/trunk/sound/core/init.c +++ b/trunk/sound/core/init.c @@ -480,104 +480,74 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -/* retrieve the last word of shortname or longname */ -static const char *retrieve_id_from_card_name(const char *name) +static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid) { - const char *spos = name; - - while (*name) { - if (isspace(*name) && isalnum(name[1])) - spos = name + 1; - name++; - } - return spos; -} - -/* return true if the given id string doesn't conflict any other card ids */ -static bool card_id_ok(struct snd_card *card, const char *id) -{ - int i; - if (!snd_info_check_reserved_words(id)) - return false; - for (i = 0; i < snd_ecards_limit; i++) { - if (snd_cards[i] && snd_cards[i] != card && - !strcmp(snd_cards[i]->id, id)) - return false; - } - return true; -} - -/* copy to card->id only with valid letters from nid */ -static void copy_valid_id_string(struct snd_card *card, const char *src, - const char *nid) -{ - char *id = card->id; - - while (*nid && !isalnum(*nid)) - nid++; - if (isdigit(*nid)) - *id++ = isalpha(*src) ? *src : 'D'; - while (*nid && (size_t)(id - card->id) < sizeof(card->id) - 1) { - if (isalnum(*nid)) - *id++ = *nid; - nid++; - } - *id = 0; -} - -/* Set card->id from the given string - * If the string conflicts with other ids, add a suffix to make it unique. - */ -static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, - const char *nid) -{ - int len, loops; - bool with_suffix; - bool is_default = false; + int i, len, idx_flag = 0, loops = SNDRV_CARDS; + const char *spos, *src; char *id; - copy_valid_id_string(card, src, nid); + if (nid == NULL) { + id = card->shortname; + spos = src = id; + while (*id != '\0') { + if (*id == ' ') + spos = id + 1; + id++; + } + } else { + spos = src = nid; + } id = card->id; + while (*spos != '\0' && !isalnum(*spos)) + spos++; + if (isdigit(*spos)) + *id++ = isalpha(src[0]) ? src[0] : 'D'; + while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) { + if (isalnum(*spos)) + *id++ = *spos; + spos++; + } + *id = '\0'; - again: - /* use "Default" for obviously invalid strings - * ("card" conflicts with proc directories) - */ - if (!*id || !strncmp(id, "card", 4)) { + id = card->id; + + if (*id == '\0') strcpy(id, "Default"); - is_default = true; - } - with_suffix = false; - for (loops = 0; loops < SNDRV_CARDS; loops++) { - if (card_id_ok(card, id)) - return; /* OK */ + while (1) { + if (loops-- == 0) { + snd_printk(KERN_ERR "unable to set card id (%s)\n", id); + strcpy(card->id, card->proc_root->name); + return; + } + if (!snd_info_check_reserved_words(id)) + goto __change; + for (i = 0; i < snd_ecards_limit; i++) { + if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) + goto __change; + } + break; + __change: len = strlen(id); - if (!with_suffix) { - /* add the "_X" suffix */ - char *spos = id + len; - if (len > sizeof(card->id) - 3) - spos = id + sizeof(card->id) - 3; - strcpy(spos, "_1"); - with_suffix = true; - } else { - /* modify the existing suffix */ - if (id[len - 1] != '9') - id[len - 1]++; + if (idx_flag) { + if (id[len-1] != '9') + id[len-1]++; else - id[len - 1] = 'A'; + id[len-1] = 'A'; + } else if ((size_t)len <= sizeof(card->id) - 3) { + strcat(id, "_1"); + idx_flag++; + } else { + spos = id + len - 2; + if ((size_t)len <= sizeof(card->id) - 2) + spos++; + *(char *)spos++ = '_'; + *(char *)spos++ = '1'; + *(char *)spos++ = '\0'; + idx_flag++; } } - /* fallback to the default id */ - if (!is_default) { - *id = 0; - goto again; - } - /* last resort... */ - snd_printk(KERN_ERR "unable to set card id (%s)\n", id); - if (card->proc_root->name) - strcpy(card->id, card->proc_root->name); } /** @@ -594,7 +564,7 @@ void snd_card_set_id(struct snd_card *card, const char *nid) if (card->id[0] != '\0') return; mutex_lock(&snd_card_mutex); - snd_card_set_id_no_lock(card, nid, nid); + snd_card_set_id_no_lock(card, nid); mutex_unlock(&snd_card_mutex); } EXPORT_SYMBOL(snd_card_set_id); @@ -626,12 +596,22 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, memcpy(buf1, buf, copy); buf1[copy] = '\0'; mutex_lock(&snd_card_mutex); - if (!card_id_ok(NULL, buf1)) { + if (!snd_info_check_reserved_words(buf1)) { + __exist: mutex_unlock(&snd_card_mutex); return -EEXIST; } + for (idx = 0; idx < snd_ecards_limit; idx++) { + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) { + if (card == snd_cards[idx]) + goto __ok; + else + goto __exist; + } + } strcpy(card->id, buf1); snd_info_card_id_change(card); +__ok: mutex_unlock(&snd_card_mutex); return count; @@ -685,18 +665,7 @@ int snd_card_register(struct snd_card *card) mutex_unlock(&snd_card_mutex); return 0; } - if (*card->id) { - /* make a unique id name from the given string */ - char tmpid[sizeof(card->id)]; - memcpy(tmpid, card->id, sizeof(card->id)); - snd_card_set_id_no_lock(card, tmpid, tmpid); - } else { - /* create an id from either shortname or longname */ - const char *src; - src = *card->shortname ? card->shortname : card->longname; - snd_card_set_id_no_lock(card, src, - retrieve_id_from_card_name(src)); - } + snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id); snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); diff --git a/trunk/sound/core/jack.c b/trunk/sound/core/jack.c index 471e1e3b0a99..26edf63b265f 100644 --- a/trunk/sound/core/jack.c +++ b/trunk/sound/core/jack.c @@ -25,7 +25,7 @@ #include #include -static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { +static int jack_switch_types[] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, @@ -128,7 +128,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) + for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); diff --git a/trunk/sound/core/misc.c b/trunk/sound/core/misc.c index 768167925409..465f0ce772cb 100644 --- a/trunk/sound/core/misc.c +++ b/trunk/sound/core/misc.c @@ -72,7 +72,7 @@ void __snd_printk(unsigned int level, const char *path, int line, char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; #endif -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_DEBUG if (debug < level) return; #endif diff --git a/trunk/sound/core/pcm_lib.c b/trunk/sound/core/pcm_lib.c index 4d18941178e6..3420bd3da5d7 100644 --- a/trunk/sound/core/pcm_lib.c +++ b/trunk/sound/core/pcm_lib.c @@ -1029,8 +1029,7 @@ static int snd_interval_ratden(struct snd_interval *i, * * Returns non-zero if the value is changed, zero if not changed. */ -int snd_interval_list(struct snd_interval *i, unsigned int count, - const unsigned int *list, unsigned int mask) +int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask) { unsigned int k; struct snd_interval list_range; diff --git a/trunk/sound/core/pcm_native.c b/trunk/sound/core/pcm_native.c index 3fe99e644eb8..25ed9fe41b89 100644 --- a/trunk/sound/core/pcm_native.c +++ b/trunk/sound/core/pcm_native.c @@ -1586,18 +1586,12 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) struct file *file; struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream1; - struct snd_pcm_group *group; file = snd_pcm_file_fd(fd); if (!file) return -EBADFD; pcm_file = file->private_data; substream1 = pcm_file->substream; - group = kmalloc(sizeof(*group), GFP_KERNEL); - if (!group) { - res = -ENOMEM; - goto _nolock; - } down_write(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || @@ -1610,7 +1604,11 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) goto _end; } if (!snd_pcm_stream_linked(substream)) { - substream->group = group; + substream->group = kmalloc(sizeof(struct snd_pcm_group), GFP_ATOMIC); + if (substream->group == NULL) { + res = -ENOMEM; + goto _end; + } spin_lock_init(&substream->group->lock); INIT_LIST_HEAD(&substream->group->substreams); list_add_tail(&substream->link_list, &substream->group->substreams); @@ -1622,10 +1620,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) _end: write_unlock_irq(&snd_pcm_link_rwlock); up_write(&snd_pcm_link_rwsem); - _nolock: fput(file); - if (res < 0) - kfree(group); return res; } diff --git a/trunk/sound/core/vmaster.c b/trunk/sound/core/vmaster.c index 14a286a7bf2b..130cfe677d60 100644 --- a/trunk/sound/core/vmaster.c +++ b/trunk/sound/core/vmaster.c @@ -37,8 +37,6 @@ struct link_master { struct link_ctl_info info; int val; /* the master value */ unsigned int tlv[4]; - void (*hook)(void *private_data, int); - void *hook_private_data; }; /* @@ -128,9 +126,7 @@ static int master_init(struct link_master *master) master->info.count = 1; /* always mono */ /* set full volume as default (= no attenuation) */ master->val = master->info.max_val; - if (master->hook) - master->hook(master->hook_private_data, master->val); - return 1; + return 0; } return -ENOENT; } @@ -333,8 +329,6 @@ static int master_put(struct snd_kcontrol *kcontrol, slave_put_val(slave, uval); } kfree(uval); - if (master->hook && !err) - master->hook(master->hook_private_data, master->val); return 1; } @@ -414,41 +408,3 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, return kctl; } EXPORT_SYMBOL(snd_ctl_make_virtual_master); - -/** - * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control - * @kcontrol: vmaster kctl element - * @hook: the hook function - * - * Adds the given hook to the vmaster control element so that it's called - * at each time when the value is changed. - */ -int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol, - void (*hook)(void *private_data, int), - void *private_data) -{ - struct link_master *master = snd_kcontrol_chip(kcontrol); - master->hook = hook; - master->hook_private_data = private_data; - return 0; -} -EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook); - -/** - * snd_ctl_sync_vmaster_hook - Sync the vmaster hook - * @kcontrol: vmaster kctl element - * - * Call the hook function to synchronize with the current value of the given - * vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't - * exist. - */ -void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol) -{ - struct link_master *master; - if (!kcontrol) - return; - master = snd_kcontrol_chip(kcontrol); - if (master->hook) - master->hook(master->hook_private_data, master->val); -} -EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook); diff --git a/trunk/sound/pci/asihpi/hpi_internal.h b/trunk/sound/pci/asihpi/hpi_internal.h index 8c63200cf339..4cc315daeda0 100644 --- a/trunk/sound/pci/asihpi/hpi_internal.h +++ b/trunk/sound/pci/asihpi/hpi_internal.h @@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned. If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle. */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, /**< memory handle */ u32 size, /**< Size in bytes to allocate */ struct pci_dev *p_os_reference diff --git a/trunk/sound/pci/asihpi/hpios.c b/trunk/sound/pci/asihpi/hpios.c index 87f4385fe8c7..2d7d1c2e1d0d 100644 --- a/trunk/sound/pci/asihpi/hpios.c +++ b/trunk/sound/pci/asihpi/hpios.c @@ -43,7 +43,7 @@ void hpios_delay_micro_seconds(u32 num_micro_sec) On error, return -ENOMEM, and *pMemArea.size = 0 */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) { /*?? any benefit in using managed dmam_alloc_coherent? */ diff --git a/trunk/sound/pci/au88x0/au88x0.h b/trunk/sound/pci/au88x0/au88x0.h index 466a5c8e8354..bb938153a964 100644 --- a/trunk/sound/pci/au88x0/au88x0.h +++ b/trunk/sound/pci/au88x0/au88x0.h @@ -26,7 +26,7 @@ #include #include #include -#include + #endif #ifndef CHIP_AU8820 @@ -107,14 +107,6 @@ #define NR_WTPB 0x20 /* WT channels per each bank. */ #define NR_PCM 0x10 -struct pcm_vol { - struct snd_kcontrol *kctl; - int active; - int dma; - int mixin[4]; - int vol[4]; -}; - /* Structs */ typedef struct { //int this_08; /* Still unknown */ @@ -176,7 +168,6 @@ struct snd_vortex { /* Xtalk canceler */ int xt_mode; /* 1: speakers, 0:headphones. */ #endif - struct pcm_vol pcm_vol[NR_PCM]; int isquad; /* cache of extended ID codec flag. */ @@ -248,7 +239,7 @@ static int vortex_alsafmt_aspfmt(int alsafmt); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, - int dir, int type, int subdev); + int dir, int type); static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype); #ifndef CHIP_AU8810 diff --git a/trunk/sound/pci/au88x0/au88x0_core.c b/trunk/sound/pci/au88x0/au88x0_core.c index 525f881f0409..6933a27a5d76 100644 --- a/trunk/sound/pci/au88x0/au88x0_core.c +++ b/trunk/sound/pci/au88x0/au88x0_core.c @@ -2050,6 +2050,8 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) } /* Default Connections */ +static int +vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type); static void vortex_connect_default(vortex_t * vortex, int en) { @@ -2109,13 +2111,15 @@ static void vortex_connect_default(vortex_t * vortex, int en) Return: Return allocated DMA or same DMA passed as "dma" when dma >= 0. */ static int -vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, - int type, int subdev) +vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) { stream_t *stream; int i, en; - struct pcm_vol *p; + if ((nr_ch == 3) + || ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2))) + return -EBUSY; + if (dma >= 0) { en = 0; vortex_adb_checkinout(vortex, @@ -2246,14 +2250,6 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, MIX_DEFIGAIN); #endif } - if (stream->type == VORTEX_PCM_ADB && en) { - p = &vortex->pcm_vol[subdev]; - p->dma = dma; - for (i = 0; i < nr_ch; i++) - p->mixin[i] = mix[i]; - for (i = 0; i < ch_top; i++) - p->vol[i] = 0; - } } #ifndef CHIP_AU8820 else { @@ -2477,7 +2473,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) hwread(vortex->mmio, VORTEX_IRQ_STAT); handled = 1; } - if ((source & IRQ_MIDI) && vortex->rmidi) { + if (source & IRQ_MIDI) { snd_mpu401_uart_interrupt(vortex->irq, vortex->rmidi->private_data); handled = 1; diff --git a/trunk/sound/pci/au88x0/au88x0_pcm.c b/trunk/sound/pci/au88x0/au88x0_pcm.c index e59f120742a4..0ef2f9712208 100644 --- a/trunk/sound/pci/au88x0/au88x0_pcm.c +++ b/trunk/sound/pci/au88x0/au88x0_pcm.c @@ -122,18 +122,6 @@ static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = { .mask = 0, }; #endif - -static void vortex_notify_pcm_vol_change(struct snd_card *card, - struct snd_kcontrol *kctl, int activate) -{ - if (activate) - kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; - else - kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE | - SNDRV_CTL_EVENT_MASK_INFO, &(kctl->id)); -} - /* open callback */ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) { @@ -242,14 +230,12 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, if (stream != NULL) vortex_adb_allocroute(chip, stream->dma, stream->nr_ch, stream->dir, - stream->type, - substream->number); + stream->type); /* Alloc routes. */ dma = vortex_adb_allocroute(chip, -1, params_channels(hw_params), - substream->stream, type, - substream->number); + substream->stream, type); if (dma < 0) { spin_unlock_irq(&chip->lock); return dma; @@ -260,11 +246,6 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, vortex_adbdma_setbuffers(chip, dma, params_period_bytes(hw_params), params_periods(hw_params)); - if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { - chip->pcm_vol[substream->number].active = 1; - vortex_notify_pcm_vol_change(chip->card, - chip->pcm_vol[substream->number].kctl, 1); - } } #ifndef CHIP_AU8810 else { @@ -294,18 +275,10 @@ static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream) spin_lock_irq(&chip->lock); // Delete audio routes. if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { - if (stream != NULL) { - if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { - chip->pcm_vol[substream->number].active = 0; - vortex_notify_pcm_vol_change(chip->card, - chip->pcm_vol[substream->number].kctl, - 0); - } + if (stream != NULL) vortex_adb_allocroute(chip, stream->dma, stream->nr_ch, stream->dir, - stream->type, - substream->number); - } + stream->type); } #ifndef CHIP_AU8810 else { @@ -533,83 +506,6 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = { }, }; -/* subdevice PCM Volume control */ - -static int snd_vortex_pcm_vol_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - vortex_t *vortex = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = (VORTEX_IS_QUAD(vortex) ? 4 : 2); - uinfo->value.integer.min = -128; - uinfo->value.integer.max = 32; - return 0; -} - -static int snd_vortex_pcm_vol_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int i; - vortex_t *vortex = snd_kcontrol_chip(kcontrol); - int subdev = kcontrol->id.subdevice; - struct pcm_vol *p = &vortex->pcm_vol[subdev]; - int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2); - for (i = 0; i < max_chn; i++) - ucontrol->value.integer.value[i] = p->vol[i]; - return 0; -} - -static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int i; - int changed = 0; - int mixin; - unsigned char vol; - vortex_t *vortex = snd_kcontrol_chip(kcontrol); - int subdev = kcontrol->id.subdevice; - struct pcm_vol *p = &vortex->pcm_vol[subdev]; - int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2); - for (i = 0; i < max_chn; i++) { - if (p->vol[i] != ucontrol->value.integer.value[i]) { - p->vol[i] = ucontrol->value.integer.value[i]; - if (p->active) { - switch (vortex->dma_adb[p->dma].nr_ch) { - case 1: - mixin = p->mixin[0]; - break; - case 2: - default: - mixin = p->mixin[(i < 2) ? i : (i - 2)]; - break; - case 4: - mixin = p->mixin[i]; - break; - }; - vol = p->vol[i]; - vortex_mix_setinputvolumebyte(vortex, - vortex->mixplayb[i], mixin, vol); - } - changed = 1; - } - } - return changed; -} - -static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400); - -static struct snd_kcontrol_new snd_vortex_pcm_vol __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "PCM Playback Volume", - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .info = snd_vortex_pcm_vol_info, - .get = snd_vortex_pcm_vol_get, - .put = snd_vortex_pcm_vol_put, - .tlv = { .p = vortex_pcm_vol_db_scale }, -}; - /* create a pcm device */ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) { @@ -659,20 +555,5 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; } } - if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_ADB) { - for (i = 0; i < NR_PCM; i++) { - chip->pcm_vol[i].active = 0; - chip->pcm_vol[i].dma = -1; - kctl = snd_ctl_new1(&snd_vortex_pcm_vol, chip); - if (!kctl) - return -ENOMEM; - chip->pcm_vol[i].kctl = kctl; - kctl->id.device = 0; - kctl->id.subdevice = i; - err = snd_ctl_add(chip->card, kctl); - if (err < 0) - return err; - } - } return 0; } diff --git a/trunk/sound/pci/azt3328.c b/trunk/sound/pci/azt3328.c index 496f14c1a731..95ffa6a9db6e 100644 --- a/trunk/sound/pci/azt3328.c +++ b/trunk/sound/pci/azt3328.c @@ -2684,9 +2684,10 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto out_err; - opl3->private_data = chip; } + opl3->private_data = chip; + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->ctrl_io, chip->irq); diff --git a/trunk/sound/pci/ctxfi/ctvmem.c b/trunk/sound/pci/ctxfi/ctvmem.c index 6109490b83e8..b78f3fc3c33c 100644 --- a/trunk/sound/pci/ctxfi/ctvmem.c +++ b/trunk/sound/pci/ctxfi/ctvmem.c @@ -36,7 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) size = CT_PAGE_ALIGN(size); if (size > vm->size) { - printk(KERN_ERR "ctxfi: Fail! No sufficient device virtual " + printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " "memory space available!\n"); return NULL; } diff --git a/trunk/sound/pci/hda/alc260_quirks.c b/trunk/sound/pci/hda/alc260_quirks.c new file mode 100644 index 000000000000..3b5170b9700f --- /dev/null +++ b/trunk/sound/pci/hda/alc260_quirks.c @@ -0,0 +1,968 @@ +/* + * ALC260 quirk models + * included by patch_realtek.c + */ + +/* ALC260 models */ +enum { + ALC260_AUTO, + ALC260_BASIC, + ALC260_FUJITSU_S702X, + ALC260_ACER, + ALC260_WILL, + ALC260_REPLACER_672V, + ALC260_FAVORIT100, +#ifdef CONFIG_SND_DEBUG + ALC260_TEST, +#endif + ALC260_MODEL_LAST /* last tag */ +}; + +static const hda_nid_t alc260_dac_nids[1] = { + /* front */ + 0x02, +}; + +static const hda_nid_t alc260_adc_nids[1] = { + /* ADC0 */ + 0x04, +}; + +static const hda_nid_t alc260_adc_nids_alt[1] = { + /* ADC1 */ + 0x05, +}; + +/* NIDs used when simultaneous access to both ADCs makes sense. Note that + * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. + */ +static const hda_nid_t alc260_dual_adc_nids[2] = { + /* ADC0, ADC1 */ + 0x04, 0x05 +}; + +#define ALC260_DIGOUT_NID 0x03 +#define ALC260_DIGIN_NID 0x06 + +static const struct hda_input_mux alc260_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, + * headphone jack and the internal CD lines since these are the only pins at + * which audio can appear. For flexibility, also allow the option of + * recording the mixer output on the second ADC (ADC0 doesn't have a + * connection to the mixer output). + */ +static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = { + { + .num_items = 3, + .items = { + { "Mic/Line", 0x0 }, + { "CD", 0x4 }, + { "Headphone", 0x2 }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic/Line", 0x0 }, + { "CD", 0x4 }, + { "Headphone", 0x2 }, + { "Mixer", 0x5 }, + }, + }, + +}; + +/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to + * the Fujitsu S702x, but jacks are marked differently. + */ +static const struct hda_input_mux alc260_acer_capture_sources[2] = { + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Headphone", 0x5 }, + }, + }, + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Headphone", 0x6 }, + { "Mixer", 0x5 }, + }, + }, +}; + +/* Maxdata Favorit 100XS */ +static const struct hda_input_mux alc260_favorit100_capture_sources[2] = { + { + .num_items = 2, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + }, + }, + { + .num_items = 3, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + { "Mixer", 0x5 }, + }, + }, +}; + +/* + * This is just place-holder, so there's something for alc_build_pcms to look + * at when it calculates the maximum number of channels. ALC260 has no mixer + * element which allows changing the channel mode, so the verb list is + * never used. + */ +static const struct hda_channel_mode alc260_modes[1] = { + { 2, NULL }, +}; + + +/* Mixer combinations + * + * basic: base_output + input + pc_beep + capture + * fujitsu: fujitsu + capture + * acer: acer + capture + */ + +static const struct snd_kcontrol_new alc260_base_output_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc260_input_mixer[] = { + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), + { } /* end */ +}; + +/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, + * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. + */ +static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), + { } /* end */ +}; + +/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current + * versions of the ALC260 don't act on requests to enable mic bias from NID + * 0x0f (used to drive the headphone jack in these laptops). The ALC260 + * datasheet doesn't mention this restriction. At this stage it's not clear + * whether this behaviour is intentional or is a hardware bug in chip + * revisions available in early 2006. Therefore for now allow the + * "Headphone Jack Mode" control to span all choices, but if it turns out + * that the lack of mic bias for this NID is intentional we could change the + * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. + * + * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006 + * don't appear to make the mic bias available from the "line" jack, even + * though the NID used for this jack (0x14) can supply it. The theory is + * that perhaps Acer have included blocking capacitors between the ALC260 + * and the output jack. If this turns out to be the case for all such + * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT + * to ALC_PIN_DIR_INOUT_NOMICBIAS. + * + * The C20x Tablet series have a mono internal speaker which is controlled + * via the chip's Mono sum widget and pin complex, so include the necessary + * controls for such models. On models without a "mono speaker" the control + * won't do anything. + */ +static const struct snd_kcontrol_new alc260_acer_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, + HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + { } /* end */ +}; + +/* Maxdata Favorit 100XS: one output and one input (0x12) jack + */ +static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + { } /* end */ +}; + +/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12, + * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17. + */ +static const struct snd_kcontrol_new alc260_will_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + { } /* end */ +}; + +/* Replacer 672V ALC260 pin usage: Mic jack = 0x12, + * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f. + */ +static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + { } /* end */ +}; + +/* + * initialization verbs + */ +static const struct hda_verb alc260_init_verbs[] = { + /* Line In pin widget for input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* CD pin widget for input */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* LINE-2 is used for line-out in rear */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* select line-out */ + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LINE-OUT pin */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* enable HP */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* enable Mono */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* set connection select to line in (default select for this ADC) */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* mute capture amp left and right */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* set connection select to line in (default select for this ADC) */ + {0x05, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* set vol=0 Line-Out mixer amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* unmute pin widget amp left and right (no gain on this amp) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* set vol=0 HP mixer amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* unmute pin widget amp left and right (no gain on this amp) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* set vol=0 Mono mixer amp left and right */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* unmute pin widget amp left and right (no gain on this amp) */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* unmute LINE-2 out pin */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & + * Line In 2 = 0x03 + */ + /* mute analog inputs */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ + /* mute Front out path */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* mute Headphone out path */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* mute Mono out path */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } +}; + +/* Initialisation sequence for ALC260 as configured in Fujitsu S702x + * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD + * audio = 0x16, internal speaker = 0x10. + */ +static const struct hda_verb alc260_fujitsu_init_verbs[] = { + /* Disable all GPIOs */ + {0x01, AC_VERB_SET_GPIO_MASK, 0}, + /* Internal speaker is connected to headphone pin */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Headphone/Line-out jack connects to Line1 pin; make it an output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + * when acting as an output. + */ + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Line1 pin widget output buffer since it starts as an output. + * If the pin mode is changed by the user the pin mode control will + * take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute input buffer of pin widget used for Line-in (no equiv + * mixer ctrl) + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - line + * in (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to line in (on mic1 pin) + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + +/* Initialisation sequence for ALC260 as configured in Acer TravelMate and + * similar laptops (adapted from Fujitsu init verbs). + */ +static const struct hda_verb alc260_acer_init_verbs[] = { + /* On TravelMate laptops, GPIO 0 enables the internal speaker and + * the headphone jack. Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Internal speaker/Headphone jack is connected to Line-out pin */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Internal microphone/Mic jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Line In jack is connected to Line1 pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right + * (no equiv mixer ctrl) + */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute mono pin widget amp output (no equiv mixer ctrl) */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to mic to match ALSA's default state. + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + +/* Initialisation sequence for Maxdata Favorit 100XS + * (adapted from Acer init verbs). + */ +static const struct hda_verb alc260_favorit100_init_verbs[] = { + /* GPIO 0 enables the output jack. + * Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Line/Mic input jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right + * (no equiv mixer ctrl) + */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to mic to match ALSA's default state. + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + +static const struct hda_verb alc260_will_verbs[] = { + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x1a, AC_VERB_SET_PROC_COEF, 0x3040}, + {} +}; + +static const struct hda_verb alc260_replacer_672v_verbs[] = { + {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x1a, AC_VERB_SET_PROC_COEF, 0x3050}, + + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + + {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {} +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc260_replacer_672v_automute(struct hda_codec *codec) +{ + unsigned int present; + + /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */ + present = snd_hda_jack_detect(codec, 0x0f); + if (present) { + snd_hda_codec_write_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 1); + snd_hda_codec_write_cache(codec, 0x0f, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_HP); + } else { + snd_hda_codec_write_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0); + snd_hda_codec_write_cache(codec, 0x0f, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_OUT); + } +} + +static void alc260_replacer_672v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC_HP_EVENT) + alc260_replacer_672v_automute(codec); +} + +static const struct hda_verb alc260_hp_dc7600_verbs[] = { + {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {} +}; + +/* Test configuration for debugging, modelled after the ALC880 test + * configuration. + */ +#ifdef CONFIG_SND_DEBUG +static const hda_nid_t alc260_test_dac_nids[1] = { + 0x02, +}; +static const hda_nid_t alc260_test_adc_nids[2] = { + 0x04, 0x05, +}; +/* For testing the ALC260, each input MUX needs its own definition since + * the signal assignments are different. This assumes that the first ADC + * is NID 0x04. + */ +static const struct hda_input_mux alc260_test_capture_sources[2] = { + { + .num_items = 7, + .items = { + { "MIC1 pin", 0x0 }, + { "MIC2 pin", 0x1 }, + { "LINE1 pin", 0x2 }, + { "LINE2 pin", 0x3 }, + { "CD pin", 0x4 }, + { "LINE-OUT pin", 0x5 }, + { "HP-OUT pin", 0x6 }, + }, + }, + { + .num_items = 8, + .items = { + { "MIC1 pin", 0x0 }, + { "MIC2 pin", 0x1 }, + { "LINE1 pin", 0x2 }, + { "LINE2 pin", 0x3 }, + { "CD pin", 0x4 }, + { "Mixer", 0x5 }, + { "LINE-OUT pin", 0x6 }, + { "HP-OUT pin", 0x7 }, + }, + }, +}; +static const struct snd_kcontrol_new alc260_test_mixer[] = { + /* Output driver widgets */ + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT), + + /* Modes for retasking pin widgets + * Note: the ALC260 doesn't seem to act on requests to enable mic + * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't + * mention this restriction. At this stage it's not clear whether + * this behaviour is intentional or is a hardware bug in chip + * revisions available at least up until early 2006. Therefore for + * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all + * choices, but if it turns out that the lack of mic bias for these + * NIDs is intentional we could change their modes from + * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. + */ + ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT), + + /* Loopback mixer controls */ + HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), + + /* Controls for GPIO pins, assuming they are configured as outputs */ + ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), + ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), + ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), + ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), + + /* Switches to allow the digital IO pins to be enabled. The datasheet + * is ambigious as to which NID is which; testing on laptops which + * make this output available should provide clarification. + */ + ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), + ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), + + /* A switch allowing EAPD to be enabled. Some laptops seem to use + * this output to turn on an external amplifier. + */ + ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02), + ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02), + + { } /* end */ +}; +static const struct hda_verb alc260_test_init_verbs[] = { + /* Enable all GPIOs as outputs with an initial value of 0 */ + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + {0x01, AC_VERB_SET_GPIO_MASK, 0x0f}, + + /* Enable retasking pins as output, initially without power amp */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* Disable digital (SPDIF) pins initially, but users can enable + * them via a mixer switch. In the case of SPDIF-out, this initverb + * payload also sets the generation to 0, output to be in "consumer" + * PCM format, copyright asserted, no pre-emphasis and no validity + * control. + */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the + * OUT1 sum bus when acting as an output. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute retasking pin widget output buffers since the default + * state appears to be output. As the pin mode is changed by the + * user the pin mode control will take care of enabling the pin's + * input/output buffers as needed. + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Also unmute the mono-out pin widget */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting (mic1 + * pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to mic1 pin + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; +#endif + +/* + * ALC260 configurations + */ +static const char * const alc260_models[ALC260_MODEL_LAST] = { + [ALC260_BASIC] = "basic", + [ALC260_FUJITSU_S702X] = "fujitsu", + [ALC260_ACER] = "acer", + [ALC260_WILL] = "will", + [ALC260_REPLACER_672V] = "replacer", + [ALC260_FAVORIT100] = "favorit100", +#ifdef CONFIG_SND_DEBUG + [ALC260_TEST] = "test", +#endif + [ALC260_AUTO] = "auto", +}; + +static const struct snd_pci_quirk alc260_cfg_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), + SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL), + SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), + SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), + SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), + SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), + SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), + SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), + SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V), + SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL), + {} +}; + +static const struct alc_config_preset alc260_presets[] = { + [ALC260_BASIC] = { + .mixers = { alc260_base_output_mixer, + alc260_input_mixer }, + .init_verbs = { alc260_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + }, + [ALC260_FUJITSU_S702X] = { + .mixers = { alc260_fujitsu_mixer }, + .init_verbs = { alc260_fujitsu_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), + .input_mux = alc260_fujitsu_capture_sources, + }, + [ALC260_ACER] = { + .mixers = { alc260_acer_mixer }, + .init_verbs = { alc260_acer_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), + .input_mux = alc260_acer_capture_sources, + }, + [ALC260_FAVORIT100] = { + .mixers = { alc260_favorit100_mixer }, + .init_verbs = { alc260_favorit100_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), + .input_mux = alc260_favorit100_capture_sources, + }, + [ALC260_WILL] = { + .mixers = { alc260_will_mixer }, + .init_verbs = { alc260_init_verbs, alc260_will_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), + .adc_nids = alc260_adc_nids, + .dig_out_nid = ALC260_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + }, + [ALC260_REPLACER_672V] = { + .mixers = { alc260_replacer_672v_mixer }, + .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), + .adc_nids = alc260_adc_nids, + .dig_out_nid = ALC260_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + .unsol_event = alc260_replacer_672v_unsol_event, + .init_hook = alc260_replacer_672v_automute, + }, +#ifdef CONFIG_SND_DEBUG + [ALC260_TEST] = { + .mixers = { alc260_test_mixer }, + .init_verbs = { alc260_test_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), + .dac_nids = alc260_test_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids), + .adc_nids = alc260_test_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources), + .input_mux = alc260_test_capture_sources, + }, +#endif +}; + diff --git a/trunk/sound/pci/hda/alc880_quirks.c b/trunk/sound/pci/hda/alc880_quirks.c new file mode 100644 index 000000000000..501501ef36a9 --- /dev/null +++ b/trunk/sound/pci/hda/alc880_quirks.c @@ -0,0 +1,1707 @@ +/* + * ALC880 quirk models + * included by patch_realtek.c + */ + +/* ALC880 board config type */ +enum { + ALC880_AUTO, + ALC880_3ST, + ALC880_3ST_DIG, + ALC880_5ST, + ALC880_5ST_DIG, + ALC880_W810, + ALC880_Z71V, + ALC880_6ST, + ALC880_6ST_DIG, + ALC880_F1734, + ALC880_ASUS, + ALC880_ASUS_DIG, + ALC880_ASUS_W1V, + ALC880_ASUS_DIG2, + ALC880_FUJITSU, + ALC880_UNIWILL_DIG, + ALC880_UNIWILL, + ALC880_UNIWILL_P53, + ALC880_CLEVO, + ALC880_TCL_S700, + ALC880_LG, +#ifdef CONFIG_SND_DEBUG + ALC880_TEST, +#endif + ALC880_MODEL_LAST /* last tag */ +}; + +/* + * ALC880 3-stack model + * + * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e) + * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, + * F-Mic = 0x1b, HP = 0x19 + */ + +static const hda_nid_t alc880_dac_nids[4] = { + /* front, rear, clfe, rear_surr */ + 0x02, 0x05, 0x04, 0x03 +}; + +static const hda_nid_t alc880_adc_nids[3] = { + /* ADC0-2 */ + 0x07, 0x08, 0x09, +}; + +/* The datasheet says the node 0x07 is connected from inputs, + * but it shows zero connection in the real implementation on some devices. + * Note: this is a 915GAV bug, fixed on 915GLV + */ +static const hda_nid_t alc880_adc_nids_alt[2] = { + /* ADC1-2 */ + 0x08, 0x09, +}; + +#define ALC880_DIGOUT_NID 0x06 +#define ALC880_DIGIN_NID 0x0a +#define ALC880_PIN_CD_NID 0x1c + +static const struct hda_input_mux alc880_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x3 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +/* channel source setting (2/6 channel selection for 3-stack) */ +/* 2ch mode */ +static const struct hda_verb alc880_threestack_ch2_init[] = { + /* set line-in to input, mute it */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + /* set mic-in to input vref 80%, mute it */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* 6ch mode */ +static const struct hda_verb alc880_threestack_ch6_init[] = { + /* set line-in to output, unmute it */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + /* set mic-in to output, unmute it */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } /* end */ +}; + +static const struct hda_channel_mode alc880_threestack_modes[2] = { + { 2, alc880_threestack_ch2_init }, + { 6, alc880_threestack_ch6_init }, +}; + +static const struct snd_kcontrol_new alc880_three_stack_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +/* + * ALC880 5-stack model + * + * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), + * Side = 0x02 (0xd) + * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16 + * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19 + */ + +/* additional mixers to alc880_three_stack_mixer */ +static const struct snd_kcontrol_new alc880_five_stack_mixer[] = { + HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT), + { } /* end */ +}; + +/* channel source setting (6/8 channel selection for 5-stack) */ +/* 6ch mode */ +static const struct hda_verb alc880_fivestack_ch6_init[] = { + /* set line-in to input, mute it */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* 8ch mode */ +static const struct hda_verb alc880_fivestack_ch8_init[] = { + /* set line-in to output, unmute it */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } /* end */ +}; + +static const struct hda_channel_mode alc880_fivestack_modes[2] = { + { 6, alc880_fivestack_ch6_init }, + { 8, alc880_fivestack_ch8_init }, +}; + + +/* + * ALC880 6-stack model + * + * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), + * Side = 0x05 (0x0f) + * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17, + * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b + */ + +static const hda_nid_t alc880_6st_dac_nids[4] = { + /* front, rear, clfe, rear_surr */ + 0x02, 0x03, 0x04, 0x05 +}; + +static const struct hda_input_mux alc880_6stack_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +/* fixed 8-channels */ +static const struct hda_channel_mode alc880_sixstack_modes[1] = { + { 8, NULL }, +}; + +static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + + +/* + * ALC880 W810 model + * + * W810 has rear IO for: + * Front (DAC 02) + * Surround (DAC 03) + * Center/LFE (DAC 04) + * Digital out (06) + * + * The system also has a pair of internal speakers, and a headphone jack. + * These are both connected to Line2 on the codec, hence to DAC 02. + * + * There is a variable resistor to control the speaker or headphone + * volume. This is a hardware-only device without a software API. + * + * Plugging headphones in will disable the internal speakers. This is + * implemented in hardware, not via the driver using jack sense. In + * a similar fashion, plugging into the rear socket marked "front" will + * disable both the speakers and headphones. + * + * For input, there's a microphone jack, and an "audio in" jack. + * These may not do anything useful with this driver yet, because I + * haven't setup any initialization verbs for these yet... + */ + +static const hda_nid_t alc880_w810_dac_nids[3] = { + /* front, rear/surround, clfe */ + 0x02, 0x03, 0x04 +}; + +/* fixed 6 channels */ +static const struct hda_channel_mode alc880_w810_modes[1] = { + { 6, NULL } +}; + +/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */ +static const struct snd_kcontrol_new alc880_w810_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + { } /* end */ +}; + + +/* + * Z710V model + * + * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d) + * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), + * Line = 0x1a + */ + +static const hda_nid_t alc880_z71v_dac_nids[1] = { + 0x02 +}; +#define ALC880_Z71V_HP_DAC 0x03 + +/* fixed 2 channels */ +static const struct hda_channel_mode alc880_2_jack_modes[1] = { + { 2, NULL } +}; + +static const struct snd_kcontrol_new alc880_z71v_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + +/* + * ALC880 F1734 model + * + * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d) + * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18 + */ + +static const hda_nid_t alc880_f1734_dac_nids[1] = { + 0x03 +}; +#define ALC880_F1734_HP_DAC 0x02 + +static const struct snd_kcontrol_new alc880_f1734_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static const struct hda_input_mux alc880_f1734_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "CD", 0x4 }, + }, +}; + + +/* + * ALC880 ASUS model + * + * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e) + * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16, + * Mic = 0x18, Line = 0x1a + */ + +#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */ +#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */ + +static const struct snd_kcontrol_new alc880_asus_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +/* + * ALC880 ASUS W1V model + * + * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e) + * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16, + * Mic = 0x18, Line = 0x1a, Line2 = 0x1b + */ + +/* additional mixers to alc880_asus_mixer */ +static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { + HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT), + { } /* end */ +}; + +/* TCL S700 */ +static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* Uniwill */ +static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* + * initialize the codec volumes, etc + */ + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static const struct hda_verb alc880_volume_init_verbs[] = { + /* + * Unmute ADC0-2 and set the default input to mic-in + */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for front + * panel mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + + /* + * Set up output mixers (0x0c - 0x0f) + */ + /* set vol=0 to output mixers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + + { } +}; + +/* + * 3-stack pin configuration: + * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b + */ +static const struct hda_verb alc880_pin_3stack_init_verbs[] = { + /* + * preset connection lists of input pins + * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround + */ + {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ + + /* + * Set pin mode and muting + */ + /* set front pin widgets 0x14 for output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mic2 (as headphone out) for HP output */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line2 (as front mic) pin widget for input and vref at 80% */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + { } +}; + +/* + * 5-stack pin configuration: + * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19, + * line-in/side = 0x1a, f-mic = 0x1b + */ +static const struct hda_verb alc880_pin_5stack_init_verbs[] = { + /* + * preset connection lists of input pins + * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround + */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */ + + /* + * Set pin mode and muting + */ + /* set pin widgets 0x14-0x17 for output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* unmute pins for output (no gain on this amp) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mic2 (as headphone out) for HP output */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line2 (as front mic) pin widget for input and vref at 80% */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + { } +}; + +/* + * W810 pin configuration: + * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b + */ +static const struct hda_verb alc880_pin_w810_init_verbs[] = { + /* hphone/speaker input selector: front DAC */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } +}; + +/* + * Z71V pin configuration: + * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?) + */ +static const struct hda_verb alc880_pin_z71v_init_verbs[] = { + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + { } +}; + +/* + * 6-stack pin configuration: + * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, + * f-mic = 0x19, line = 0x1a, HP = 0x1b + */ +static const struct hda_verb alc880_pin_6stack_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + { } +}; + +/* + * Uniwill pin configuration: + * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19, + * line = 0x1a + */ +static const struct hda_verb alc880_uniwill_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */ + /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, + + { } +}; + +/* +* Uniwill P53 +* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, + */ +static const struct hda_verb alc880_uniwill_p53_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_DCVOL_EVENT}, + + { } +}; + +static const struct hda_verb alc880_beep_init_verbs[] = { + { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, + { } +}; + +static void alc880_uniwill_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x16; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + +static void alc880_uniwill_init_hook(struct hda_codec *codec) +{ + alc_hp_automute(codec); + alc88x_simple_mic_automute(codec); +} + +static void alc880_uniwill_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + res >>= 28; + switch (res) { + case ALC_MIC_EVENT: + alc88x_simple_mic_automute(codec); + break; + default: + alc_exec_unsol_event(codec, res); + break; + } +} + +static void alc880_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 28); +} + +static void alc880_uniwill_p53_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + +static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); + present &= HDA_AMP_VOLMASK; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, + HDA_AMP_VOLMASK, present); + snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0, + HDA_AMP_VOLMASK, present); +} + +static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + res >>= 28; + if (res == ALC_DCVOL_EVENT) + alc880_uniwill_p53_dcvol_automute(codec); + else + alc_exec_unsol_event(codec, res); +} + +/* + * F1734 pin configuration: + * HP = 0x14, speaker-out = 0x15, mic = 0x18 + */ +static const struct hda_verb alc880_pin_f1734_init_verbs[] = { + {0x07, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_DCVOL_EVENT}, + + { } +}; + +/* + * ASUS pin configuration: + * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a + */ +static const struct hda_verb alc880_pin_asus_init_verbs[] = { + {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + { } +}; + +/* Enable GPIO mask and set output */ +#define alc880_gpio1_init_verbs alc_gpio1_init_verbs +#define alc880_gpio2_init_verbs alc_gpio2_init_verbs +#define alc880_gpio3_init_verbs alc_gpio3_init_verbs + +/* Clevo m520g init */ +static const struct hda_verb alc880_pin_clevo_init_verbs[] = { + /* headphone output */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* line-out */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Line-in */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* CD */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Mic1 (rear panel) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Mic2 (front panel) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* headphone */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + + { } +}; + +static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + + /* Headphone output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Front output*/ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, + + { } +}; + +/* + * LG m1 express dual + * + * Pin assignment: + * Rear Line-In/Out (blue): 0x14 + * Build-in Mic-In: 0x15 + * Speaker-out: 0x17 + * HP-Out (green): 0x1b + * Mic-In/Out (red): 0x19 + * SPDIF-Out: 0x1e + */ + +/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ +static const hda_nid_t alc880_lg_dac_nids[3] = { + 0x05, 0x02, 0x03 +}; + +/* seems analog CD is not working */ +static const struct hda_input_mux alc880_lg_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x5 }, + { "Internal Mic", 0x6 }, + }, +}; + +/* 2,4,6 channel modes */ +static const struct hda_verb alc880_lg_ch2_init[] = { + /* set line-in and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static const struct hda_verb alc880_lg_ch4_init[] = { + /* set line-in to out and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static const struct hda_verb alc880_lg_ch6_init[] = { + /* set line-in and mic-in to output */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { } +}; + +static const struct hda_channel_mode alc880_lg_ch_modes[3] = { + { 2, alc880_lg_ch2_init }, + { 4, alc880_lg_ch4_init }, + { 6, alc880_lg_ch6_init }, +}; + +static const struct snd_kcontrol_new alc880_lg_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static const struct hda_verb alc880_lg_init_verbs[] = { + /* set capture source to mic-in */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* mute all amp mixer inputs */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* line-in to input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* built-in mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* speaker-out */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* mic-in to input */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* HP-out */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* jack sense */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc880_lg_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x17; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static const struct hda_amp_list alc880_lg_loopbacks[] = { + { 0x0b, HDA_INPUT, 1 }, + { 0x0b, HDA_INPUT, 6 }, + { 0x0b, HDA_INPUT, 7 }, + { } /* end */ +}; +#endif + +/* + * Test configuration for debugging + * + * Almost all inputs/outputs are enabled. I/O pins can be configured via + * enum controls. + */ +#ifdef CONFIG_SND_DEBUG +static const hda_nid_t alc880_test_dac_nids[4] = { + 0x02, 0x03, 0x04, 0x05 +}; + +static const struct hda_input_mux alc880_test_capture_source = { + .num_items = 7, + .items = { + { "In-1", 0x0 }, + { "In-2", 0x1 }, + { "In-3", 0x2 }, + { "In-4", 0x3 }, + { "CD", 0x4 }, + { "Front", 0x5 }, + { "Surround", 0x6 }, + }, +}; + +static const struct hda_channel_mode alc880_test_modes[4] = { + { 2, NULL }, + { 4, NULL }, + { 6, NULL }, + { 8, NULL }, +}; + +static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "N/A", "Line Out", "HP Out", + "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 8; + if (uinfo->value.enumerated.item >= 8) + uinfo->value.enumerated.item = 7; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = (hda_nid_t)kcontrol->private_value; + unsigned int pin_ctl, item = 0; + + pin_ctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (pin_ctl & AC_PINCTL_OUT_EN) { + if (pin_ctl & AC_PINCTL_HP_EN) + item = 2; + else + item = 1; + } else if (pin_ctl & AC_PINCTL_IN_EN) { + switch (pin_ctl & AC_PINCTL_VREFEN) { + case AC_PINCTL_VREF_HIZ: item = 3; break; + case AC_PINCTL_VREF_50: item = 4; break; + case AC_PINCTL_VREF_GRD: item = 5; break; + case AC_PINCTL_VREF_80: item = 6; break; + case AC_PINCTL_VREF_100: item = 7; break; + } + } + ucontrol->value.enumerated.item[0] = item; + return 0; +} + +static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = (hda_nid_t)kcontrol->private_value; + static const unsigned int ctls[] = { + 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_50, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_80, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_100, + }; + unsigned int old_ctl, new_ctl; + + old_ctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + new_ctl = ctls[ucontrol->value.enumerated.item[0]]; + if (old_ctl != new_ctl) { + int val; + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + new_ctl); + val = ucontrol->value.enumerated.item[0] >= 3 ? + HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, val); + return 1; + } + return 0; +} + +static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "Front", "Surround", "CLFE", "Side" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item >= 4) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = (hda_nid_t)kcontrol->private_value; + unsigned int sel; + + sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + ucontrol->value.enumerated.item[0] = sel & 3; + return 0; +} + +static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = (hda_nid_t)kcontrol->private_value; + unsigned int sel; + + sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3; + if (ucontrol->value.enumerated.item[0] != sel) { + sel = ucontrol->value.enumerated.item[0] & 3; + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, sel); + return 1; + } + return 0; +} + +#define PIN_CTL_TEST(xname,nid) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ + .info = alc_test_pin_ctl_info, \ + .get = alc_test_pin_ctl_get, \ + .put = alc_test_pin_ctl_put, \ + .private_value = nid \ + } + +#define PIN_SRC_TEST(xname,nid) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ + .info = alc_test_pin_src_info, \ + .get = alc_test_pin_src_get, \ + .put = alc_test_pin_src_put, \ + .private_value = nid \ + } + +static const struct snd_kcontrol_new alc880_test_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + PIN_CTL_TEST("Front Pin Mode", 0x14), + PIN_CTL_TEST("Surround Pin Mode", 0x15), + PIN_CTL_TEST("CLFE Pin Mode", 0x16), + PIN_CTL_TEST("Side Pin Mode", 0x17), + PIN_CTL_TEST("In-1 Pin Mode", 0x18), + PIN_CTL_TEST("In-2 Pin Mode", 0x19), + PIN_CTL_TEST("In-3 Pin Mode", 0x1a), + PIN_CTL_TEST("In-4 Pin Mode", 0x1b), + PIN_SRC_TEST("In-1 Pin Source", 0x18), + PIN_SRC_TEST("In-2 Pin Source", 0x19), + PIN_SRC_TEST("In-3 Pin Source", 0x1a), + PIN_SRC_TEST("In-4 Pin Source", 0x1b), + HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static const struct hda_verb alc880_test_init_verbs[] = { + /* Unmute inputs of 0x0c - 0x0f */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Vol output for 0x0c-0x0f */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Set output pins 0x14-0x17 */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Unmute output pins 0x14-0x17 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Set input pins 0x18-0x1c */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Mute input pins 0x18-0x1b */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* ADC set up */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Analog input/passthru */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; +#endif + +/* + */ + +static const char * const alc880_models[ALC880_MODEL_LAST] = { + [ALC880_3ST] = "3stack", + [ALC880_TCL_S700] = "tcl", + [ALC880_3ST_DIG] = "3stack-digout", + [ALC880_CLEVO] = "clevo", + [ALC880_5ST] = "5stack", + [ALC880_5ST_DIG] = "5stack-digout", + [ALC880_W810] = "w810", + [ALC880_Z71V] = "z71v", + [ALC880_6ST] = "6stack", + [ALC880_6ST_DIG] = "6stack-digout", + [ALC880_ASUS] = "asus", + [ALC880_ASUS_W1V] = "asus-w1v", + [ALC880_ASUS_DIG] = "asus-dig", + [ALC880_ASUS_DIG2] = "asus-dig2", + [ALC880_UNIWILL_DIG] = "uniwill", + [ALC880_UNIWILL_P53] = "uniwill-p53", + [ALC880_FUJITSU] = "fujitsu", + [ALC880_F1734] = "F1734", + [ALC880_LG] = "lg", +#ifdef CONFIG_SND_DEBUG + [ALC880_TEST] = "test", +#endif + [ALC880_AUTO] = "auto", +}; + +static const struct snd_pci_quirk alc880_cfg_tbl[] = { + SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810), + SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), + SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), + SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), + SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V), + /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ + SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), + SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), + SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), + SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), + SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), + SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST), + SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST), + SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO), + SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO), + SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), + SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), + SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734), + SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), + SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), + SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), + SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), + SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), + SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG), + SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG), + SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), + SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ + SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), + SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), + SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), + SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), + {} +}; + +/* + * ALC880 codec presets + */ +static const struct alc_config_preset alc880_presets[] = { + [ALC880_3ST] = { + .mixers = { alc880_three_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_3ST_DIG] = { + .mixers = { alc880_three_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_TCL_S700] = { + .mixers = { alc880_tcl_s700_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_tcl_S700_init_verbs, + alc880_gpio2_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */ + .num_adc_nids = 1, /* single ADC */ + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_capture_source, + }, + [ALC880_5ST] = { + .mixers = { alc880_three_stack_mixer, + alc880_five_stack_mixer}, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_5stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes), + .channel_mode = alc880_fivestack_modes, + .input_mux = &alc880_capture_source, + }, + [ALC880_5ST_DIG] = { + .mixers = { alc880_three_stack_mixer, + alc880_five_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_5stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes), + .channel_mode = alc880_fivestack_modes, + .input_mux = &alc880_capture_source, + }, + [ALC880_6ST] = { + .mixers = { alc880_six_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_6stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), + .dac_nids = alc880_6st_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), + .channel_mode = alc880_sixstack_modes, + .input_mux = &alc880_6stack_capture_source, + }, + [ALC880_6ST_DIG] = { + .mixers = { alc880_six_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_6stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), + .dac_nids = alc880_6st_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), + .channel_mode = alc880_sixstack_modes, + .input_mux = &alc880_6stack_capture_source, + }, + [ALC880_W810] = { + .mixers = { alc880_w810_base_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_w810_init_verbs, + alc880_gpio2_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids), + .dac_nids = alc880_w810_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_w810_modes), + .channel_mode = alc880_w810_modes, + .input_mux = &alc880_capture_source, + }, + [ALC880_Z71V] = { + .mixers = { alc880_z71v_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_z71v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids), + .dac_nids = alc880_z71v_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_capture_source, + }, + [ALC880_F1734] = { + .mixers = { alc880_f1734_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_f1734_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids), + .dac_nids = alc880_f1734_dac_nids, + .hp_nid = 0x02, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_f1734_capture_source, + .unsol_event = alc880_uniwill_p53_unsol_event, + .setup = alc880_uniwill_p53_setup, + .init_hook = alc_hp_automute, + }, + [ALC880_ASUS] = { + .mixers = { alc880_asus_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_asus_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), + .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_ASUS_DIG] = { + .mixers = { alc880_asus_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_asus_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), + .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_ASUS_DIG2] = { + .mixers = { alc880_asus_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_asus_init_verbs, + alc880_gpio2_init_verbs }, /* use GPIO2 */ + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), + .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_ASUS_W1V] = { + .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_asus_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), + .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_UNIWILL_DIG] = { + .mixers = { alc880_asus_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_asus_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), + .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_UNIWILL] = { + .mixers = { alc880_uniwill_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_uniwill_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + .unsol_event = alc880_uniwill_unsol_event, + .setup = alc880_uniwill_setup, + .init_hook = alc880_uniwill_init_hook, + }, + [ALC880_UNIWILL_P53] = { + .mixers = { alc880_uniwill_p53_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_uniwill_p53_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_w810_modes), + .channel_mode = alc880_threestack_modes, + .input_mux = &alc880_capture_source, + .unsol_event = alc880_uniwill_p53_unsol_event, + .setup = alc880_uniwill_p53_setup, + .init_hook = alc_hp_automute, + }, + [ALC880_FUJITSU] = { + .mixers = { alc880_fujitsu_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_uniwill_p53_init_verbs, + alc880_beep_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_capture_source, + .unsol_event = alc880_uniwill_p53_unsol_event, + .setup = alc880_uniwill_p53_setup, + .init_hook = alc_hp_automute, + }, + [ALC880_CLEVO] = { + .mixers = { alc880_three_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_clevo_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc880_capture_source, + }, + [ALC880_LG] = { + .mixers = { alc880_lg_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_lg_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids), + .dac_nids = alc880_lg_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), + .channel_mode = alc880_lg_ch_modes, + .need_dac_fix = 1, + .input_mux = &alc880_lg_capture_source, + .unsol_event = alc880_unsol_event, + .setup = alc880_lg_setup, + .init_hook = alc_hp_automute, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .loopbacks = alc880_lg_loopbacks, +#endif + }, +#ifdef CONFIG_SND_DEBUG + [ALC880_TEST] = { + .mixers = { alc880_test_mixer }, + .init_verbs = { alc880_test_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_test_dac_nids), + .dac_nids = alc880_test_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_test_modes), + .channel_mode = alc880_test_modes, + .input_mux = &alc880_test_capture_source, + }, +#endif +}; + diff --git a/trunk/sound/pci/hda/alc882_quirks.c b/trunk/sound/pci/hda/alc882_quirks.c new file mode 100644 index 000000000000..bb364a53f546 --- /dev/null +++ b/trunk/sound/pci/hda/alc882_quirks.c @@ -0,0 +1,866 @@ +/* + * ALC882/ALC883/ALC888/ALC889 quirk models + * included by patch_realtek.c + */ + +/* ALC882 models */ +enum { + ALC882_AUTO, + ALC885_MBA21, + ALC885_MBP3, + ALC885_MB5, + ALC885_MACMINI3, + ALC885_IMAC91, + ALC889A_MB31, + ALC882_MODEL_LAST, +}; + +#define ALC882_DIGOUT_NID 0x06 +#define ALC882_DIGIN_NID 0x0a +#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID +#define ALC883_DIGIN_NID ALC882_DIGIN_NID +#define ALC1200_DIGOUT_NID 0x10 + + +static const struct hda_channel_mode alc882_ch_modes[1] = { + { 8, NULL } +}; + +/* DACs */ +static const hda_nid_t alc882_dac_nids[4] = { + /* front, rear, clfe, rear_surr */ + 0x02, 0x03, 0x04, 0x05 +}; +#define alc883_dac_nids alc882_dac_nids + +/* ADCs */ +#define alc882_adc_nids alc880_adc_nids +#define alc882_adc_nids_alt alc880_adc_nids_alt +#define alc883_adc_nids alc882_adc_nids_alt + +static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; +#define alc883_capsrc_nids alc882_capsrc_nids_alt + +/* input MUX */ +/* FIXME: should be a matrix-type input source selection */ + +static const struct hda_input_mux alc882_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +#define alc883_capture_source alc882_capture_source + +static const struct hda_input_mux mb5_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x7 }, + { "CD", 0x4 }, + }, +}; + +static const struct hda_input_mux macmini3_capture_source = { + .num_items = 2, + .items = { + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static const struct hda_input_mux alc883_3stack_6ch_intel = { + .num_items = 4, + .items = { + { "Mic", 0x1 }, + { "Front Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static const struct hda_input_mux alc889A_mb31_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + /* Front Mic (0x01) unused */ + { "Line", 0x2 }, + /* Line 2 (0x03) unused */ + /* CD (0x04) unused? */ + }, +}; + +static const struct hda_input_mux alc889A_imac91_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x01 }, + { "Line", 0x2 }, /* Not sure! */ + }, +}; + +/* Macbook Air 2,1 */ + +static const struct hda_channel_mode alc885_mba21_ch_modes[1] = { + { 2, NULL }, +}; + +/* + * macbook pro ALC885 can switch LineIn to LineOut without losing Mic + */ + +/* + * 2ch mode + */ +static const struct hda_verb alc885_mbp_ch2_init[] = { + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +/* + * 4ch mode + */ +static const struct hda_verb alc885_mbp_ch4_init[] = { + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } /* end */ +}; + +static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = { + { 2, alc885_mbp_ch2_init }, + { 4, alc885_mbp_ch4_init }, +}; + +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static const struct hda_verb alc885_mb5_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static const struct hda_verb alc885_mb5_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; + +#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes + +/* Macbook Air 2,1 same control for HP and internal Speaker */ + +static const struct snd_kcontrol_new alc885_mba21_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), + { } +}; + + +static const struct snd_kcontrol_new alc885_mbp3_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc885_mb5_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc885_macmini3_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc885_imac91_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), + { } /* end */ +}; + + +static const struct snd_kcontrol_new alc882_chmode_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static const struct hda_verb alc882_base_init_verbs[] = { + /* Front mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* CLFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Side mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + + /* Front Pin: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Rear Pin: output 1 (0x0d) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* CLFE Pin: output 2 (0x0e) */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* Side Pin: output 3 (0x0f) */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* Mic (rear) pin: input vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin: input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line-2 In: Headphone output (output 0 - 0x0c) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* ADC2: mute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* ADC3: mute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + + { } +}; + +#define alc883_init_verbs alc882_base_init_verbs + +/* Macbook 5,1 */ +static const struct hda_verb alc885_mb5_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)}, + { } +}; + +/* Macmini 3,1 */ +static const struct hda_verb alc885_macmini3_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + + +static const struct hda_verb alc885_mba21_init_verbs[] = { + /*Internal and HP Speaker Mixer*/ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /*Internal Speaker Pin (0x0c)*/ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: output 0 (0x0e) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)}, + /* Line in (is hp when jack connected)*/ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } + }; + + +/* Macbook Pro rev3 */ +static const struct hda_verb alc885_mbp3_init_verbs[] = { + /* Front mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: output 0 (0x0e) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, + /* Mic (rear) pin: input vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin: use output 1 when in LineOut mode */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* ADC1: mute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* ADC2: mute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* ADC3: mute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + + { } +}; + +/* iMac 9,1 */ +static const struct hda_verb alc885_imac91_init_verbs[] = { + /* Internal Speaker Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: Rear */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)}, + /* Line in Rear */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* Toggle speaker-output according to the hp-jack state */ +static void alc885_imac24_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + spec->autocfg.speaker_pins[1] = 0x1a; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + +#define alc885_mb5_setup alc885_imac24_setup +#define alc885_macmini3_setup alc885_imac24_setup + +/* Macbook Air 2,1 */ +static void alc885_mba21_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + + + +static void alc885_mbp3_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + +static void alc885_imac91_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + spec->autocfg.speaker_pins[1] = 0x1a; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); +} + +/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ +static const struct hda_verb alc889A_mb31_ch2_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ + { } /* end */ +}; + +/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ +static const struct hda_verb alc889A_mb31_ch4_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ + { } /* end */ +}; + +/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ +static const struct hda_verb alc889A_mb31_ch5_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ + { } /* end */ +}; + +/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ +static const struct hda_verb alc889A_mb31_ch6_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ + { } /* end */ +}; + +static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { + { 2, alc889A_mb31_ch2_init }, + { 4, alc889A_mb31_ch4_init }, + { 5, alc889A_mb31_ch5_init }, + { 6, alc889A_mb31_ch6_init }, +}; + +static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { + /* Output mixers */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT), + /* Output switches */ + HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), + /* Boost mixers */ + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), + /* Input mixers */ + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + +static const struct snd_kcontrol_new alc883_chmode_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static const struct hda_verb alc889A_mb31_verbs[] = { + /* Init rear pin (used as headphone output) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, + /* Init line pin (used as output in 4ch and 6ch mode) */ + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */ + /* Init line 2 pin (used as headphone out by default) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */ + { } /* end */ +}; + +/* Mute speakers according to the headphone jack state */ +static void alc889A_mb31_automute(struct hda_codec *codec) +{ + unsigned int present; + + /* Mute only in 2ch or 4ch mode */ + if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) + == 0x00) { + present = snd_hda_jack_detect(codec, 0x15); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + } +} + +static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC_HP_EVENT) + alc889A_mb31_automute(codec); +} + +static void alc882_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 26); +} + +/* + * configuration and preset + */ +static const char * const alc882_models[ALC882_MODEL_LAST] = { + [ALC885_MB5] = "mb5", + [ALC885_MACMINI3] = "macmini3", + [ALC885_MBA21] = "mba21", + [ALC885_MBP3] = "mbp3", + [ALC885_IMAC91] = "imac91", + [ALC889A_MB31] = "mb31", + [ALC882_AUTO] = "auto", +}; + +/* codec SSID table for Intel Mac */ +static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21), + SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), + SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91), + SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5), + /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, + * so apparently no perfect solution yet + */ + SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), + {} /* terminator */ +}; + +static const struct alc_config_preset alc882_presets[] = { + [ALC885_MBA21] = { + .mixers = { alc885_mba21_mixer }, + .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc882_capture_source, + .unsol_event = alc882_unsol_event, + .setup = alc885_mba21_setup, + .init_hook = alc_hp_automute, + }, + [ALC885_MBP3] = { + .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_mbp3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .hp_nid = 0x04, + .channel_mode = alc885_mbp_4ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), + .input_mux = &alc882_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc882_unsol_event, + .setup = alc885_mbp3_setup, + .init_hook = alc_hp_automute, + }, + [ALC885_MB5] = { + .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_mb5_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mb5_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), + .input_mux = &mb5_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc882_unsol_event, + .setup = alc885_mb5_setup, + .init_hook = alc_hp_automute, + }, + [ALC885_MACMINI3] = { + .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_macmini3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_macmini3_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), + .input_mux = &macmini3_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc882_unsol_event, + .setup = alc885_macmini3_setup, + .init_hook = alc_hp_automute, + }, + [ALC885_IMAC91] = { + .mixers = {alc885_imac91_mixer}, + .init_verbs = { alc885_imac91_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc889A_imac91_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc882_unsol_event, + .setup = alc885_imac91_setup, + .init_hook = alc_hp_automute, + }, + [ALC889A_MB31] = { + .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, + .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, + alc880_gpio1_init_verbs }, + .adc_nids = alc883_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .capsrc_nids = alc883_capsrc_nids, + .dac_nids = alc883_dac_nids, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .channel_mode = alc889A_mb31_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes), + .input_mux = &alc889A_mb31_capture_source, + .dig_out_nid = ALC883_DIGOUT_NID, + .unsol_event = alc889A_mb31_unsol_event, + .init_hook = alc889A_mb31_automute, + }, +}; + + diff --git a/trunk/sound/pci/hda/alc_quirks.c b/trunk/sound/pci/hda/alc_quirks.c new file mode 100644 index 000000000000..a18952ed4311 --- /dev/null +++ b/trunk/sound/pci/hda/alc_quirks.c @@ -0,0 +1,480 @@ +/* + * Common codes for Realtek codec quirks + * included by patch_realtek.c + */ + +/* + * configuration template - to be copied to the spec instance + */ +struct alc_config_preset { + const struct snd_kcontrol_new *mixers[5]; /* should be identical size + * with spec + */ + const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + const struct hda_verb *init_verbs[5]; + unsigned int num_dacs; + const hda_nid_t *dac_nids; + hda_nid_t dig_out_nid; /* optional */ + hda_nid_t hp_nid; /* optional */ + const hda_nid_t *slave_dig_outs; + unsigned int num_adc_nids; + const hda_nid_t *adc_nids; + const hda_nid_t *capsrc_nids; + hda_nid_t dig_in_nid; + unsigned int num_channel_mode; + const struct hda_channel_mode *channel_mode; + int need_dac_fix; + int const_channel_count; + unsigned int num_mux_defs; + const struct hda_input_mux *input_mux; + void (*unsol_event)(struct hda_codec *, unsigned int); + void (*setup)(struct hda_codec *); + void (*init_hook)(struct hda_codec *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + const struct hda_amp_list *loopbacks; + void (*power_hook)(struct hda_codec *codec); +#endif +}; + +/* + * channel mode setting + */ +static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, + spec->num_channel_mode); +} + +static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + spec->ext_channel_count); +} + +static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + &spec->ext_channel_count); + if (err >= 0 && !spec->const_channel_count) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } + return err; +} + +/* + * Control the mode of pin widget settings via the mixer. "pc" is used + * instead of "%" to avoid consequences of accidentally treating the % as + * being part of a format specifier. Maximum allowed length of a value is + * 63 characters plus NULL terminator. + * + * Note: some retasking pin complexes seem to ignore requests for input + * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these + * are requested. Therefore order this list so that this behaviour will not + * cause problems when mixer clients move through the enum sequentially. + * NIDs 0x0f and 0x10 have been observed to have this behaviour as of + * March 2006. + */ +static const char * const alc_pin_mode_names[] = { + "Mic 50pc bias", "Mic 80pc bias", + "Line in", "Line out", "Headphone out", +}; +static const unsigned char alc_pin_mode_values[] = { + PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, +}; +/* The control can present all 5 options, or it can limit the options based + * in the pin being assumed to be exclusively an input or an output pin. In + * addition, "input" pins may or may not process the mic bias option + * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to + * accept requests for bias as of chip versions up to March 2006) and/or + * wiring in the computer. + */ +#define ALC_PIN_DIR_IN 0x00 +#define ALC_PIN_DIR_OUT 0x01 +#define ALC_PIN_DIR_INOUT 0x02 +#define ALC_PIN_DIR_IN_NOMICBIAS 0x03 +#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04 + +/* Info about the pin modes supported by the different pin direction modes. + * For each direction the minimum and maximum values are given. + */ +static const signed char alc_pin_mode_dir_info[5][2] = { + { 0, 2 }, /* ALC_PIN_DIR_IN */ + { 3, 4 }, /* ALC_PIN_DIR_OUT */ + { 0, 4 }, /* ALC_PIN_DIR_INOUT */ + { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */ + { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */ +}; +#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0]) +#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1]) +#define alc_pin_mode_n_items(_dir) \ + (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) + +static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int item_num = uinfo->value.enumerated.item; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = alc_pin_mode_n_items(dir); + + if (item_numalc_pin_mode_max(dir)) + item_num = alc_pin_mode_min(dir); + strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]); + return 0; +} + +static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int i; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int pinctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0x00); + + /* Find enumerated value for current pinctl setting */ + i = alc_pin_mode_min(dir); + while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl) + i++; + *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir); + return 0; +} + +static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int pinctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0x00); + + if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir)) + val = alc_pin_mode_min(dir); + + change = pinctl != alc_pin_mode_values[val]; + if (change) { + /* Set pin mode to that requested */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + alc_pin_mode_values[val]); + + /* Also enable the retasking pin's input/output as required + * for the requested pin mode. Enum values of 2 or less are + * input modes. + * + * Dynamically switching the input/output buffers probably + * reduces noise slightly (particularly on input) so we'll + * do it. However, having both input and output buffers + * enabled simultaneously doesn't seem to be problematic if + * this turns out to be necessary in the future. + */ + if (val <= 2) { + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, + HDA_AMP_MUTE, 0); + } else { + snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, 0); + } + } + return change; +} + +#define ALC_PIN_MODE(xname, nid, dir) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ + .info = alc_pin_mode_info, \ + .get = alc_pin_mode_get, \ + .put = alc_pin_mode_put, \ + .private_value = nid | (dir<<16) } + +/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged + * together using a mask with more than one bit set. This control is + * currently used only by the ALC260 test model. At this stage they are not + * needed for any "production" models. + */ +#ifdef CONFIG_SND_DEBUG +#define alc_gpio_data_info snd_ctl_boolean_mono_info + +static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DATA, 0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DATA, + 0x00); + + /* Set/unset the masked GPIO bit(s) as needed */ + change = (val == 0 ? 0 : mask) != (gpio_data & mask); + if (val == 0) + gpio_data &= ~mask; + else + gpio_data |= mask; + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_GPIO_DATA, gpio_data); + + return change; +} +#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ + .info = alc_gpio_data_info, \ + .get = alc_gpio_data_get, \ + .put = alc_gpio_data_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + +/* A switch control to allow the enabling of the digital IO pins on the + * ALC260. This is incredibly simplistic; the intention of this control is + * to provide something in the test model allowing digital outputs to be + * identified if present. If models are found which can utilise these + * outputs a more complete mixer control can be devised for those models if + * necessary. + */ +#ifdef CONFIG_SND_DEBUG +#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info + +static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, + 0x00); + + /* Set/unset the masked control bit(s) as needed */ + change = (val == 0 ? 0 : mask) != (ctrl_data & mask); + if (val==0) + ctrl_data &= ~mask; + else + ctrl_data |= mask; + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + ctrl_data); + + return change; +} +#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ + .info = alc_spdif_ctrl_info, \ + .get = alc_spdif_ctrl_get, \ + .put = alc_spdif_ctrl_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + +/* A switch control to allow the enabling EAPD digital outputs on the ALC26x. + * Again, this is only used in the ALC26x test models to help identify when + * the EAPD line must be asserted for features to work. + */ +#ifdef CONFIG_SND_DEBUG +#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info + +static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, 0x00); + + *valp = (val & mask) != 0; + return 0; +} + +static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_EAPD_BTLENABLE, + 0x00); + + /* Set/unset the masked control bit(s) as needed */ + change = (!val ? 0 : mask) != (ctrl_data & mask); + if (!val) + ctrl_data &= ~mask; + else + ctrl_data |= mask; + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, + ctrl_data); + + return change; +} + +#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ + .info = alc_eapd_ctrl_info, \ + .get = alc_eapd_ctrl_get, \ + .put = alc_eapd_ctrl_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + +static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (!cfg->line_outs) { + while (cfg->line_outs < AUTO_CFG_MAX_OUTS && + cfg->line_out_pins[cfg->line_outs]) + cfg->line_outs++; + } + if (!cfg->speaker_outs) { + while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS && + cfg->speaker_pins[cfg->speaker_outs]) + cfg->speaker_outs++; + } + if (!cfg->hp_outs) { + while (cfg->hp_outs < AUTO_CFG_MAX_OUTS && + cfg->hp_pins[cfg->hp_outs]) + cfg->hp_outs++; + } +} + +/* + * set up from the preset table + */ +static void setup_preset(struct hda_codec *codec, + const struct alc_config_preset *preset) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) + add_mixer(spec, preset->mixers[i]); + spec->cap_mixer = preset->cap_mixer; + for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; + i++) + add_verb(spec, preset->init_verbs[i]); + + spec->channel_mode = preset->channel_mode; + spec->num_channel_mode = preset->num_channel_mode; + spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; + + if (preset->const_channel_count) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; + + spec->multiout.num_dacs = preset->num_dacs; + spec->multiout.dac_nids = preset->dac_nids; + spec->multiout.dig_out_nid = preset->dig_out_nid; + spec->multiout.slave_dig_outs = preset->slave_dig_outs; + spec->multiout.hp_nid = preset->hp_nid; + + spec->num_mux_defs = preset->num_mux_defs; + if (!spec->num_mux_defs) + spec->num_mux_defs = 1; + spec->input_mux = preset->input_mux; + + spec->num_adc_nids = preset->num_adc_nids; + spec->adc_nids = preset->adc_nids; + spec->capsrc_nids = preset->capsrc_nids; + spec->dig_in_nid = preset->dig_in_nid; + + spec->unsol_event = preset->unsol_event; + spec->init_hook = preset->init_hook; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = preset->power_hook; + spec->loopback.amplist = preset->loopbacks; +#endif + + if (preset->setup) + preset->setup(codec); + + alc_fixup_autocfg_pin_nums(codec); +} + +static void alc_simple_setup_automute(struct alc_spec *spec, int mode) +{ + int lo_pin = spec->autocfg.line_out_pins[0]; + + if (lo_pin == spec->autocfg.speaker_pins[0] || + lo_pin == spec->autocfg.hp_pins[0]) + lo_pin = 0; + spec->automute_mode = mode; + spec->detect_hp = !!spec->autocfg.hp_pins[0]; + spec->detect_lo = !!lo_pin; + spec->automute_lo = spec->automute_lo_possible = !!lo_pin; + spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; +} + +/* auto-toggle front mic */ +static void alc88x_simple_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_jack_detect(codec, 0x18); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); +} + diff --git a/trunk/sound/pci/hda/hda_codec.c b/trunk/sound/pci/hda/hda_codec.c index 7a8fcc4c15f8..c2c65f63bf06 100644 --- a/trunk/sound/pci/hda/hda_codec.c +++ b/trunk/sound/pci/hda/hda_codec.c @@ -19,7 +19,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include @@ -1760,11 +1759,7 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && - (info->amp_caps & AC_AMPCAP_MIN_MUTE)) - ; /* set the zero value as a fake mute */ - else - parm |= val; + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2031,7 +2026,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) + if (min_mute) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; @@ -2305,7 +2300,7 @@ typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *); /* apply the function to all matching slave ctls in the mixer list */ static int map_slaves(struct hda_codec *codec, const char * const *slaves, - const char *suffix, map_slave_func_t func, void *data) + map_slave_func_t func, void *data) { struct hda_nid_item *items; const char * const *s; @@ -2318,14 +2313,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER) continue; for (s = slaves; *s; s++) { - char tmpname[sizeof(sctl->id.name)]; - const char *name = *s; - if (suffix) { - snprintf(tmpname, sizeof(tmpname), "%s %s", - name, suffix); - name = tmpname; - } - if (!strcmp(sctl->id.name, name)) { + if (!strcmp(sctl->id.name, *s)) { err = func(data, sctl); if (err) return err; @@ -2341,65 +2329,12 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) return 1; } -/* guess the value corresponding to 0dB */ -static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) -{ - int _tlv[4]; - const int *tlv = NULL; - int val = -1; - - if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - /* FIXME: set_fs() hack for obtaining user-space TLV data */ - mm_segment_t fs = get_fs(); - set_fs(get_ds()); - if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv)) - tlv = _tlv; - set_fs(fs); - } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) - tlv = kctl->tlv.p; - if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) - val = -tlv[2] / tlv[3]; - return val; -} - -/* call kctl->put with the given value(s) */ -static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) -{ - struct snd_ctl_elem_value *ucontrol; - ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL); - if (!ucontrol) - return -ENOMEM; - ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[1] = val; - kctl->put(kctl, ucontrol); - kfree(ucontrol); - return 0; -} - -/* initialize the slave volume with 0dB */ -static int init_slave_0dB(void *data, struct snd_kcontrol *slave) -{ - int offset = get_kctl_0dB_offset(slave); - if (offset > 0) - put_kctl_with_value(slave, offset); - return 0; -} - -/* unmute the slave */ -static int init_slave_unmute(void *data, struct snd_kcontrol *slave) -{ - return put_kctl_with_value(slave, 1); -} - /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec * @name: vmaster control name * @tlv: TLV data (optional) * @slaves: slave control names (optional) - * @suffix: suffix string to each slave name (optional) - * @init_slave_vol: initialize slaves to unmute/0dB - * @ctl_ret: store the vmaster kcontrol in return * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. @@ -2410,18 +2345,13 @@ static int init_slave_unmute(void *data, struct snd_kcontrol *slave) * * This function returns zero if successful or a negative error code. */ -int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves, - const char *suffix, bool init_slave_vol, - struct snd_kcontrol **ctl_ret) +int snd_hda_add_vmaster(struct hda_codec *codec, char *name, + unsigned int *tlv, const char * const *slaves) { struct snd_kcontrol *kctl; int err; - if (ctl_ret) - *ctl_ret = NULL; - - err = map_slaves(codec, slaves, suffix, check_slave_present, NULL); + err = map_slaves(codec, slaves, check_slave_present, NULL); if (err != 1) { snd_printdd("No slave found for %s\n", name); return 0; @@ -2433,119 +2363,13 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (err < 0) return err; - err = map_slaves(codec, slaves, suffix, - (map_slave_func_t)snd_ctl_add_slave, kctl); + err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave, + kctl); if (err < 0) return err; - - /* init with master mute & zero volume */ - put_kctl_with_value(kctl, 0); - if (init_slave_vol) - map_slaves(codec, slaves, suffix, - tlv ? init_slave_0dB : init_slave_unmute, kctl); - - if (ctl_ret) - *ctl_ret = kctl; - return 0; -} -EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster); - -/* - * mute-LED control using vmaster - */ -static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "Off", "On", "Follow Master" - }; - unsigned int index; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - index = uinfo->value.enumerated.item; - if (index >= 3) - index = 2; - strcpy(uinfo->value.enumerated.name, texts[index]); - return 0; -} - -static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hook->mute_mode; return 0; } - -static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol); - unsigned int old_mode = hook->mute_mode; - - hook->mute_mode = ucontrol->value.enumerated.item[0]; - if (hook->mute_mode > HDA_VMUTE_FOLLOW_MASTER) - hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; - if (old_mode == hook->mute_mode) - return 0; - snd_hda_sync_vmaster_hook(hook); - return 1; -} - -static struct snd_kcontrol_new vmaster_mute_mode = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mute-LED Mode", - .info = vmaster_mute_mode_info, - .get = vmaster_mute_mode_get, - .put = vmaster_mute_mode_put, -}; - -/* - * Add a mute-LED hook with the given vmaster switch kctl - * "Mute-LED Mode" control is automatically created and associated with - * the given hook. - */ -int snd_hda_add_vmaster_hook(struct hda_codec *codec, - struct hda_vmaster_mute_hook *hook, - bool expose_enum_ctl) -{ - struct snd_kcontrol *kctl; - - if (!hook->hook || !hook->sw_kctl) - return 0; - snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec); - hook->codec = codec; - hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; - if (!expose_enum_ctl) - return 0; - kctl = snd_ctl_new1(&vmaster_mute_mode, hook); - if (!kctl) - return -ENOMEM; - return snd_hda_ctl_add(codec, 0, kctl); -} -EXPORT_SYMBOL_HDA(snd_hda_add_vmaster_hook); - -/* - * Call the hook with the current value for synchronization - * Should be called in init callback - */ -void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook) -{ - if (!hook->hook || !hook->codec) - return; - switch (hook->mute_mode) { - case HDA_VMUTE_FOLLOW_MASTER: - snd_ctl_sync_vmaster_hook(hook->sw_kctl); - break; - default: - hook->hook(hook->codec, hook->mute_mode); - break; - } -} -EXPORT_SYMBOL_HDA(snd_hda_sync_vmaster_hook); - +EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch @@ -5290,7 +5114,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5349,7 +5173,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line Out", + return fill_audio_out_name(codec, nid, cfg, "Line-Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", @@ -5444,10 +5268,6 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); - else /* forcibly change the power to D3 even if not used */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D3); if (codec->patch_ops.post_suspend) codec->patch_ops.post_suspend(codec); } diff --git a/trunk/sound/pci/hda/hda_codec.h b/trunk/sound/pci/hda/hda_codec.h index 9a9f372e1be4..e9f71dc0d464 100644 --- a/trunk/sound/pci/hda/hda_codec.h +++ b/trunk/sound/pci/hda/hda_codec.h @@ -298,9 +298,6 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 -/* driver-specific amp-caps: using bits 24-30 */ -#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ - /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) @@ -855,7 +852,6 @@ struct hda_codec { unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ - unsigned int no_jack_detect:1; /* Machine has no jack-detection */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/trunk/sound/pci/hda/hda_eld.c b/trunk/sound/pci/hda/hda_eld.c index b58b4b1687fa..c1da422e085a 100644 --- a/trunk/sound/pci/hda/hda_eld.c +++ b/trunk/sound/pci/hda/hda_eld.c @@ -385,8 +385,8 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) { static unsigned int alsa_rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, - 88200, 96000, 176400, 192000, 384000 + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 384000 }; int i, j; diff --git a/trunk/sound/pci/hda/hda_intel.c b/trunk/sound/pci/hda/hda_intel.c index c19e71a94e1b..95dfb6874941 100644 --- a/trunk/sound/pci/hda/hda_intel.c +++ b/trunk/sound/pci/hda/hda_intel.c @@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); + "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -94,7 +94,7 @@ MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization."); module_param(single_cmd, bool, 0444); MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); -module_param(enable_msi, bint, 0444); +module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); #ifdef CONFIG_SND_HDA_PATCH_LOADER module_param_array(patch, charp, NULL, 0444); @@ -121,8 +121,8 @@ module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif -static int align_buffer_size = -1; -module_param(align_buffer_size, bint, 0644); +static bool align_buffer_size = 1; +module_param(align_buffer_size, bool, 0644); MODULE_PARM_DESC(align_buffer_size, "Force buffer and period sizes to be multiple of 128 bytes."); @@ -148,7 +148,6 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, PCH}," "{Intel, CPT}," "{Intel, PPT}," - "{Intel, LPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -330,7 +329,6 @@ enum { POS_FIX_LPIB, POS_FIX_POSBUF, POS_FIX_VIACOMBO, - POS_FIX_COMBO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -517,7 +515,6 @@ enum { #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ -#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -530,8 +527,7 @@ enum { /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ - AZX_DCAPS_ALIGN_BUFSIZE) + (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI) static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2351,6 +2347,17 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ +static int snd_hda_codecs_inuse(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + if (snd_hda_codec_needs_resume(codec)) + return 1; + } + return 0; +} + static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -2397,7 +2404,8 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + if (snd_hda_codecs_inuse(chip->bus)) + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -2509,7 +2517,6 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: - case POS_FIX_COMBO: return fix; } @@ -2689,12 +2696,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); - /* combo mode uses LPIB for playback */ - if (chip->position_fix[0] == POS_FIX_COMBO) { - chip->position_fix[0] = POS_FIX_LPIB; - chip->position_fix[1] = POS_FIX_AUTO; - } - check_probe_mask(chip, dev); chip->single_cmd = single_cmd; @@ -2773,16 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } /* disable buffer size rounding to 128-byte multiples if supported */ - if (align_buffer_size >= 0) - chip->align_buffer_size = !!align_buffer_size; - else { - if (chip->driver_caps & AZX_DCAPS_BUFSIZE) - chip->align_buffer_size = 0; - else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE) - chip->align_buffer_size = 1; - else - chip->align_buffer_size = 1; - } + chip->align_buffer_size = align_buffer_size; + if (chip->driver_caps & AZX_DCAPS_BUFSIZE) + chip->align_buffer_size = 0; /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) @@ -2998,10 +2992,6 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE}, - /* Lynx Point */ - { PCI_DEVICE(0x8086, 0x8c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | diff --git a/trunk/sound/pci/hda/hda_jack.c b/trunk/sound/pci/hda/hda_jack.c index d68948499fbc..9d819c4b4923 100644 --- a/trunk/sound/pci/hda/hda_jack.c +++ b/trunk/sound/pci/hda/hda_jack.c @@ -19,22 +19,6 @@ #include "hda_local.h" #include "hda_jack.h" -bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) -{ - if (codec->no_jack_detect) - return false; - if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) - return false; - if (!codec->ignore_misc_bit && - (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - return false; - if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) - return false; - return true; -} -EXPORT_SYMBOL_HDA(is_jack_detectable); - /* execute pin sense measurement */ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) { diff --git a/trunk/sound/pci/hda/hda_jack.h b/trunk/sound/pci/hda/hda_jack.h index c66655cf413a..f8f97c71c9c1 100644 --- a/trunk/sound/pci/hda/hda_jack.h +++ b/trunk/sound/pci/hda/hda_jack.h @@ -62,7 +62,18 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); -bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); +static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) +{ + if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) + return false; + if (!codec->ignore_misc_bit && + (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + return false; + if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) + return false; + return true; +} int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, const char *name, int idx); diff --git a/trunk/sound/pci/hda/hda_local.h b/trunk/sound/pci/hda/hda_local.h index 0ec9248165bc..aca8d3193b95 100644 --- a/trunk/sound/pci/hda/hda_local.h +++ b/trunk/sound/pci/hda/hda_local.h @@ -139,36 +139,10 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv); struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); -int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves, - const char *suffix, bool init_slave_vol, - struct snd_kcontrol **ctl_ret); -#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ - __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) +int snd_hda_add_vmaster(struct hda_codec *codec, char *name, + unsigned int *tlv, const char * const *slaves); int snd_hda_codec_reset(struct hda_codec *codec); -enum { - HDA_VMUTE_OFF, - HDA_VMUTE_ON, - HDA_VMUTE_FOLLOW_MASTER, -}; - -struct hda_vmaster_mute_hook { - /* below two fields must be filled by the caller of - * snd_hda_add_vmaster_hook() beforehand - */ - struct snd_kcontrol *sw_kctl; - void (*hook)(void *, int); - /* below are initialized automatically */ - unsigned int mute_mode; /* HDA_VMUTE_XXX */ - struct hda_codec *codec; -}; - -int snd_hda_add_vmaster_hook(struct hda_codec *codec, - struct hda_vmaster_mute_hook *hook, - bool expose_enum_ctl); -void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook); - /* amp value bits */ #define HDA_AMP_MUTE 0x80 #define HDA_AMP_UNMUTE 0x00 diff --git a/trunk/sound/pci/hda/patch_analog.c b/trunk/sound/pci/hda/patch_analog.c index 7143393927da..9cb14b42dfff 100644 --- a/trunk/sound/pci/hda/patch_analog.c +++ b/trunk/sound/pci/hda/patch_analog.c @@ -82,7 +82,6 @@ struct ad198x_spec { unsigned int inv_jack_detect: 1;/* inverted jack-detection */ unsigned int inv_eapd: 1; /* inverted EAPD implementation */ unsigned int analog_beep: 1; /* analog beep input present */ - unsigned int avoid_init_slave_vol:1; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -138,17 +137,51 @@ static int ad198x_init(struct hda_codec *codec) return 0; } -static const char * const ad_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Mono", "Speaker", "IEC958", +static const char * const ad_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Mono Playback Volume", + "Speaker Playback Volume", + "IEC958 Playback Volume", NULL }; -static const char * const ad1988_6stack_fp_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", "IEC958", +static const char * const ad_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Mono Playback Switch", + "Speaker Playback Switch", + "IEC958 Playback Switch", NULL }; +static const char * const ad1988_6stack_fp_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "IEC958 Playback Volume", + NULL +}; + +static const char * const ad1988_6stack_fp_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "IEC958 Playback Switch", + NULL +}; static void ad198x_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -224,12 +257,10 @@ static int ad198x_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); - err = __snd_hda_add_vmaster(codec, "Master Playback Volume", + err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, (spec->slave_vols ? - spec->slave_vols : ad_slave_pfxs), - "Playback Volume", - !spec->avoid_init_slave_vol, NULL); + spec->slave_vols : ad_slave_vols)); if (err < 0) return err; } @@ -237,8 +268,7 @@ static int ad198x_build_controls(struct hda_codec *codec) err = snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, (spec->slave_sws ? - spec->slave_sws : ad_slave_pfxs), - "Playback Switch"); + spec->slave_sws : ad_slave_sws)); if (err < 0) return err; } @@ -3355,8 +3385,8 @@ static int patch_ad1988(struct hda_codec *codec) if (spec->autocfg.hp_pins[0]) { spec->mixers[spec->num_mixers++] = ad1988_hp_mixers; - spec->slave_vols = ad1988_6stack_fp_slave_pfxs; - spec->slave_sws = ad1988_6stack_fp_slave_pfxs; + spec->slave_vols = ad1988_6stack_fp_slave_vols; + spec->slave_sws = ad1988_6stack_fp_slave_sws; spec->alt_dac_nid = ad1988_alt_dac_nid; spec->stream_analog_alt_playback = &ad198x_pcm_analog_alt_playback; @@ -3564,8 +3594,16 @@ static const struct hda_amp_list ad1884_loopbacks[] = { #endif static const char * const ad1884_slave_vols[] = { - "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958", + "PCM Playback Volume", + "Mic Playback Volume", + "Mono Playback Volume", + "Front Mic Playback Volume", + "Mic Playback Volume", + "CD Playback Volume", + "Internal Mic Playback Volume", + "Docking Mic Playback Volume", + /* "Beep Playback Volume", */ + "IEC958 Playback Volume", NULL }; @@ -3606,8 +3644,6 @@ static int patch_ad1884(struct hda_codec *codec) spec->vmaster_nid = 0x04; /* we need to cover all playback volumes */ spec->slave_vols = ad1884_slave_vols; - /* slaves may contain input volumes, so we can't raise to 0dB blindly */ - spec->avoid_init_slave_vol = 1; codec->patch_ops = ad198x_patch_ops; diff --git a/trunk/sound/pci/hda/patch_cirrus.c b/trunk/sound/pci/hda/patch_cirrus.c index c83ccdba1e5a..bc5a993d1146 100644 --- a/trunk/sound/pci/hda/patch_cirrus.c +++ b/trunk/sound/pci/hda/patch_cirrus.c @@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, "Front Speaker", "Surround Speaker", "Bass Speaker" }; static const char * const line_outs[] = { - "Front Line Out", "Surround Line Out", "Bass Line Out" + "Front Line-Out", "Surround Line-Out", "Bass Line-Out" }; fix_volume_caps(codec, dac); @@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, if (num_ctls > 1) name = line_outs[idx]; else - name = "Line Out"; + name = "Line-Out"; break; } diff --git a/trunk/sound/pci/hda/patch_conexant.c b/trunk/sound/pci/hda/patch_conexant.c index e6eafb18c8f5..a7a5733aa4d2 100644 --- a/trunk/sound/pci/hda/patch_conexant.c +++ b/trunk/sound/pci/hda/patch_conexant.c @@ -70,8 +70,6 @@ struct conexant_spec { const struct snd_kcontrol_new *mixers[5]; int num_mixers; hda_nid_t vmaster_nid; - struct hda_vmaster_mute_hook vmaster_mute; - bool vmaster_mute_led; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -467,8 +465,21 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = { }; #endif -static const char * const slave_pfxs[] = { - "Headphone", "Speaker", "Front", "Surround", "CLFE", +static const char * const slave_vols[] = { + "Headphone Playback Volume", + "Speaker Playback Volume", + "Front Playback Volume", + "Surround Playback Volume", + "CLFE Playback Volume", + NULL +}; + +static const char * const slave_sws[] = { + "Headphone Playback Switch", + "Speaker Playback Switch", + "Front Playback Switch", + "Surround Playback Switch", + "CLFE Playback Switch", NULL }; @@ -508,17 +519,14 @@ static int conexant_build_controls(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); + vmaster_tlv, slave_vols); if (err < 0) return err; } if (spec->vmaster_nid && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch", true, - &spec->vmaster_mute.sw_kctl); + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_sws); if (err < 0) return err; } @@ -3026,6 +3034,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3473,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line Out+Speaker" + "Disabled", "Speaker Only", "Line-Out+Speaker" }; const char * const *texts; @@ -3934,63 +3943,6 @@ static void enable_unsol_pins(struct hda_codec *codec, int num_pins, snd_hda_jack_detect_enable(codec, pins[i], action); } -static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) -{ - int i; - for (i = 0; i < nums; i++) - if (list[i] == nid) - return true; - return false; -} - -/* is the given NID found in any of autocfg items? */ -static bool found_in_autocfg(struct auto_pin_cfg *cfg, hda_nid_t nid) -{ - int i; - - if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) || - found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) || - found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs) || - found_in_nid_list(nid, cfg->dig_out_pins, cfg->dig_outs)) - return true; - for (i = 0; i < cfg->num_inputs; i++) - if (cfg->inputs[i].pin == nid) - return true; - if (cfg->dig_in_pin == nid) - return true; - return false; -} - -/* clear unsol-event tags on unused pins; Conexant codecs seem to leave - * invalid unsol tags by some reason - */ -static void clear_unsol_on_unused_pins(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); - if (!found_in_autocfg(cfg, pin->nid)) - snd_hda_codec_write(codec, pin->nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, 0); - } -} - -/* turn on/off EAPD according to Master switch */ -static void cx_auto_vmaster_hook(void *private_data, int enabled) -{ - struct hda_codec *codec = private_data; - struct conexant_spec *spec = codec->spec; - - if (enabled && spec->pin_eapd_ctrls) { - cx_auto_update_speakers(codec); - return; - } - cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled); -} - static void cx_auto_init_output(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -4031,7 +3983,6 @@ static void cx_auto_init_output(struct hda_codec *codec) /* turn on all EAPDs if no individual EAPD control is available */ if (!spec->pin_eapd_ctrls) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); - clear_unsol_on_unused_pins(codec); } static void cx_auto_init_input(struct hda_codec *codec) @@ -4095,13 +4046,11 @@ static void cx_auto_init_digital(struct hda_codec *codec) static int cx_auto_init(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; /*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/ cx_auto_init_output(codec); cx_auto_init_input(codec); cx_auto_init_digital(codec); snd_hda_jack_report_sync(codec); - snd_hda_sync_vmaster_hook(&spec->vmaster_mute); return 0; } @@ -4130,8 +4079,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & - (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) + if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) break; } return 0; @@ -4347,13 +4295,6 @@ static int cx_auto_build_controls(struct hda_codec *codec) err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; - if (spec->vmaster_mute.sw_kctl) { - spec->vmaster_mute.hook = cx_auto_vmaster_hook; - err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, - spec->vmaster_mute_led); - if (err < 0) - return err; - } return 0; } @@ -4378,6 +4319,7 @@ static int cx_auto_search_adcs(struct hda_codec *codec) return 0; } + static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = conexant_build_pcms, @@ -4425,7 +4367,6 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ { 0x19, 0x2121103f }, /* dock-HP */ - { 0x1c, 0x21440100 }, /* dock SPDIF out */ {} }; @@ -4438,22 +4379,6 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; -/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches - * can be created (bko#42825) - */ -static void add_cx5051_fake_mutes(struct hda_codec *codec) -{ - static hda_nid_t out_nids[] = { - 0x10, 0x11, 0 - }; - hda_nid_t *p; - - for (p = out_nids; *p; p++) - snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, - AC_AMPCAP_MIN_MUTE | - query_amp_caps(codec, *p, HDA_OUTPUT)); -} - static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4472,25 +4397,10 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; - case 0x14f15051: - add_cx5051_fake_mutes(codec); - break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); - /* Show mute-led control only on HP laptops - * This is a sort of white-list: on HP laptops, EAPD corresponds - * only to the mute-LED without actualy amp function. Meanwhile, - * others may use EAPD really as an amp switch, so it might be - * not good to expose it blindly. - */ - switch (codec->subsystem_id >> 16) { - case 0x103c: - spec->vmaster_mute_led = 1; - break; - } - err = cx_auto_search_adcs(codec); if (err < 0) return err; @@ -4504,18 +4414,6 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->patch_ops = cx_auto_patch_ops; if (spec->beep_amp) snd_hda_attach_beep_device(codec, spec->beep_amp); - - /* Some laptops with Conexant chips show stalls in S3 resume, - * which falls into the single-cmd mode. - * Better to make reset, then. - */ - if (!codec->bus->sync_write) { - snd_printd("hda_codec: " - "Enable sync_write for stable communication\n"); - codec->bus->sync_write = 1; - codec->bus->allow_bus_reset = 1; - } - return 0; } diff --git a/trunk/sound/pci/hda/patch_hdmi.c b/trunk/sound/pci/hda/patch_hdmi.c index 540cd13f7f15..1168ebd3fb5c 100644 --- a/trunk/sound/pci/hda/patch_hdmi.c +++ b/trunk/sound/pci/hda/patch_hdmi.c @@ -1912,7 +1912,6 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -1959,7 +1958,6 @@ MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); -MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL"); diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c index 9917e55d6f11..1358987c49d8 100644 --- a/trunk/sound/pci/hda/patch_realtek.c +++ b/trunk/sound/pci/hda/patch_realtek.c @@ -80,8 +80,6 @@ enum { ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; -#define MAX_VOL_NIDS 0x40 - struct alc_spec { /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -120,8 +118,8 @@ struct alc_spec { const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ hda_nid_t mixer_nid; /* analog-mixer NID */ - DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1); - DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1); + DECLARE_BITMAP(vol_ctls, 0x20 << 1); + DECLARE_BITMAP(sw_ctls, 0x20 << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -198,11 +196,8 @@ struct alc_spec { /* for virtual master */ hda_nid_t vmaster_nid; - struct hda_vmaster_mute_hook vmaster_mute; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; - int num_loopbacks; - struct hda_amp_list loopback_list[8]; #endif /* for PLL fix */ @@ -223,6 +218,8 @@ struct alc_spec { struct snd_array bind_ctls; }; +#define ALC_MODEL_AUTO 0 /* common for all chips */ + static bool check_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int bits) { @@ -301,9 +298,6 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, int i, type, num_conns; hda_nid_t nid; - if (!spec->input_mux) - return 0; - mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; if (!imux->num_items && mux_idx > 0) @@ -655,51 +649,15 @@ static void alc_exec_unsol_event(struct hda_codec *codec, int action) snd_hda_jack_report_sync(codec); } -/* update the master volume per volume-knob's unsol event */ -static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int val; - struct snd_kcontrol *kctl; - struct snd_ctl_elem_value *uctl; - - kctl = snd_hda_find_mixer_ctl(codec, "Master Playback Volume"); - if (!kctl) - return; - uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); - if (!uctl) - return; - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); - val &= HDA_AMP_VOLMASK; - uctl->value.integer.value[0] = val; - uctl->value.integer.value[1] = val; - kctl->put(kctl, uctl); - kfree(uctl); -} - /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { - int action; - if (codec->vendor_id == 0x10ec0880) res >>= 28; else res >>= 26; - action = snd_hda_jack_get_action(codec, res); - if (action == ALC_DCVOL_EVENT) { - /* Execute the dc-vol event here as it requires the NID - * but we don't pass NID to alc_exec_unsol_event(). - * Once when we convert all static quirks to the auto-parser, - * this can be integerated into there. - */ - struct hda_jack_tbl *jack; - jack = snd_hda_jack_tbl_get_from_tag(codec, res); - if (jack) - alc_update_knob_master(codec, jack->nid); - return; - } - alc_exec_unsol_event(codec, action); + res = snd_hda_jack_get_action(codec, res); + alc_exec_unsol_event(codec, res); } /* call init functions of standard auto-mute helpers */ @@ -842,7 +800,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line Out+Speaker" + "Disabled", "Speaker Only", "Line-Out+Speaker" }; const char * const *texts; @@ -1073,6 +1031,45 @@ static bool alc_check_dyn_adc_switch(struct hda_codec *codec) return true; } +/* rebuild imux for matching with the given auto-mic pins (if not yet) */ +static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct hda_input_mux *imux; + static char * const texts[3] = { + "Mic", "Internal Mic", "Dock Mic" + }; + int i; + + if (!spec->auto_mic) + return false; + imux = &spec->private_imux[0]; + if (spec->input_mux == imux) + return true; + spec->imux_pins[0] = spec->ext_mic_pin; + spec->imux_pins[1] = spec->int_mic_pin; + spec->imux_pins[2] = spec->dock_mic_pin; + for (i = 0; i < 3; i++) { + strcpy(imux->items[i].label, texts[i]); + if (spec->imux_pins[i]) { + hda_nid_t pin = spec->imux_pins[i]; + int c; + for (c = 0; c < spec->num_adc_nids; c++) { + hda_nid_t cap = get_capsrc(spec, c); + int idx = get_connection_index(codec, cap, pin); + if (idx >= 0) { + imux->items[i].index = idx; + break; + } + } + imux->num_items = i + 1; + } + } + spec->num_mux_defs = 1; + spec->input_mux = imux; + return true; +} + /* check whether all auto-mic pins are valid; setup indices if OK */ static bool alc_auto_mic_check_imux(struct hda_codec *codec) { @@ -1442,7 +1439,6 @@ enum { ALC_FIXUP_ACT_PRE_PROBE, ALC_FIXUP_ACT_PROBE, ALC_FIXUP_ACT_INIT, - ALC_FIXUP_ACT_BUILD, }; static void alc_apply_fixup(struct hda_codec *codec, int action) @@ -1522,13 +1518,6 @@ static void alc_pick_fixup(struct hda_codec *codec, int id = -1; const char *name = NULL; - /* when model=nofixup is given, don't pick up any fixups */ - if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { - spec->fixup_list = NULL; - spec->fixup_id = -1; - return; - } - if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { @@ -1856,10 +1845,36 @@ DEFINE_CAPMIX_NOSRC(3); /* * slave controls for virtual master */ -static const char * const alc_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "Mono", "Line Out", - "CLFE", "Bass Speaker", "PCM", +static const char * const alc_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Speaker Playback Volume", + "Mono Playback Volume", + "Line-Out Playback Volume", + "CLFE Playback Volume", + "Bass Speaker Playback Volume", + "PCM Playback Volume", + NULL, +}; + +static const char * const alc_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Speaker Playback Switch", + "Mono Playback Switch", + "IEC958 Playback Switch", + "Line-Out Playback Switch", + "CLFE Playback Switch", + "Bass Speaker Playback Switch", + "PCM Playback Switch", NULL, }; @@ -1950,17 +1965,14 @@ static int __alc_build_controls(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, alc_slave_pfxs, - "Playback Volume"); + vmaster_tlv, alc_slave_vols); if (err < 0) return err; } if (!spec->no_analog && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, alc_slave_pfxs, - "Playback Switch", - true, &spec->vmaster_mute.sw_kctl); + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, alc_slave_sws); if (err < 0) return err; } @@ -2045,11 +2057,7 @@ static int alc_build_controls(struct hda_codec *codec) int err = __alc_build_controls(codec); if (err < 0) return err; - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); - if (err < 0) - return err; - alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD); - return 0; + return snd_hda_jack_add_kctls(codec, &spec->autocfg); } @@ -2058,23 +2066,21 @@ static int alc_build_controls(struct hda_codec *codec) */ static void alc_init_special_input_src(struct hda_codec *codec); -static void alc_auto_init_std(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int i; - if (spec->init_hook) - spec->init_hook(codec); - alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); alc_init_special_input_src(codec); - alc_auto_init_std(codec); + + if (spec->init_hook) + spec->init_hook(codec); alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT); @@ -2663,25 +2669,6 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, return channel_name[ch]; } -#ifdef CONFIG_SND_HDA_POWER_SAVE -/* add the powersave loopback-list entry */ -static void add_loopback_list(struct alc_spec *spec, hda_nid_t mix, int idx) -{ - struct hda_amp_list *list; - - if (spec->num_loopbacks >= ARRAY_SIZE(spec->loopback_list) - 1) - return; - list = spec->loopback_list + spec->num_loopbacks; - list->nid = mix; - list->dir = HDA_INPUT; - list->idx = idx; - spec->num_loopbacks++; - spec->loopback.amplist = spec->loopback_list; -} -#else -#define add_loopback_list(spec, mix, idx) /* NOP */ -#endif - /* create input playback/capture controls for the given pin */ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2697,7 +2684,6 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; - add_loopback_list(spec, mix_nid, idx); return 0; } @@ -2717,6 +2703,9 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) int max_nums = ARRAY_SIZE(spec->private_adc_nids); int i, nums = 0; + if (spec->shared_mic_hp) + max_nums = 1; /* no multi streams with the shared HP/mic */ + nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { hda_nid_t src; @@ -2959,27 +2948,10 @@ static int alc_auto_select_dac(struct hda_codec *codec, hda_nid_t pin, return 0; } -static bool alc_is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) -{ - struct alc_spec *spec = codec->spec; - int i; - if (found_in_nid_list(nid, spec->multiout.dac_nids, - ARRAY_SIZE(spec->private_dac_nids)) || - found_in_nid_list(nid, spec->multiout.hp_out_nid, - ARRAY_SIZE(spec->multiout.hp_out_nid)) || - found_in_nid_list(nid, spec->multiout.extra_out_nid, - ARRAY_SIZE(spec->multiout.extra_out_nid))) - return true; - for (i = 0; i < spec->multi_ios; i++) { - if (spec->multi_io[i].dac == nid) - return true; - } - return false; -} - /* look for an empty DAC slot */ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) { + struct alc_spec *spec = codec->spec; hda_nid_t srcs[5]; int i, num; @@ -2989,8 +2961,16 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]); if (!nid) continue; - if (!alc_is_dac_already_used(codec, nid)) - return nid; + if (found_in_nid_list(nid, spec->multiout.dac_nids, + ARRAY_SIZE(spec->private_dac_nids))) + continue; + if (found_in_nid_list(nid, spec->multiout.hp_out_nid, + ARRAY_SIZE(spec->multiout.hp_out_nid))) + continue; + if (found_in_nid_list(nid, spec->multiout.extra_out_nid, + ARRAY_SIZE(spec->multiout.extra_out_nid))) + continue; + return nid; } return 0; } @@ -3002,8 +2982,6 @@ static bool alc_auto_is_dac_reachable(struct hda_codec *codec, hda_nid_t srcs[5]; int i, num; - if (!pin || !dac) - return false; pin = alc_go_down_to_selector(codec, pin); num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs)); for (i = 0; i < num; i++) { @@ -3016,260 +2994,83 @@ static bool alc_auto_is_dac_reachable(struct hda_codec *codec, static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { - struct alc_spec *spec = codec->spec; hda_nid_t sel = alc_go_down_to_selector(codec, pin); - hda_nid_t nid, nid_found, srcs[5]; - int i, num = snd_hda_get_connections(codec, sel, srcs, - ARRAY_SIZE(srcs)); - if (num == 1) + if (snd_hda_get_conn_list(codec, sel, NULL) == 1) return alc_auto_look_for_dac(codec, pin); - nid_found = 0; - for (i = 0; i < num; i++) { - if (srcs[i] == spec->mixer_nid) - continue; - nid = alc_auto_mix_to_dac(codec, srcs[i]); - if (nid && !alc_is_dac_already_used(codec, nid)) { - if (nid_found) - return 0; - nid_found = nid; - } - } - return nid_found; -} - -/* mark up volume and mute control NIDs: used during badness parsing and - * at creating actual controls - */ -static inline unsigned int get_ctl_pos(unsigned int data) -{ - hda_nid_t nid = get_amp_nid_(data); - unsigned int dir; - if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) - return 0; - dir = get_amp_direction_(data); - return (nid << 1) | dir; -} - -#define is_ctl_used(bits, data) \ - test_bit(get_ctl_pos(data), bits) -#define mark_ctl_usage(bits, data) \ - set_bit(get_ctl_pos(data), bits) - -static void clear_vol_marks(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - memset(spec->vol_ctls, 0, sizeof(spec->vol_ctls)); - memset(spec->sw_ctls, 0, sizeof(spec->sw_ctls)); + return 0; } -/* badness definition */ -enum { - /* No primary DAC is found for the main output */ - BAD_NO_PRIMARY_DAC = 0x10000, - /* No DAC is found for the extra output */ - BAD_NO_DAC = 0x4000, - /* No possible multi-ios */ - BAD_MULTI_IO = 0x103, - /* No individual DAC for extra output */ - BAD_NO_EXTRA_DAC = 0x102, - /* No individual DAC for extra surrounds */ - BAD_NO_EXTRA_SURR_DAC = 0x101, - /* Primary DAC shared with main surrounds */ - BAD_SHARED_SURROUND = 0x100, - /* Primary DAC shared with main CLFE */ - BAD_SHARED_CLFE = 0x10, - /* Primary DAC shared with extra surrounds */ - BAD_SHARED_EXTRA_SURROUND = 0x10, - /* Volume widget is shared */ - BAD_SHARED_VOL = 0x10, -}; - -static hda_nid_t alc_look_for_out_mute_nid(struct hda_codec *codec, - hda_nid_t pin, hda_nid_t dac); -static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, - hda_nid_t pin, hda_nid_t dac); - -static int eval_shared_vol_badness(struct hda_codec *codec, hda_nid_t pin, - hda_nid_t dac) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid; - unsigned int val; - int badness = 0; - - nid = alc_look_for_out_vol_nid(codec, pin, dac); - if (nid) { - val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - if (is_ctl_used(spec->vol_ctls, nid)) - badness += BAD_SHARED_VOL; - else - mark_ctl_usage(spec->vol_ctls, val); - } else - badness += BAD_SHARED_VOL; - nid = alc_look_for_out_mute_nid(codec, pin, dac); - if (nid) { - unsigned int wid_type = get_wcaps_type(get_wcaps(codec, nid)); - if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) - val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - else - val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT); - if (is_ctl_used(spec->sw_ctls, val)) - badness += BAD_SHARED_VOL; - else - mark_ctl_usage(spec->sw_ctls, val); - } else - badness += BAD_SHARED_VOL; - return badness; -} - -struct badness_table { - int no_primary_dac; /* no primary DAC */ - int no_dac; /* no secondary DACs */ - int shared_primary; /* primary DAC is shared with main output */ - int shared_surr; /* secondary DAC shared with main or primary */ - int shared_clfe; /* third DAC shared with main or primary */ - int shared_surr_main; /* secondary DAC sahred with main/DAC0 */ -}; - -static struct badness_table main_out_badness = { - .no_primary_dac = BAD_NO_PRIMARY_DAC, - .no_dac = BAD_NO_DAC, - .shared_primary = BAD_NO_PRIMARY_DAC, - .shared_surr = BAD_SHARED_SURROUND, - .shared_clfe = BAD_SHARED_CLFE, - .shared_surr_main = BAD_SHARED_SURROUND, -}; - -static struct badness_table extra_out_badness = { - .no_primary_dac = BAD_NO_DAC, - .no_dac = BAD_NO_DAC, - .shared_primary = BAD_NO_EXTRA_DAC, - .shared_surr = BAD_SHARED_EXTRA_SURROUND, - .shared_clfe = BAD_SHARED_EXTRA_SURROUND, - .shared_surr_main = BAD_NO_EXTRA_SURR_DAC, -}; - -/* try to assign DACs to pins and return the resultant badness */ -static int alc_auto_fill_dacs(struct hda_codec *codec, int num_outs, - const hda_nid_t *pins, hda_nid_t *dacs, - const struct badness_table *bad) +/* return 0 if no possible DAC is found, 1 if one or more found */ +static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, hda_nid_t *dacs) { - struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i, j; - int badness = 0; - hda_nid_t dac; + int i; - if (!num_outs) - return 0; + if (num_outs && !dacs[0]) { + dacs[0] = alc_auto_look_for_dac(codec, pins[0]); + if (!dacs[0]) + return 0; + } - for (i = 0; i < num_outs; i++) { - hda_nid_t pin = pins[i]; + for (i = 1; i < num_outs; i++) + dacs[i] = get_dac_if_single(codec, pins[i]); + for (i = 1; i < num_outs; i++) { if (!dacs[i]) - dacs[i] = alc_auto_look_for_dac(codec, pin); - if (!dacs[i] && !i) { - for (j = 1; j < num_outs; j++) { - if (alc_auto_is_dac_reachable(codec, pin, dacs[j])) { - dacs[0] = dacs[j]; - dacs[j] = 0; - break; - } - } - } - dac = dacs[i]; - if (!dac) { - if (alc_auto_is_dac_reachable(codec, pin, dacs[0])) - dac = dacs[0]; - else if (cfg->line_outs > i && - alc_auto_is_dac_reachable(codec, pin, - spec->private_dac_nids[i])) - dac = spec->private_dac_nids[i]; - if (dac) { - if (!i) - badness += bad->shared_primary; - else if (i == 1) - badness += bad->shared_surr; - else - badness += bad->shared_clfe; - } else if (alc_auto_is_dac_reachable(codec, pin, - spec->private_dac_nids[0])) { - dac = spec->private_dac_nids[0]; - badness += bad->shared_surr_main; - } else if (!i) - badness += bad->no_primary_dac; - else - badness += bad->no_dac; - } - if (dac) - badness += eval_shared_vol_badness(codec, pin, dac); + dacs[i] = alc_auto_look_for_dac(codec, pins[i]); } - - return badness; + return 1; } static int alc_auto_fill_multi_ios(struct hda_codec *codec, - hda_nid_t reference_pin, - bool hardwired, int offset); - -static bool alc_map_singles(struct hda_codec *codec, int outs, - const hda_nid_t *pins, hda_nid_t *dacs) -{ - int i; - bool found = false; - for (i = 0; i < outs; i++) { - if (dacs[i]) - continue; - dacs[i] = get_dac_if_single(codec, pins[i]); - if (dacs[i]) - found = true; - } - return found; -} + unsigned int location, int offset); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ -static int fill_and_eval_dacs(struct hda_codec *codec, - bool fill_hardwired, - bool fill_mio_first) +static int alc_auto_fill_dac_nids(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int i, err, badness; + unsigned int location, defcfg; + int num_pins; + bool redone = false; + int i; + again: /* set num_dacs once to full for alc_auto_look_for_dac() */ spec->multiout.num_dacs = cfg->line_outs; - spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.hp_out_nid[0] = 0; + spec->multiout.extra_out_nid[0] = 0; memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); - memset(spec->multiout.hp_out_nid, 0, sizeof(spec->multiout.hp_out_nid)); - memset(spec->multiout.extra_out_nid, 0, sizeof(spec->multiout.extra_out_nid)); + spec->multiout.dac_nids = spec->private_dac_nids; spec->multi_ios = 0; - clear_vol_marks(codec); - badness = 0; /* fill hard-wired DACs first */ - if (fill_hardwired) { - bool mapped; - do { - mapped = alc_map_singles(codec, cfg->line_outs, - cfg->line_out_pins, - spec->private_dac_nids); - mapped |= alc_map_singles(codec, cfg->hp_outs, - cfg->hp_pins, - spec->multiout.hp_out_nid); - mapped |= alc_map_singles(codec, cfg->speaker_outs, - cfg->speaker_pins, - spec->multiout.extra_out_nid); - if (fill_mio_first && cfg->line_outs == 1 && - cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { - err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], true, 0); - if (!err) - mapped = true; - } - } while (mapped); + if (!redone) { + for (i = 0; i < cfg->line_outs; i++) + spec->private_dac_nids[i] = + get_dac_if_single(codec, cfg->line_out_pins[i]); + if (cfg->hp_outs) + spec->multiout.hp_out_nid[0] = + get_dac_if_single(codec, cfg->hp_pins[0]); + if (cfg->speaker_outs) + spec->multiout.extra_out_nid[0] = + get_dac_if_single(codec, cfg->speaker_pins[0]); } - badness += alc_auto_fill_dacs(codec, cfg->line_outs, cfg->line_out_pins, - spec->private_dac_nids, - &main_out_badness); + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t pin = cfg->line_out_pins[i]; + if (spec->private_dac_nids[i]) + continue; + spec->private_dac_nids[i] = alc_auto_look_for_dac(codec, pin); + if (!spec->private_dac_nids[i] && !redone) { + /* if we can't find primary DACs, re-probe without + * checking the hard-wired DACs + */ + redone = true; + goto again; + } + } /* re-count num_dacs and squash invalid entries */ spec->multiout.num_dacs = 0; @@ -3284,144 +3085,30 @@ static int fill_and_eval_dacs(struct hda_codec *codec, } } - if (fill_mio_first && - cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { /* try to fill multi-io first */ - err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], false, 0); - if (err < 0) - return err; - /* we don't count badness at this stage yet */ + defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); + location = get_defcfg_location(defcfg); + + num_pins = alc_auto_fill_multi_ios(codec, location, 0); + if (num_pins > 0) { + spec->multi_ios = num_pins; + spec->ext_channel_count = 2; + spec->multiout.num_dacs = num_pins + 1; + } } - if (cfg->line_out_type != AUTO_PIN_HP_OUT) { - err = alc_auto_fill_dacs(codec, cfg->hp_outs, cfg->hp_pins, - spec->multiout.hp_out_nid, - &extra_out_badness); - if (err < 0) - return err; - badness += err; - } + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins, + spec->multiout.hp_out_nid); if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { - err = alc_auto_fill_dacs(codec, cfg->speaker_outs, - cfg->speaker_pins, - spec->multiout.extra_out_nid, - &extra_out_badness); - if (err < 0) - return err; - badness += err; - } - if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { - err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], false, 0); - if (err < 0) - return err; - badness += err; - } - if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { - /* try multi-ios with HP + inputs */ - int offset = 0; - if (cfg->line_outs >= 3) - offset = 1; - err = alc_auto_fill_multi_ios(codec, cfg->hp_pins[0], false, - offset); - if (err < 0) - return err; - badness += err; - } - - if (spec->multi_ios == 2) { - for (i = 0; i < 2; i++) - spec->private_dac_nids[spec->multiout.num_dacs++] = - spec->multi_io[i].dac; - spec->ext_channel_count = 2; - } else if (spec->multi_ios) { - spec->multi_ios = 0; - badness += BAD_MULTI_IO; - } - - return badness; -} - -#define DEBUG_BADNESS - -#ifdef DEBUG_BADNESS -#define debug_badness snd_printdd -#else -#define debug_badness(...) -#endif - -static void debug_show_configs(struct alc_spec *spec, struct auto_pin_cfg *cfg) -{ - debug_badness("multi_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", - cfg->line_out_pins[0], cfg->line_out_pins[1], - cfg->line_out_pins[2], cfg->line_out_pins[2], - spec->multiout.dac_nids[0], - spec->multiout.dac_nids[1], - spec->multiout.dac_nids[2], - spec->multiout.dac_nids[3]); - if (spec->multi_ios > 0) - debug_badness("multi_ios(%d) = %x/%x : %x/%x\n", - spec->multi_ios, - spec->multi_io[0].pin, spec->multi_io[1].pin, - spec->multi_io[0].dac, spec->multi_io[1].dac); - debug_badness("hp_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", - cfg->hp_pins[0], cfg->hp_pins[1], - cfg->hp_pins[2], cfg->hp_pins[2], - spec->multiout.hp_out_nid[0], - spec->multiout.hp_out_nid[1], - spec->multiout.hp_out_nid[2], - spec->multiout.hp_out_nid[3]); - debug_badness("spk_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", - cfg->speaker_pins[0], cfg->speaker_pins[1], - cfg->speaker_pins[2], cfg->speaker_pins[3], - spec->multiout.extra_out_nid[0], - spec->multiout.extra_out_nid[1], - spec->multiout.extra_out_nid[2], - spec->multiout.extra_out_nid[3]); -} - -static int alc_auto_fill_dac_nids(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - struct auto_pin_cfg *best_cfg; - int best_badness = INT_MAX; - int badness; - bool fill_hardwired = true, fill_mio_first = true; - bool best_wired = true, best_mio = true; - bool hp_spk_swapped = false; - - best_cfg = kmalloc(sizeof(*best_cfg), GFP_KERNEL); - if (!best_cfg) - return -ENOMEM; - *best_cfg = *cfg; - - for (;;) { - badness = fill_and_eval_dacs(codec, fill_hardwired, - fill_mio_first); - if (badness < 0) - return badness; - debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", - cfg->line_out_type, fill_hardwired, fill_mio_first, - badness); - debug_show_configs(spec, cfg); - if (badness < best_badness) { - best_badness = badness; - *best_cfg = *cfg; - best_wired = fill_hardwired; - best_mio = fill_mio_first; - } - if (!badness) - break; - fill_mio_first = !fill_mio_first; - if (!fill_mio_first) - continue; - fill_hardwired = !fill_hardwired; - if (!fill_hardwired) - continue; - if (hp_spk_swapped) - break; - hp_spk_swapped = true; - if (cfg->speaker_outs > 0 && + int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, + cfg->speaker_pins, + spec->multiout.extra_out_nid); + /* if no speaker volume is assigned, try again as the primary + * output + */ + if (!err && cfg->speaker_outs > 0 && cfg->line_out_type == AUTO_PIN_HP_OUT) { cfg->hp_outs = cfg->line_outs; memcpy(cfg->hp_pins, cfg->line_out_pins, @@ -3432,45 +3119,45 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->speaker_outs = 0; memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; - fill_hardwired = true; - continue; - } - if (cfg->hp_outs > 0 && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { - cfg->speaker_outs = cfg->line_outs; - memcpy(cfg->speaker_pins, cfg->line_out_pins, - sizeof(cfg->speaker_pins)); - cfg->line_outs = cfg->hp_outs; - memcpy(cfg->line_out_pins, cfg->hp_pins, - sizeof(cfg->hp_pins)); - cfg->hp_outs = 0; - memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); - cfg->line_out_type = AUTO_PIN_HP_OUT; - fill_hardwired = true; - continue; - } - break; + redone = false; + goto again; + } } - if (badness) { - *cfg = *best_cfg; - fill_and_eval_dacs(codec, best_wired, best_mio); + if (!spec->multi_ios && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && + cfg->hp_outs) { + /* try multi-ios with HP + inputs */ + defcfg = snd_hda_codec_get_pincfg(codec, cfg->hp_pins[0]); + location = get_defcfg_location(defcfg); + + num_pins = alc_auto_fill_multi_ios(codec, location, 1); + if (num_pins > 0) { + spec->multi_ios = num_pins; + spec->ext_channel_count = 2; + spec->multiout.num_dacs = num_pins + 1; + } } - debug_badness("==> Best config: lo_type=%d, wired=%d, mio=%d\n", - cfg->line_out_type, best_wired, best_mio); - debug_show_configs(spec, cfg); if (cfg->line_out_pins[0]) spec->vmaster_nid = alc_look_for_out_vol_nid(codec, cfg->line_out_pins[0], spec->multiout.dac_nids[0]); - - /* clear the bitmap flags for creating controls */ - clear_vol_marks(codec); - kfree(best_cfg); return 0; } +static inline unsigned int get_ctl_pos(unsigned int data) +{ + hda_nid_t nid = get_amp_nid_(data); + unsigned int dir = get_amp_direction_(data); + return (nid << 1) | dir; +} + +#define is_ctl_used(bits, data) \ + test_bit(get_ctl_pos(data), bits) +#define mark_ctl_usage(bits, data) \ + set_bit(get_ctl_pos(data), bits) + static int alc_auto_add_vol_ctl(struct hda_codec *codec, const char *pfx, int cidx, hda_nid_t nid, unsigned int chs) @@ -3582,17 +3269,14 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, dac = spec->multiout.dac_nids[i]; if (!dac) continue; - if (i >= cfg->line_outs) { + if (i >= cfg->line_outs) pin = spec->multi_io[i - 1].pin; - index = 0; - name = channel_name[i]; - } else { + else pin = cfg->line_out_pins[i]; - name = alc_get_line_out_pfx(spec, i, true, &index); - } sw = alc_look_for_out_mute_nid(codec, pin, dac); vol = alc_look_for_out_vol_nid(codec, pin, dac); + name = alc_get_line_out_pfx(spec, i, true, &index); if (!name || !strcmp(name, "CLFE")) { /* Center/LFE */ err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1); @@ -3689,31 +3373,41 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins, return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0); } - for (i = 0; i < num_pins; i++) { - hda_nid_t dac; - if (dacs[num_pins - 1]) - dac = dacs[i]; /* with individual volumes */ - else - dac = 0; - if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) { - err = alc_auto_create_extra_out(codec, pins[i], dac, - "Bass Speaker", 0); - } else if (num_pins >= 3) { - snprintf(name, sizeof(name), "%s %s", - pfx, channel_name[i]); - err = alc_auto_create_extra_out(codec, pins[i], dac, - name, 0); - } else { - err = alc_auto_create_extra_out(codec, pins[i], dac, - pfx, i); + if (dacs[num_pins - 1]) { + /* OK, we have a multi-output system with individual volumes */ + for (i = 0; i < num_pins; i++) { + if (num_pins >= 3) { + snprintf(name, sizeof(name), "%s %s", + pfx, channel_name[i]); + err = alc_auto_create_extra_out(codec, pins[i], dacs[i], + name, 0); + } else { + err = alc_auto_create_extra_out(codec, pins[i], dacs[i], + pfx, i); + } + if (err < 0) + return err; } + return 0; + } + + /* Let's create a bind-controls */ + ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_sw); + if (!ctl) + return -ENOMEM; + n = 0; + for (i = 0; i < num_pins; i++) { + if (get_wcaps(codec, pins[i]) & AC_WCAP_OUT_AMP) + ctl->values[n++] = + HDA_COMPOSE_AMP_VAL(pins[i], 3, 0, HDA_OUTPUT); + } + if (n) { + snprintf(name, sizeof(name), "%s Playback Switch", pfx); + err = add_control(spec, ALC_CTL_BIND_SW, name, 0, (long)ctl); if (err < 0) return err; } - if (dacs[num_pins - 1]) - return 0; - /* Let's create a bind-controls for volumes */ ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_vol); if (!ctl) return -ENOMEM; @@ -3849,111 +3543,58 @@ static void alc_auto_init_extra_out(struct hda_codec *codec) } } -/* check whether the given pin can be a multi-io pin */ -static bool can_be_multiio_pin(struct hda_codec *codec, - unsigned int location, hda_nid_t nid) -{ - unsigned int defcfg, caps; - - defcfg = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX) - return false; - if (location && get_defcfg_location(defcfg) != location) - return false; - caps = snd_hda_query_pin_caps(codec, nid); - if (!(caps & AC_PINCAP_OUT)) - return false; - return true; -} - /* * multi-io helper - * - * When hardwired is set, try to fill ony hardwired pins, and returns - * zero if any pins are filled, non-zero if nothing found. - * When hardwired is off, try to fill possible input pins, and returns - * the badness value. */ static int alc_auto_fill_multi_ios(struct hda_codec *codec, - hda_nid_t reference_pin, - bool hardwired, int offset) + unsigned int location, + int offset) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int type, i, j, dacs, num_pins, old_pins; - unsigned int defcfg = snd_hda_codec_get_pincfg(codec, reference_pin); - unsigned int location = get_defcfg_location(defcfg); - int badness = 0; - - old_pins = spec->multi_ios; - if (old_pins >= 2) - goto end_fill; + hda_nid_t prime_dac = spec->private_dac_nids[0]; + int type, i, dacs, num_pins = 0; - num_pins = 0; + dacs = spec->multiout.num_dacs; for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + hda_nid_t dac = 0; + unsigned int defcfg, caps; if (cfg->inputs[i].type != type) continue; - if (can_be_multiio_pin(codec, location, - cfg->inputs[i].pin)) - num_pins++; - } - } - if (num_pins < 2) - goto end_fill; - - dacs = spec->multiout.num_dacs; - for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - hda_nid_t dac = 0; - - if (cfg->inputs[i].type != type) + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX) continue; - if (!can_be_multiio_pin(codec, location, nid)) + if (location && get_defcfg_location(defcfg) != location) continue; - for (j = 0; j < spec->multi_ios; j++) { - if (nid == spec->multi_io[j].pin) - break; - } - if (j < spec->multi_ios) + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_OUT)) continue; - - if (offset && offset + spec->multi_ios < dacs) { - dac = spec->private_dac_nids[offset + spec->multi_ios]; + if (offset && offset + num_pins < dacs) { + dac = spec->private_dac_nids[offset + num_pins]; if (!alc_auto_is_dac_reachable(codec, nid, dac)) dac = 0; } - if (hardwired) - dac = get_dac_if_single(codec, nid); - else if (!dac) + if (!dac) dac = alc_auto_look_for_dac(codec, nid); - if (!dac) { - badness++; + if (!dac) continue; - } - spec->multi_io[spec->multi_ios].pin = nid; - spec->multi_io[spec->multi_ios].dac = dac; - spec->multi_ios++; - if (spec->multi_ios >= 2) - break; + spec->multi_io[num_pins].pin = nid; + spec->multi_io[num_pins].dac = dac; + num_pins++; + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } } - end_fill: - if (badness) - badness = BAD_MULTI_IO; - if (old_pins == spec->multi_ios) { - if (hardwired) - return 1; /* nothing found */ - else - return badness; /* no badness if nothing found */ - } - if (!hardwired && spec->multi_ios < 2) { - spec->multi_ios = old_pins; - return badness; + spec->multiout.num_dacs = dacs; + if (num_pins < 2) { + /* clear up again */ + memset(spec->private_dac_nids + dacs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - dacs)); + spec->private_dac_nids[0] = prime_dac; + return 0; } - - return 0; + return num_pins; } static int alc_auto_ch_mode_info(struct snd_kcontrol *kcontrol, @@ -4073,7 +3714,6 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) if (spec->dyn_adc_switch) return; - again: nums = 0; for (n = 0; n < spec->num_adc_nids; n++) { hda_nid_t cap = spec->private_capsrc_nids[n]; @@ -4094,11 +3734,6 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) if (!nums) { /* check whether ADC-switch is possible */ if (!alc_check_dyn_adc_switch(codec)) { - if (spec->shared_mic_hp) { - spec->shared_mic_hp = 0; - spec->private_imux[0].num_items = 1; - goto again; - } printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" " using fallback 0x%x\n", codec->chip_name, spec->private_adc_nids[0]); @@ -4116,7 +3751,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) if (spec->auto_mic) alc_auto_mic_check_imux(codec); /* check auto-mic setups */ - else if (spec->input_mux->num_items == 1 || spec->shared_mic_hp) + else if (spec->input_mux->num_items == 1) spec->num_adc_nids = 1; /* reduce to a single ADC */ } @@ -4157,7 +3792,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec) else nums = spec->num_adc_nids; for (c = 0; c < nums; c++) - alc_mux_select(codec, c, spec->cur_mux[c], true); + alc_mux_select(codec, 0, spec->cur_mux[c], true); } /* add mic boosts if needed */ @@ -4314,7 +3949,6 @@ static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1), - SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; @@ -4414,9 +4048,6 @@ static int alc_parse_auto_config(struct hda_codec *codec, if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - return 1; } @@ -4427,47 +4058,26 @@ static int alc880_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static const struct hda_amp_list alc880_loopbacks[] = { + { 0x0b, HDA_INPUT, 0 }, + { 0x0b, HDA_INPUT, 1 }, + { 0x0b, HDA_INPUT, 2 }, + { 0x0b, HDA_INPUT, 3 }, + { 0x0b, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + /* * ALC880 fix-ups */ enum { - ALC880_FIXUP_GPIO1, ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, - ALC880_FIXUP_LG, - ALC880_FIXUP_W810, - ALC880_FIXUP_EAPD_COEF, - ALC880_FIXUP_TCL_S700, - ALC880_FIXUP_VOL_KNOB, - ALC880_FIXUP_FUJITSU, - ALC880_FIXUP_F1734, - ALC880_FIXUP_UNIWILL, - ALC880_FIXUP_UNIWILL_DIG, - ALC880_FIXUP_Z71V, - ALC880_FIXUP_3ST_BASE, - ALC880_FIXUP_3ST, - ALC880_FIXUP_3ST_DIG, - ALC880_FIXUP_5ST_BASE, - ALC880_FIXUP_5ST, - ALC880_FIXUP_5ST_DIG, - ALC880_FIXUP_6ST_BASE, - ALC880_FIXUP_6ST, - ALC880_FIXUP_6ST_DIG, }; -/* enable the volume-knob widget support on NID 0x21 */ -static void alc880_fixup_vol_knob(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - if (action == ALC_FIXUP_ACT_PROBE) - snd_hda_jack_detect_enable(codec, 0x21, ALC_DCVOL_EVENT); -} - static const struct alc_fixup alc880_fixups[] = { - [ALC880_FIXUP_GPIO1] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, - }, [ALC880_FIXUP_GPIO2] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio2_init_verbs, @@ -4482,323 +4092,40 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, - [ALC880_FIXUP_LG] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - /* disable bogus unused pins */ - { 0x16, 0x411111f0 }, - { 0x18, 0x411111f0 }, - { 0x1a, 0x411111f0 }, - { } - } - }, - [ALC880_FIXUP_W810] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - /* disable bogus unused pins */ - { 0x17, 0x411111f0 }, - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_GPIO2, - }, - [ALC880_FIXUP_EAPD_COEF] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* change to EAPD mode */ - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x3060 }, - {} - }, - }, - [ALC880_FIXUP_TCL_S700] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* change to EAPD mode */ - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 }, - {} - }, - .chained = true, - .chain_id = ALC880_FIXUP_GPIO2, - }, - [ALC880_FIXUP_VOL_KNOB] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc880_fixup_vol_knob, - }, - [ALC880_FIXUP_FUJITSU] = { - /* override all pins as BIOS on old Amilo is broken */ - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x0121411f }, /* HP */ - { 0x15, 0x99030120 }, /* speaker */ - { 0x16, 0x99030130 }, /* bass speaker */ - { 0x17, 0x411111f0 }, /* N/A */ - { 0x18, 0x411111f0 }, /* N/A */ - { 0x19, 0x01a19950 }, /* mic-in */ - { 0x1a, 0x411111f0 }, /* N/A */ - { 0x1b, 0x411111f0 }, /* N/A */ - { 0x1c, 0x411111f0 }, /* N/A */ - { 0x1d, 0x411111f0 }, /* N/A */ - { 0x1e, 0x01454140 }, /* SPDIF out */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_VOL_KNOB, - }, - [ALC880_FIXUP_F1734] = { - /* almost compatible with FUJITSU, but no bass and SPDIF */ - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x0121411f }, /* HP */ - { 0x15, 0x99030120 }, /* speaker */ - { 0x16, 0x411111f0 }, /* N/A */ - { 0x17, 0x411111f0 }, /* N/A */ - { 0x18, 0x411111f0 }, /* N/A */ - { 0x19, 0x01a19950 }, /* mic-in */ - { 0x1a, 0x411111f0 }, /* N/A */ - { 0x1b, 0x411111f0 }, /* N/A */ - { 0x1c, 0x411111f0 }, /* N/A */ - { 0x1d, 0x411111f0 }, /* N/A */ - { 0x1e, 0x411111f0 }, /* N/A */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_VOL_KNOB, - }, - [ALC880_FIXUP_UNIWILL] = { - /* need to fix HP and speaker pins to be parsed correctly */ - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x0121411f }, /* HP */ - { 0x15, 0x99030120 }, /* speaker */ - { 0x16, 0x99030130 }, /* bass speaker */ - { } - }, - }, - [ALC880_FIXUP_UNIWILL_DIG] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - /* disable bogus unused pins */ - { 0x17, 0x411111f0 }, - { 0x19, 0x411111f0 }, - { 0x1b, 0x411111f0 }, - { 0x1f, 0x411111f0 }, - { } - } - }, - [ALC880_FIXUP_Z71V] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - /* set up the whole pins as BIOS is utterly broken */ - { 0x14, 0x99030120 }, /* speaker */ - { 0x15, 0x0121411f }, /* HP */ - { 0x16, 0x411111f0 }, /* N/A */ - { 0x17, 0x411111f0 }, /* N/A */ - { 0x18, 0x01a19950 }, /* mic-in */ - { 0x19, 0x411111f0 }, /* N/A */ - { 0x1a, 0x01813031 }, /* line-in */ - { 0x1b, 0x411111f0 }, /* N/A */ - { 0x1c, 0x411111f0 }, /* N/A */ - { 0x1d, 0x411111f0 }, /* N/A */ - { 0x1e, 0x0144111e }, /* SPDIF */ - { } - } - }, - [ALC880_FIXUP_3ST_BASE] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x01014010 }, /* line-out */ - { 0x15, 0x411111f0 }, /* N/A */ - { 0x16, 0x411111f0 }, /* N/A */ - { 0x17, 0x411111f0 }, /* N/A */ - { 0x18, 0x01a19c30 }, /* mic-in */ - { 0x19, 0x0121411f }, /* HP */ - { 0x1a, 0x01813031 }, /* line-in */ - { 0x1b, 0x02a19c40 }, /* front-mic */ - { 0x1c, 0x411111f0 }, /* N/A */ - { 0x1d, 0x411111f0 }, /* N/A */ - /* 0x1e is filled in below */ - { 0x1f, 0x411111f0 }, /* N/A */ - { } - } - }, - [ALC880_FIXUP_3ST] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1e, 0x411111f0 }, /* N/A */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_3ST_BASE, - }, - [ALC880_FIXUP_3ST_DIG] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1e, 0x0144111e }, /* SPDIF */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_3ST_BASE, - }, - [ALC880_FIXUP_5ST_BASE] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x01014010 }, /* front */ - { 0x15, 0x411111f0 }, /* N/A */ - { 0x16, 0x01011411 }, /* CLFE */ - { 0x17, 0x01016412 }, /* surr */ - { 0x18, 0x01a19c30 }, /* mic-in */ - { 0x19, 0x0121411f }, /* HP */ - { 0x1a, 0x01813031 }, /* line-in */ - { 0x1b, 0x02a19c40 }, /* front-mic */ - { 0x1c, 0x411111f0 }, /* N/A */ - { 0x1d, 0x411111f0 }, /* N/A */ - /* 0x1e is filled in below */ - { 0x1f, 0x411111f0 }, /* N/A */ - { } - } - }, - [ALC880_FIXUP_5ST] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1e, 0x411111f0 }, /* N/A */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_5ST_BASE, - }, - [ALC880_FIXUP_5ST_DIG] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1e, 0x0144111e }, /* SPDIF */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_5ST_BASE, - }, - [ALC880_FIXUP_6ST_BASE] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x01014010 }, /* front */ - { 0x15, 0x01016412 }, /* surr */ - { 0x16, 0x01011411 }, /* CLFE */ - { 0x17, 0x01012414 }, /* side */ - { 0x18, 0x01a19c30 }, /* mic-in */ - { 0x19, 0x02a19c40 }, /* front-mic */ - { 0x1a, 0x01813031 }, /* line-in */ - { 0x1b, 0x0121411f }, /* HP */ - { 0x1c, 0x411111f0 }, /* N/A */ - { 0x1d, 0x411111f0 }, /* N/A */ - /* 0x1e is filled in below */ - { 0x1f, 0x411111f0 }, /* N/A */ - { } - } - }, - [ALC880_FIXUP_6ST] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1e, 0x411111f0 }, /* N/A */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_6ST_BASE, - }, - [ALC880_FIXUP_6ST_DIG] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1e, 0x0144111e }, /* SPDIF */ - { } - }, - .chained = true, - .chain_id = ALC880_FIXUP_6ST_BASE, - }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), - SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), - SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), - SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), - SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), - SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), - SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), - SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_FIXUP_UNIWILL), - SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_FIXUP_VOL_KNOB), - SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), - SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734), - SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_FIXUP_F1734), - SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU), - SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), - SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), - SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), - SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), - - /* Below is the copied entries from alc880_quirks.c. - * It's not quite sure whether BIOS sets the correct pin-config table - * on these machines, thus they are kept to be compatible with - * the old static quirks. Once when it's confirmed to work without - * these overrides, it'd be better to remove. - */ - SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_FIXUP_6ST), - SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_FIXUP_3ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_FIXUP_3ST_DIG), - SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_FIXUP_3ST), - SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_FIXUP_3ST), - SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_FIXUP_3ST), - SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_FIXUP_5ST), - SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_FIXUP_5ST), - SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_FIXUP_5ST), - SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_FIXUP_6ST_DIG), /* broken BIOS */ - SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_FIXUP_6ST_DIG), - SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_FIXUP_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_FIXUP_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_FIXUP_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_FIXUP_5ST_DIG), - /* default Intel */ - SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_FIXUP_3ST), - SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_FIXUP_5ST_DIG), - SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_FIXUP_6ST_DIG), {} }; -static const struct alc_model_fixup alc880_fixup_models[] = { - {.id = ALC880_FIXUP_3ST, .name = "3stack"}, - {.id = ALC880_FIXUP_3ST_DIG, .name = "3stack-digout"}, - {.id = ALC880_FIXUP_5ST, .name = "5stack"}, - {.id = ALC880_FIXUP_5ST_DIG, .name = "5stack-digout"}, - {.id = ALC880_FIXUP_6ST, .name = "6stack"}, - {.id = ALC880_FIXUP_6ST_DIG, .name = "6stack-digout"}, - {} -}; +/* + * board setups + */ +#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS +#define alc_board_config \ + snd_hda_check_board_config +#define alc_board_codec_sid_config \ + snd_hda_check_board_codec_sid_config +#include "alc_quirks.c" +#else +#define alc_board_config(codec, nums, models, tbl) -1 +#define alc_board_codec_sid_config(codec, nums, models, tbl) -1 +#define setup_preset(codec, x) /* NOP */ +#endif /* * OK, here we have finally the patch for ALC880 */ +#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS +#include "alc880_quirks.c" +#endif + static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; + int board_config; int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -4810,14 +4137,47 @@ static int patch_alc880(struct hda_codec *codec) spec->mixer_nid = 0x0b; spec->need_dac_fix = 1; - alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, - alc880_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + board_config = alc_board_config(codec, ALC880_MODEL_LAST, + alc880_models, alc880_cfg_tbl); + if (board_config < 0) { + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + board_config = ALC_MODEL_AUTO; + } - /* automatic parse from the BIOS config */ - err = alc880_parse_auto_config(codec); - if (err < 0) - goto error; + if (board_config == ALC_MODEL_AUTO) { + alc_pick_fixup(codec, NULL, alc880_fixup_tbl, alc880_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } + + if (board_config == ALC_MODEL_AUTO) { + /* automatic parse from the BIOS config */ + err = alc880_parse_auto_config(codec); + if (err < 0) + goto error; +#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS + else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using 3-stack mode...\n"); + board_config = ALC880_3ST; + } +#endif + } + + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; + setup_preset(codec, &alc880_presets[board_config]); + } + + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x1); @@ -4826,10 +4186,18 @@ static int patch_alc880(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - codec->patch_ops = alc_patch_ops; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + if (board_config == ALC_MODEL_AUTO) + spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc880_loopbacks; +#endif + return 0; error: @@ -4848,115 +4216,49 @@ static int alc260_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc260_ignore, alc260_ssids); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static const struct hda_amp_list alc260_loopbacks[] = { + { 0x07, HDA_INPUT, 0 }, + { 0x07, HDA_INPUT, 1 }, + { 0x07, HDA_INPUT, 2 }, + { 0x07, HDA_INPUT, 3 }, + { 0x07, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + /* * Pin config fixes */ enum { - ALC260_FIXUP_HP_DC5750, - ALC260_FIXUP_HP_PIN_0F, - ALC260_FIXUP_COEF, - ALC260_FIXUP_GPIO1, - ALC260_FIXUP_GPIO1_TOGGLE, - ALC260_FIXUP_REPLACER, - ALC260_FIXUP_HP_B1900, + PINFIX_HP_DC5750, }; -static void alc260_gpio1_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->hp_jack_present); -} - -static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - if (action == ALC_FIXUP_ACT_PROBE) { - /* although the machine has only one output pin, we need to - * toggle GPIO1 according to the jack state - */ - spec->automute_hook = alc260_gpio1_automute; - spec->detect_hp = 1; - spec->automute_speaker = 1; - spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ - snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); - spec->unsol_event = alc_sku_unsol_event; - add_verb(codec->spec, alc_gpio1_init_verbs); - } -} - static const struct alc_fixup alc260_fixups[] = { - [ALC260_FIXUP_HP_DC5750] = { + [PINFIX_HP_DC5750] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x11, 0x90130110 }, /* speaker */ { } } }, - [ALC260_FIXUP_HP_PIN_0F] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x0f, 0x01214000 }, /* HP */ - { } - } - }, - [ALC260_FIXUP_COEF] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x3040 }, - { } - }, - .chained = true, - .chain_id = ALC260_FIXUP_HP_PIN_0F, - }, - [ALC260_FIXUP_GPIO1] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, - }, - [ALC260_FIXUP_GPIO1_TOGGLE] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc260_fixup_gpio1_toggle, - .chained = true, - .chain_id = ALC260_FIXUP_HP_PIN_0F, - }, - [ALC260_FIXUP_REPLACER] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, - { } - }, - .chained = true, - .chain_id = ALC260_FIXUP_GPIO1_TOGGLE, - }, - [ALC260_FIXUP_HP_B1900] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc260_fixup_gpio1_toggle, - .chained = true, - .chain_id = ALC260_FIXUP_COEF, - } }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_FIXUP_GPIO1), - SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), - SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), - SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), - SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), - SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), - SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), - SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), + SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750), {} }; /* */ +#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS +#include "alc260_quirks.c" +#endif + static int patch_alc260(struct hda_codec *codec) { struct alc_spec *spec; - int err; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4966,13 +4268,47 @@ static int patch_alc260(struct hda_codec *codec) spec->mixer_nid = 0x07; - alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + board_config = alc_board_config(codec, ALC260_MODEL_LAST, + alc260_models, alc260_cfg_tbl); + if (board_config < 0) { + snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + board_config = ALC_MODEL_AUTO; + } - /* automatic parse from the BIOS config */ - err = alc260_parse_auto_config(codec); - if (err < 0) - goto error; + if (board_config == ALC_MODEL_AUTO) { + alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } + + if (board_config == ALC_MODEL_AUTO) { + /* automatic parse from the BIOS config */ + err = alc260_parse_auto_config(codec); + if (err < 0) + goto error; +#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS + else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC260_BASIC; + } +#endif + } + + if (board_config != ALC_MODEL_AUTO) { + setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } + + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x1); @@ -4981,10 +4317,18 @@ static int patch_alc260(struct hda_codec *codec) set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + if (board_config == ALC_MODEL_AUTO) + spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; spec->shutup = alc_eapd_shutup; - - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc260_loopbacks; +#endif return 0; @@ -5005,6 +4349,9 @@ static int patch_alc260(struct hda_codec *codec) * In addition, an independent DAC for the multi-playback (not used in this * driver yet). */ +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc882_loopbacks alc880_loopbacks +#endif /* * Pin config fixes @@ -5015,14 +4362,11 @@ enum { ALC882_FIXUP_PB_M5210, ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, - ALC889_FIXUP_CD, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, - ALC882_FIXUP_GPIO1, - ALC882_FIXUP_GPIO2, ALC882_FIXUP_GPIO3, ALC889_FIXUP_COEF, ALC882_FIXUP_ASUS_W2JC, @@ -5031,8 +4375,6 @@ enum { ALC882_FIXUP_ASPIRE_8930G_VERBS, ALC885_FIXUP_MACPRO_GPIO, ALC889_FIXUP_DAC_ROUTE, - ALC889_FIXUP_MBP_VREF, - ALC889_FIXUP_IMAC91_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -5094,66 +4436,13 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { - /* fake the connections during parsing the tree */ hda_nid_t conn1[2] = { 0x0c, 0x0d }; hda_nid_t conn2[2] = { 0x0e, 0x0f }; snd_hda_override_conn_list(codec, 0x14, 2, conn1); snd_hda_override_conn_list(codec, 0x15, 2, conn1); snd_hda_override_conn_list(codec, 0x18, 2, conn2); snd_hda_override_conn_list(codec, 0x1a, 2, conn2); - } else if (action == ALC_FIXUP_ACT_PROBE) { - /* restore the connections */ - hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; - snd_hda_override_conn_list(codec, 0x14, 5, conn); - snd_hda_override_conn_list(codec, 0x15, 5, conn); - snd_hda_override_conn_list(codec, 0x18, 5, conn); - snd_hda_override_conn_list(codec, 0x1a, 5, conn); - } -} - -/* Set VREF on HP pin */ -static void alc889_fixup_mbp_vref(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x14, 0x15 }; - int i; - - if (action != ALC_FIXUP_ACT_INIT) - return; - for (i = 0; i < ARRAY_SIZE(nids); i++) { - unsigned int val = snd_hda_codec_get_pincfg(codec, nids[i]); - if (get_defcfg_device(val) != AC_JACK_HP_OUT) - continue; - val = snd_hda_codec_read(codec, nids[i], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - val |= AC_PINCTL_VREF_80; - snd_hda_codec_write(codec, nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); - spec->keep_vref_in_automute = 1; - break; - } -} - -/* Set VREF on speaker pins on imac91 */ -static void alc889_fixup_imac91_vref(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x18, 0x1a }; - int i; - - if (action != ALC_FIXUP_ACT_INIT) - return; - for (i = 0; i < ARRAY_SIZE(nids); i++) { - unsigned int val; - val = snd_hda_codec_read(codec, nids[i], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - val |= AC_PINCTL_VREF_50; - snd_hda_codec_write(codec, nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); } - spec->keep_vref_in_automute = 1; } static const struct alc_fixup alc882_fixups[] = { @@ -5192,13 +4481,6 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, - [ALC889_FIXUP_CD] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x1c, 0x993301f0 }, /* CD */ - { } - } - }, [ALC889_FIXUP_VAIO_TT] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { @@ -5241,14 +4523,6 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, - [ALC882_FIXUP_GPIO1] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, - }, - [ALC882_FIXUP_GPIO2] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = alc_gpio2_init_verbs, - }, [ALC882_FIXUP_GPIO3] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio3_init_verbs, @@ -5322,18 +4596,6 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc889_fixup_dac_route, }, - [ALC889_FIXUP_MBP_VREF] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc889_fixup_mbp_vref, - .chained = true, - .chain_id = ALC882_FIXUP_GPIO1, - }, - [ALC889_FIXUP_IMAC91_VREF] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc889_fixup_imac91_vref, - .chained = true, - .chain_id = ALC882_FIXUP_GPIO1, - }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -5367,30 +4629,14 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), /* All Apple entries are in codec SSIDs */ - SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), - SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), - SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), - SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), @@ -5412,10 +4658,14 @@ static int alc882_parse_auto_config(struct hda_codec *codec) /* */ +#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS +#include "alc882_quirks.c" +#endif + static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; - int err; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5439,15 +4689,45 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + board_config = alc_board_config(codec, ALC882_MODEL_LAST, + alc882_models, NULL); + if (board_config < 0) + board_config = alc_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl); + + if (board_config < 0) { + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + board_config = ALC_MODEL_AUTO; + } + + if (board_config == ALC_MODEL_AUTO) { + alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } alc_auto_parse_customize_define(codec); - /* automatic parse from the BIOS config */ - err = alc882_parse_auto_config(codec); - if (err < 0) - goto error; + if (board_config == ALC_MODEL_AUTO) { + /* automatic parse from the BIOS config */ + err = alc882_parse_auto_config(codec); + if (err < 0) + goto error; + } + + if (board_config != ALC_MODEL_AUTO) { + setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } + + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); @@ -5456,9 +4736,18 @@ static int patch_alc882(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + if (board_config == ALC_MODEL_AUTO) + spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc882_loopbacks; +#endif return 0; @@ -5554,6 +4843,10 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc262_loopbacks alc880_loopbacks +#endif + /* */ static int patch_alc262(struct hda_codec *codec) @@ -5593,6 +4886,15 @@ static int patch_alc262(struct hda_codec *codec) if (err < 0) goto error; + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -5600,10 +4902,16 @@ static int patch_alc262(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc262_loopbacks; +#endif return 0; @@ -5697,7 +5005,17 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + codec->patch_ops = alc_patch_ops; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; return 0; @@ -5710,6 +5028,10 @@ static int patch_alc268(struct hda_codec *codec) /* * ALC269 */ +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc269_loopbacks alc880_loopbacks +#endif + static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = { .substreams = 1, .channels_min = 2, @@ -5731,6 +5053,35 @@ static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = { /* NID is set in alc_build_pcms */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int alc269_mic2_for_mute_led(struct hda_codec *codec) +{ + switch (codec->subsystem_id) { + case 0x103c1586: + return 1; + } + return 0; +} + +static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid) +{ + /* update mute-LED according to the speaker mute state */ + if (nid == 0x01 || nid == 0x14) { + int pinval; + if (snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + pinval = 0x24; + else + pinval = 0x20; + /* mic2 vref pin is used for mute LED control */ + snd_hda_codec_update_cache(codec, 0x19, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinval); + } + return alc_check_power_status(codec, nid); +} +#endif /* CONFIG_SND_HDA_POWER_SAVE */ + /* different alc269-variants */ enum { ALC269_TYPE_ALC269VA, @@ -5881,31 +5232,6 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } -/* update mute-LED according to the speaker mute state via mic2 VREF pin */ -static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) -{ - struct hda_codec *codec = private_data; - unsigned int pinval = enabled ? 0x20 : 0x24; - snd_hda_codec_update_cache(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinval); -} - -static void alc269_fixup_mic2_mute(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - switch (action) { - case ALC_FIXUP_ACT_BUILD: - spec->vmaster_mute.hook = alc269_fixup_mic2_mute_hook; - snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); - /* fallthru */ - case ALC_FIXUP_ACT_INIT: - snd_hda_sync_vmaster_hook(&spec->vmaster_mute); - break; - } -} - enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5923,7 +5249,6 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, - ALC269_FIXUP_MIC2_MUTE_LED, }; static const struct alc_fixup alc269_fixups[] = { @@ -6044,14 +5369,9 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, - [ALC269_FIXUP_MIC2_MUTE_LED] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc269_fixup_mic2_mute, - }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), @@ -6074,7 +5394,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -#if 0 +#if 1 /* Below is a quirk table taken from the old code. * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding @@ -6132,14 +5452,10 @@ static const struct alc_model_fixup alc269_fixup_models[] = { }; -static void alc269_fill_coef(struct hda_codec *codec) +static int alc269_fill_coef(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; int val; - if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return; - if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); alc_write_coef_idx(codec, 0xe, 0x8817); @@ -6174,6 +5490,8 @@ static void alc269_fill_coef(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x4); /* HP */ alc_write_coef_idx(codec, 0x4, val | (1<<11)); + + return 0; } /* @@ -6217,7 +5535,6 @@ static int patch_alc269(struct hda_codec *codec) } if (err < 0) goto error; - spec->init_hook = alc269_fill_coef; alc269_fill_coef(codec); } @@ -6230,6 +5547,15 @@ static int patch_alc269(struct hda_codec *codec) if (err < 0) goto error; + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -6237,13 +5563,21 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; #endif + spec->init_hook = alc_auto_init_std; spec->shutup = alc269_shutup; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc269_loopbacks; + if (alc269_mic2_for_mute_led(codec)) + codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps; +#endif return 0; @@ -6263,12 +5597,21 @@ static int alc861_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc861_ignore, alc861_ssids); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static const struct hda_amp_list alc861_loopbacks[] = { + { 0x15, HDA_INPUT, 0 }, + { 0x15, HDA_INPUT, 1 }, + { 0x15, HDA_INPUT, 2 }, + { 0x15, HDA_INPUT, 3 }, + { } /* end */ +}; +#endif + + /* Pin config fixes */ enum { - ALC861_FIXUP_FSC_AMILO_PI1505, - ALC861_FIXUP_AMP_VREF_0F, - ALC861_FIXUP_NO_JACK_DETECT, - ALC861_FIXUP_ASUS_A6RP, + PINFIX_FSC_AMILO_PI1505, + PINFIX_ASUS_A6RP, }; /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ @@ -6290,16 +5633,8 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, spec->keep_vref_in_automute = 1; } -/* suppress the jack-detection */ -static void alc_fixup_no_jack_detect(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - if (action == ALC_FIXUP_ACT_PRE_PROBE) - codec->no_jack_detect = 1; -} - static const struct alc_fixup alc861_fixups[] = { - [ALC861_FIXUP_FSC_AMILO_PI1505] = { + [PINFIX_FSC_AMILO_PI1505] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x0b, 0x0221101f }, /* HP */ @@ -6307,29 +5642,17 @@ static const struct alc_fixup alc861_fixups[] = { { } } }, - [ALC861_FIXUP_AMP_VREF_0F] = { + [PINFIX_ASUS_A6RP] = { .type = ALC_FIXUP_FUNC, .v.func = alc861_fixup_asus_amp_vref_0f, }, - [ALC861_FIXUP_NO_JACK_DETECT] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc_fixup_no_jack_detect, - }, - [ALC861_FIXUP_ASUS_A6RP] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc861_fixup_asus_amp_vref_0f, - .chained = true, - .chain_id = ALC861_FIXUP_NO_JACK_DETECT, - } }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), - SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), - SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), - SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F), - SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F), - SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} }; @@ -6356,6 +5679,15 @@ static int patch_alc861(struct hda_codec *codec) if (err < 0) goto error; + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); if (err < 0) @@ -6363,13 +5695,16 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->power_hook = alc_power_eapd; + if (!spec->loopback.amplist) + spec->loopback.amplist = alc861_loopbacks; #endif - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - return 0; error: @@ -6384,6 +5719,10 @@ static int patch_alc861(struct hda_codec *codec) * * In addition, an independent DAC */ +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc861vd_loopbacks alc880_loopbacks +#endif + static int alc861vd_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 }; @@ -6464,6 +5803,15 @@ static int patch_alc861vd(struct hda_codec *codec) add_verb(spec, alc660vd_eapd_verbs); } + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); if (err < 0) @@ -6471,11 +5819,16 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; - - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc861vd_loopbacks; +#endif return 0; @@ -6495,6 +5848,9 @@ static int patch_alc861vd(struct hda_codec *codec) * In addition, an independent DAC for the multi-playback (not used in this * driver yet). */ +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc662_loopbacks alc880_loopbacks +#endif /* * BIOS auto configuration @@ -6544,7 +5900,6 @@ enum { ALC662_FIXUP_ASUS_MODE6, ALC662_FIXUP_ASUS_MODE7, ALC662_FIXUP_ASUS_MODE8, - ALC662_FIXUP_NO_JACK_DETECT, }; static const struct alc_fixup alc662_fixups[] = { @@ -6690,10 +6045,6 @@ static const struct alc_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_SKU_IGNORE }, - [ALC662_FIXUP_NO_JACK_DETECT] = { - .type = ALC_FIXUP_FUNC, - .v.func = alc_fixup_no_jack_detect, - }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6702,7 +6053,6 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), - SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), @@ -6824,6 +6174,15 @@ static int patch_alc662(struct hda_codec *codec) if (err < 0) goto error; + if (!spec->no_analog && !spec->adc_nids) { + alc_auto_fill_adc_caps(codec); + alc_rebuild_imux_for_auto_mic(codec); + alc_remove_invalid_adc_nids(codec); + } + + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -6843,10 +6202,16 @@ static int patch_alc662(struct hda_codec *codec) } } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc662_loopbacks; +#endif return 0; @@ -6886,7 +6251,11 @@ static int patch_alc680(struct hda_codec *codec) return err; } + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + codec->patch_ops = alc_patch_ops; + spec->init_hook = alc_auto_init_std; return 0; } diff --git a/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c index 33a9946b492c..6345df131a00 100644 --- a/trunk/sound/pci/hda/patch_sigmatel.c +++ b/trunk/sound/pci/hda/patch_sigmatel.c @@ -99,7 +99,6 @@ enum { STAC_DELL_VOSTRO_3500, STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, - STAC_HP_ZEPHYR, STAC_92HD83XXX_MODELS }; @@ -310,8 +309,6 @@ struct sigmatel_spec { unsigned long auto_capvols[MAX_ADCS_NUM]; unsigned auto_dmic_cnt; hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; - - struct hda_vmaster_mute_hook vmaster_mute; }; static const hda_nid_t stac9200_adc_nids[1] = { @@ -665,6 +662,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac_vrefout_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) { @@ -688,6 +686,7 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } +#endif static unsigned int stac92xx_vref_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) @@ -895,13 +894,6 @@ static const struct hda_verb stac92hd83xxx_core_init[] = { {} }; -static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = { - { 0x22, 0x785, 0x43 }, - { 0x22, 0x782, 0xe0 }, - { 0x22, 0x795, 0x00 }, - {} -}; - static const struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -1007,8 +999,8 @@ static const struct hda_verb stac9205_core_init[] = { } static const struct snd_kcontrol_new stac9200_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), { } /* end */ @@ -1035,8 +1027,8 @@ static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = { }; static const struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), { } /* end */ }; @@ -1068,25 +1060,34 @@ static struct snd_kcontrol_new stac_smux_mixer = { .put = stac92xx_smux_enum_put, }; -static const char * const slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "IEC958", +static const char * const slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Speaker Playback Volume", NULL }; -static void stac92xx_update_led_status(struct hda_codec *codec, int enabled); - -static void stac92xx_vmaster_hook(void *private_data, int val) -{ - stac92xx_update_led_status(private_data, val); -} +static const char * const slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Speaker Playback Switch", + "IEC958 Playback Switch", + NULL +}; static void stac92xx_free_kctls(struct hda_codec *codec); static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int vmaster_tlv[4]; int err; int i; @@ -1143,28 +1144,22 @@ static int stac92xx_build_controls(struct hda_codec *codec) } /* if we have no master control, let's create it */ - snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], - HDA_OUTPUT, vmaster_tlv); - /* correct volume offset */ - vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; - /* minimum value is actually mute */ - vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; - - err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch", true, - &spec->vmaster_mute.sw_kctl); - if (err < 0) - return err; - - if (spec->gpio_led) { - spec->vmaster_mute.hook = stac92xx_vmaster_hook; - err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; + snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], + HDA_OUTPUT, vmaster_tlv); + /* correct volume offset */ + vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; + /* minimum value is actually mute */ + vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + vmaster_tlv, slave_vols); + if (err < 0) + return err; + } + if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_sws); if (err < 0) return err; } @@ -1641,12 +1636,6 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = { 0x40f000f0, 0x40f000f0, }; -static const unsigned int hp_zephyr_pin_configs[10] = { - 0x01813050, 0x0421201f, 0x04a1205e, 0x96130310, - 0x96130310, 0x0101401f, 0x1111611f, 0xd5a30130, - 0, 0, -}; - static const unsigned int hp_cNB11_intquad_pin_configs[10] = { 0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110, 0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130, @@ -1660,7 +1649,6 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs, [STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs, [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, - [STAC_HP_ZEPHYR] = hp_zephyr_pin_configs, }; static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { @@ -1671,7 +1659,6 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_DELL_VOSTRO_3500] = "dell-vostro-3500", [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", - [STAC_HP_ZEPHYR] = "hp-zephyr", }; static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1724,14 +1711,6 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561, - "HP", STAC_HP_ZEPHYR), - {} /* terminator */ -}; - -static const struct snd_pci_quirk stac92hd83xxx_codec_id_cfg_tbl[] = { - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561, - "HP", STAC_HP_ZEPHYR), {} /* terminator */ }; @@ -4431,7 +4410,8 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_jack_report_sync(codec); /* sync mute LED */ - snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + if (spec->gpio_led) + hda_call_check_power_status(codec, 0x01); if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -4649,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (1 /*presence*/) + if (presence) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions @@ -5009,6 +4989,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac92xx_pre_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -5043,41 +5024,83 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, afg_power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } -#else -#define stac92xx_suspend NULL -#define stac92xx_resume NULL -#define stac92xx_pre_resume NULL -#define stac92xx_set_power_state NULL -#endif /* CONFIG_PM */ -/* update mute-LED accoring to the master switch */ -static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) +/* + * For this feature CONFIG_SND_HDA_POWER_SAVE is needed + * as mute LED state is updated in check_power_status hook + */ +static int stac92xx_update_led_status(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int muted = !enabled; + int i, num_ext_dacs, muted = 1; + unsigned int muted_lvl, notmtd_lvl; + hda_nid_t nid; if (!spec->gpio_led) - return; - - /* LED state is inverted on these systems */ - if (spec->gpio_led_polarity) - muted = !muted; + return 0; + for (i = 0; i < spec->multiout.num_dacs; i++) { + nid = spec->multiout.dac_nids[i]; + if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* something heard */ + break; + } + } + if (muted && spec->multiout.hp_nid) + if (!(snd_hda_codec_amp_read(codec, + spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* HP is not muted */ + } + num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid); + for (i = 0; muted && i < num_ext_dacs; i++) { + nid = spec->multiout.extra_out_nid[i]; + if (nid == 0) + break; + if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* extra output is not muted */ + } + } /*polarity defines *not* muted state level*/ if (!spec->vref_mute_led_nid) { if (muted) spec->gpio_data &= ~spec->gpio_led; /* orange */ else spec->gpio_data |= spec->gpio_led; /* white */ + + if (!spec->gpio_led_polarity) { + /* LED state is inverted on these systems */ + spec->gpio_data ^= spec->gpio_led; + } stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } else { - spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; + notmtd_lvl = spec->gpio_led_polarity ? + AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; + muted_lvl = spec->gpio_led_polarity ? + AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50; + spec->vref_led = muted ? muted_lvl : notmtd_lvl; stac_vrefout_set(codec, spec->vref_mute_led_nid, spec->vref_led); } + return 0; } +/* + * use power check for controlling mute led of HP notebooks + */ +static int stac92xx_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + stac92xx_update_led_status(codec); + + return 0; +} +#endif /* CONFIG_SND_HDA_POWER_SAVE */ +#endif /* CONFIG_PM */ + static const struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, .build_pcms = stac92xx_build_pcms, @@ -5557,12 +5580,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) STAC_92HD83XXX_MODELS, stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); - /* check codec subsystem id if not found */ - if (spec->board_config < 0) - spec->board_config = - snd_hda_check_board_codec_sid_config(codec, - STAC_92HD83XXX_MODELS, stac92hd83xxx_models, - stac92hd83xxx_codec_id_cfg_tbl); again: if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", @@ -5573,17 +5590,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; - switch (spec->board_config) { - case STAC_HP_ZEPHYR: - spec->init = stac92hd83xxx_hp_zephyr_init; - break; - } - if (find_mute_led_cfg(codec, -1/*no default cfg*/)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); +#ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; @@ -5593,10 +5605,11 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->patch_ops.set_power_state = stac92xx_set_power_state; } -#ifdef CONFIG_PM codec->patch_ops.pre_resume = stac92xx_pre_resume; -#endif + codec->patch_ops.check_power_status = + stac92xx_check_power_status; } +#endif err = stac92xx_parse_auto_config(codec); if (!err) { @@ -5893,6 +5906,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->gpio_led, spec->gpio_led_polarity); +#ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; @@ -5902,10 +5916,11 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->patch_ops.set_power_state = stac92xx_set_power_state; } -#ifdef CONFIG_PM codec->patch_ops.pre_resume = stac92xx_pre_resume; -#endif + codec->patch_ops.check_power_status = + stac92xx_check_power_status; } +#endif spec->multiout.dac_nids = spec->dac_nids; diff --git a/trunk/sound/pci/hda/patch_via.c b/trunk/sound/pci/hda/patch_via.c index 06214fdc9486..dff9a00ee8fb 100644 --- a/trunk/sound/pci/hda/patch_via.c +++ b/trunk/sound/pci/hda/patch_via.c @@ -550,10 +550,7 @@ static void via_auto_init_output(struct hda_codec *codec, pin = path->path[path->depth - 1]; init_output_pin(codec, pin, pin_type); - if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) - caps = query_amp_caps(codec, pin, HDA_OUTPUT); - else - caps = 0; + caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_MUTE) { unsigned int val; val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; @@ -648,10 +645,6 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init ADCs */ for (i = 0; i < spec->num_adc_nids; i++) { - hda_nid_t nid = spec->adc_nids[i]; - if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP) || - !(query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE)) - continue; snd_hda_codec_write(codec, spec->adc_nids[i], 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); @@ -1452,9 +1445,25 @@ static const struct hda_pcm_stream via_pcm_digital_capture = { /* * slave controls for virtual master */ -static const char * const via_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", +static const char * const via_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "Headphone Playback Volume", + "Speaker Playback Volume", + NULL, +}; + +static const char * const via_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "Headphone Playback Switch", + "Speaker Playback Switch", NULL, }; @@ -1499,15 +1508,13 @@ static int via_build_controls(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, via_slave_pfxs, - "Playback Volume"); + vmaster_tlv, via_slave_vols); if (err < 0) return err; } if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, via_slave_pfxs, - "Playback Switch"); + NULL, via_slave_sws); if (err < 0) return err; } @@ -1515,8 +1522,6 @@ static int via_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - if (!spec->mux_nids[i]) - continue; err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]); if (err < 0) return err; @@ -2483,8 +2488,6 @@ static int create_mic_boost_ctls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; - const char *prev_label = NULL; - int type_idx = 0; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -2499,13 +2502,8 @@ static int create_mic_boost_ctls(struct hda_codec *codec) if (caps == -1 || !(caps & AC_AMPCAP_NUM_STEPS)) continue; label = hda_get_autocfg_input_label(codec, cfg, i); - if (prev_label && !strcmp(label, prev_label)) - type_idx++; - else - type_idx = 0; - prev_label = label; snprintf(name, sizeof(name), "%s Boost Volume", label); - err = __via_add_control(spec, VIA_CTL_WIDGET_VOL, name, type_idx, + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT)); if (err < 0) return err; diff --git a/trunk/sound/pci/ice1712/ice1724.c b/trunk/sound/pci/ice1712/ice1724.c index 812d10e43ae0..92362973764d 100644 --- a/trunk/sound/pci/ice1712/ice1724.c +++ b/trunk/sound/pci/ice1712/ice1724.c @@ -1013,25 +1013,6 @@ static int set_rate_constraints(struct snd_ice1712 *ice, ice->hw_rates); } -/* if the card has the internal rate locked (is_pro_locked), limit runtime - hw rates to the current internal rate only. -*/ -static void constrain_rate_if_locked(struct snd_pcm_substream *substream) -{ - struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int rate; - if (is_pro_rate_locked(ice)) { - rate = ice->get_rate(ice); - if (rate >= runtime->hw.rate_min - && rate <= runtime->hw.rate_max) { - runtime->hw.rate_min = rate; - runtime->hw.rate_max = rate; - } - } -} - - /* multi-channel playback needs alignment 8x32bit regardless of the channels * actually used */ @@ -1065,7 +1046,6 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); - constrain_rate_if_locked(substream); if (ice->pro_open) ice->pro_open(ice, substream); return 0; @@ -1086,7 +1066,6 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); - constrain_rate_if_locked(substream); if (ice->pro_open) ice->pro_open(ice, substream); return 0; @@ -1236,7 +1215,6 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); - constrain_rate_if_locked(substream); if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); return 0; @@ -1273,7 +1251,6 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); - constrain_rate_if_locked(substream); if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); return 0; diff --git a/trunk/sound/pci/rme9652/hdspm.c b/trunk/sound/pci/rme9652/hdspm.c index bc030a2088da..cc9f6c83d661 100644 --- a/trunk/sound/pci/rme9652/hdspm.c +++ b/trunk/sound/pci/rme9652/hdspm.c @@ -6333,7 +6333,6 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; - hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl; hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; diff --git a/trunk/sound/pci/ymfpci/ymfpci_main.c b/trunk/sound/pci/ymfpci/ymfpci_main.c index a8159b81e9c4..12a9a2b03387 100644 --- a/trunk/sound/pci/ymfpci/ymfpci_main.c +++ b/trunk/sound/pci/ymfpci/ymfpci_main.c @@ -2317,10 +2317,6 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++) chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]); chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE); - pci_read_config_word(chip->pci, PCIR_DSXG_LEGACY, - &chip->saved_dsxg_legacy); - pci_read_config_word(chip->pci, PCIR_DSXG_ELEGACY, - &chip->saved_dsxg_elegacy); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0); snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0); snd_ymfpci_disable_dsp(chip); @@ -2355,11 +2351,6 @@ int snd_ymfpci_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97); - pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY, - chip->saved_dsxg_legacy); - pci_write_config_word(chip->pci, PCIR_DSXG_ELEGACY, - chip->saved_dsxg_elegacy); - /* start hw again */ if (chip->start_count > 0) { spin_lock_irq(&chip->reg_lock); diff --git a/trunk/sound/soc/codecs/ak4642.c b/trunk/sound/soc/codecs/ak4642.c index f8e10ced244a..16bd1e7d2384 100644 --- a/trunk/sound/soc/codecs/ak4642.c +++ b/trunk/sound/soc/codecs/ak4642.c @@ -146,10 +146,13 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), + + SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; -static const struct snd_kcontrol_new ak4642_headphone_control = - SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); +static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { + SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), +}; static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), @@ -162,12 +165,13 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("LINEOUT"), - SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), - SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, - &ak4642_headphone_control), + SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, + &ak4642_hpout_mixer_controls[0], + ARRAY_SIZE(ak4642_hpout_mixer_controls)), - SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, + &ak4642_hpout_mixer_controls[0], + ARRAY_SIZE(ak4642_hpout_mixer_controls)), SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], @@ -180,17 +184,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ - {"HPOUTL", NULL, "HPL Out"}, - {"HPOUTR", NULL, "HPR Out"}, + {"HPOUTL", NULL, "HPOUTL Mixer"}, + {"HPOUTR", NULL, "HPOUTR Mixer"}, {"LINEOUT", NULL, "LINEOUT Mixer"}, - {"HPL Out", NULL, "Headphone Enable"}, - {"HPR Out", NULL, "Headphone Enable"}, - - {"Headphone Enable", "Switch", "DACH"}, - - {"DACH", NULL, "DAC"}, - + {"HPOUTL Mixer", "DACH", "DAC"}, + {"HPOUTR Mixer", "DACH", "DAC"}, {"LINEOUT Mixer", "DACL", "DAC"}, }; diff --git a/trunk/sound/soc/codecs/wm8962.c b/trunk/sound/soc/codecs/wm8962.c index 15d467ff91b4..5bcb350bacc1 100644 --- a/trunk/sound/soc/codecs/wm8962.c +++ b/trunk/sound/soc/codecs/wm8962.c @@ -1988,7 +1988,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *st_text[] = { "None", "Left", "Right" }; +static const char *st_text[] = { "None", "Right", "Left" }; static const struct soc_enum str_enum = SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); diff --git a/trunk/sound/soc/codecs/wm8994.c b/trunk/sound/soc/codecs/wm8994.c index 9685dff44dd8..10d27890eed5 100644 --- a/trunk/sound/soc/codecs/wm8994.c +++ b/trunk/sound/soc/codecs/wm8994.c @@ -3652,7 +3652,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case 2: case 3: wm8994->hubs.dcs_codes_l = -9; - wm8994->hubs.dcs_codes_r = -5; + wm8994->hubs.dcs_codes_r = -7; break; default: break; diff --git a/trunk/sound/soc/imx/imx-ssi.c b/trunk/sound/soc/imx/imx-ssi.c index 4f81ed456325..9203cdd0a154 100644 --- a/trunk/sound/soc/imx/imx-ssi.c +++ b/trunk/sound/soc/imx/imx-ssi.c @@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; + strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; break; } diff --git a/trunk/sound/soc/omap/Kconfig b/trunk/sound/soc/omap/Kconfig index e00dd0b1139c..47b23fea20c2 100644 --- a/trunk/sound/soc/omap/Kconfig +++ b/trunk/sound/soc/omap/Kconfig @@ -7,6 +7,7 @@ config SND_OMAP_SOC_DMIC config SND_OMAP_SOC_MCBSP tristate + select OMAP_MCBSP config SND_OMAP_SOC_MCPDM tristate @@ -26,6 +27,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 diff --git a/trunk/sound/soc/omap/Makefile b/trunk/sound/soc/omap/Makefile index 1d656bce01d4..123ac18303e5 100644 --- a/trunk/sound/soc/omap/Makefile +++ b/trunk/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-objs := omap-hdmi.o diff --git a/trunk/sound/soc/omap/am3517evm.c b/trunk/sound/soc/omap/am3517evm.c index 009533ab8d18..add4866d7e67 100644 --- a/trunk/sound/soc/omap/am3517evm.c +++ b/trunk/sound/soc/omap/am3517evm.c @@ -95,7 +95,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_dai_link am3517evm_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", - .cpu_dai_name = "omap-mcbsp.1", + .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec.2-001a", diff --git a/trunk/sound/soc/omap/ams-delta.c b/trunk/sound/soc/omap/ams-delta.c index 49fe63ce51f7..78563bbbbf01 100644 --- a/trunk/sound/soc/omap/ams-delta.c +++ b/trunk/sound/soc/omap/ams-delta.c @@ -584,7 +584,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link ams_delta_dai_link = { .name = "CX20442", .stream_name = "CX20442", - .cpu_dai_name = "omap-mcbsp.1", + .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "cx20442-voice", .init = ams_delta_cx20442_init, .platform_name = "omap-pcm-audio", diff --git a/trunk/sound/soc/omap/igep0020.c b/trunk/sound/soc/omap/igep0020.c index e8357819175b..ccae58a1339c 100644 --- a/trunk/sound/soc/omap/igep0020.c +++ b/trunk/sound/soc/omap/igep0020.c @@ -60,7 +60,7 @@ static struct snd_soc_ops igep2_ops = { static struct snd_soc_dai_link igep2_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/omap/mcbsp.h b/trunk/sound/soc/omap/mcbsp.h deleted file mode 100644 index a944fcc9073c..000000000000 --- a/trunk/sound/soc/omap/mcbsp.h +++ /dev/null @@ -1,346 +0,0 @@ -/* - * sound/soc/omap/mcbsp.h - * - * OMAP Multi-Channel Buffered Serial Port - * - * Contact: Jarkko Nikula - * Peter Ujfalusi - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ -#ifndef __ASOC_MCBSP_H -#define __ASOC_MCBSP_H - -#include "omap-pcm.h" - -/* McBSP register numbers. Register address offset = num * reg_step */ -enum { - /* Common registers */ - OMAP_MCBSP_REG_SPCR2 = 4, - OMAP_MCBSP_REG_SPCR1, - OMAP_MCBSP_REG_RCR2, - OMAP_MCBSP_REG_RCR1, - OMAP_MCBSP_REG_XCR2, - OMAP_MCBSP_REG_XCR1, - OMAP_MCBSP_REG_SRGR2, - OMAP_MCBSP_REG_SRGR1, - OMAP_MCBSP_REG_MCR2, - OMAP_MCBSP_REG_MCR1, - OMAP_MCBSP_REG_RCERA, - OMAP_MCBSP_REG_RCERB, - OMAP_MCBSP_REG_XCERA, - OMAP_MCBSP_REG_XCERB, - OMAP_MCBSP_REG_PCR0, - OMAP_MCBSP_REG_RCERC, - OMAP_MCBSP_REG_RCERD, - OMAP_MCBSP_REG_XCERC, - OMAP_MCBSP_REG_XCERD, - OMAP_MCBSP_REG_RCERE, - OMAP_MCBSP_REG_RCERF, - OMAP_MCBSP_REG_XCERE, - OMAP_MCBSP_REG_XCERF, - OMAP_MCBSP_REG_RCERG, - OMAP_MCBSP_REG_RCERH, - OMAP_MCBSP_REG_XCERG, - OMAP_MCBSP_REG_XCERH, - - /* OMAP1-OMAP2420 registers */ - OMAP_MCBSP_REG_DRR2 = 0, - OMAP_MCBSP_REG_DRR1, - OMAP_MCBSP_REG_DXR2, - OMAP_MCBSP_REG_DXR1, - - /* OMAP2430 and onwards */ - OMAP_MCBSP_REG_DRR = 0, - OMAP_MCBSP_REG_DXR = 2, - OMAP_MCBSP_REG_SYSCON = 35, - OMAP_MCBSP_REG_THRSH2, - OMAP_MCBSP_REG_THRSH1, - OMAP_MCBSP_REG_IRQST = 40, - OMAP_MCBSP_REG_IRQEN, - OMAP_MCBSP_REG_WAKEUPEN, - OMAP_MCBSP_REG_XCCR, - OMAP_MCBSP_REG_RCCR, - OMAP_MCBSP_REG_XBUFFSTAT, - OMAP_MCBSP_REG_RBUFFSTAT, - OMAP_MCBSP_REG_SSELCR, -}; - -/* OMAP3 sidetone control registers */ -#define OMAP_ST_REG_REV 0x00 -#define OMAP_ST_REG_SYSCONFIG 0x10 -#define OMAP_ST_REG_IRQSTATUS 0x18 -#define OMAP_ST_REG_IRQENABLE 0x1C -#define OMAP_ST_REG_SGAINCR 0x24 -#define OMAP_ST_REG_SFIRCR 0x28 -#define OMAP_ST_REG_SSELCR 0x2C - -/************************** McBSP SPCR1 bit definitions ***********************/ -#define RRST BIT(0) -#define RRDY BIT(1) -#define RFULL BIT(2) -#define RSYNC_ERR BIT(3) -#define RINTM(value) (((value) & 0x3) << 4) /* bits 4:5 */ -#define ABIS BIT(6) -#define DXENA BIT(7) -#define CLKSTP(value) (((value) & 0x3) << 11) /* bits 11:12 */ -#define RJUST(value) (((value) & 0x3) << 13) /* bits 13:14 */ -#define ALB BIT(15) -#define DLB BIT(15) - -/************************** McBSP SPCR2 bit definitions ***********************/ -#define XRST BIT(0) -#define XRDY BIT(1) -#define XEMPTY BIT(2) -#define XSYNC_ERR BIT(3) -#define XINTM(value) (((value) & 0x3) << 4) /* bits 4:5 */ -#define GRST BIT(6) -#define FRST BIT(7) -#define SOFT BIT(8) -#define FREE BIT(9) - -/************************** McBSP PCR bit definitions *************************/ -#define CLKRP BIT(0) -#define CLKXP BIT(1) -#define FSRP BIT(2) -#define FSXP BIT(3) -#define DR_STAT BIT(4) -#define DX_STAT BIT(5) -#define CLKS_STAT BIT(6) -#define SCLKME BIT(7) -#define CLKRM BIT(8) -#define CLKXM BIT(9) -#define FSRM BIT(10) -#define FSXM BIT(11) -#define RIOEN BIT(12) -#define XIOEN BIT(13) -#define IDLE_EN BIT(14) - -/************************** McBSP RCR1 bit definitions ************************/ -#define RWDLEN1(value) (((value) & 0x7) << 5) /* Bits 5:7 */ -#define RFRLEN1(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ - -/************************** McBSP XCR1 bit definitions ************************/ -#define XWDLEN1(value) (((value) & 0x7) << 5) /* Bits 5:7 */ -#define XFRLEN1(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ - -/*************************** McBSP RCR2 bit definitions ***********************/ -#define RDATDLY(value) ((value) & 0x3) /* Bits 0:1 */ -#define RFIG BIT(2) -#define RCOMPAND(value) (((value) & 0x3) << 3) /* Bits 3:4 */ -#define RWDLEN2(value) (((value) & 0x7) << 5) /* Bits 5:7 */ -#define RFRLEN2(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ -#define RPHASE BIT(15) - -/*************************** McBSP XCR2 bit definitions ***********************/ -#define XDATDLY(value) ((value) & 0x3) /* Bits 0:1 */ -#define XFIG BIT(2) -#define XCOMPAND(value) (((value) & 0x3) << 3) /* Bits 3:4 */ -#define XWDLEN2(value) (((value) & 0x7) << 5) /* Bits 5:7 */ -#define XFRLEN2(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ -#define XPHASE BIT(15) - -/************************* McBSP SRGR1 bit definitions ************************/ -#define CLKGDV(value) ((value) & 0x7f) /* Bits 0:7 */ -#define FWID(value) (((value) & 0xff) << 8) /* Bits 8:15 */ - -/************************* McBSP SRGR2 bit definitions ************************/ -#define FPER(value) ((value) & 0x0fff) /* Bits 0:11 */ -#define FSGM BIT(12) -#define CLKSM BIT(13) -#define CLKSP BIT(14) -#define GSYNC BIT(15) - -/************************* McBSP MCR1 bit definitions *************************/ -#define RMCM BIT(0) -#define RCBLK(value) (((value) & 0x7) << 2) /* Bits 2:4 */ -#define RPABLK(value) (((value) & 0x3) << 5) /* Bits 5:6 */ -#define RPBBLK(value) (((value) & 0x3) << 7) /* Bits 7:8 */ - -/************************* McBSP MCR2 bit definitions *************************/ -#define XMCM(value) ((value) & 0x3) /* Bits 0:1 */ -#define XCBLK(value) (((value) & 0x7) << 2) /* Bits 2:4 */ -#define XPABLK(value) (((value) & 0x3) << 5) /* Bits 5:6 */ -#define XPBBLK(value) (((value) & 0x3) << 7) /* Bits 7:8 */ - -/*********************** McBSP XCCR bit definitions *************************/ -#define XDISABLE BIT(0) -#define XDMAEN BIT(3) -#define DILB BIT(5) -#define XFULL_CYCLE BIT(11) -#define DXENDLY(value) (((value) & 0x3) << 12) /* Bits 12:13 */ -#define PPCONNECT BIT(14) -#define EXTCLKGATE BIT(15) - -/********************** McBSP RCCR bit definitions *************************/ -#define RDISABLE BIT(0) -#define RDMAEN BIT(3) -#define RFULL_CYCLE BIT(11) - -/********************** McBSP SYSCONFIG bit definitions ********************/ -#define SOFTRST BIT(1) -#define ENAWAKEUP BIT(2) -#define SIDLEMODE(value) (((value) & 0x3) << 3) -#define CLOCKACTIVITY(value) (((value) & 0x3) << 8) - -/********************** McBSP SSELCR bit definitions ***********************/ -#define SIDETONEEN BIT(10) - -/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ -#define ST_AUTOIDLE BIT(0) - -/********************** McBSP Sidetone SGAINCR bit definitions *************/ -#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ -#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ - -/********************** McBSP Sidetone SFIRCR bit definitions **************/ -#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ - -/********************** McBSP Sidetone SSELCR bit definitions **************/ -#define ST_SIDETONEEN BIT(0) -#define ST_COEFFWREN BIT(1) -#define ST_COEFFWRDONE BIT(2) - -/********************** McBSP DMA operating modes **************************/ -#define MCBSP_DMA_MODE_ELEMENT 0 -#define MCBSP_DMA_MODE_THRESHOLD 1 -#define MCBSP_DMA_MODE_FRAME 2 - -/********************** McBSP WAKEUPEN bit definitions *********************/ -#define RSYNCERREN BIT(0) -#define RFSREN BIT(1) -#define REOFEN BIT(2) -#define RRDYEN BIT(3) -#define XSYNCERREN BIT(7) -#define XFSXEN BIT(8) -#define XEOFEN BIT(9) -#define XRDYEN BIT(10) -#define XEMPTYEOFEN BIT(14) - -/* Clock signal muxing options */ -#define CLKR_SRC_CLKR 0 /* CLKR signal is from the CLKR pin */ -#define CLKR_SRC_CLKX 1 /* CLKR signal is from the CLKX pin */ -#define FSR_SRC_FSR 2 /* FSR signal is from the FSR pin */ -#define FSR_SRC_FSX 3 /* FSR signal is from the FSX pin */ - -/* McBSP functional clock sources */ -#define MCBSP_CLKS_PRCM_SRC 0 -#define MCBSP_CLKS_PAD_SRC 1 - -/* we don't do multichannel for now */ -struct omap_mcbsp_reg_cfg { - u16 spcr2; - u16 spcr1; - u16 rcr2; - u16 rcr1; - u16 xcr2; - u16 xcr1; - u16 srgr2; - u16 srgr1; - u16 mcr2; - u16 mcr1; - u16 pcr0; - u16 rcerc; - u16 rcerd; - u16 xcerc; - u16 xcerd; - u16 rcere; - u16 rcerf; - u16 xcere; - u16 xcerf; - u16 rcerg; - u16 rcerh; - u16 xcerg; - u16 xcerh; - u16 xccr; - u16 rccr; -}; - -struct omap_mcbsp_st_data { - void __iomem *io_base_st; - bool running; - bool enabled; - s16 taps[128]; /* Sidetone filter coefficients */ - int nr_taps; /* Number of filter coefficients in use */ - s16 ch0gain; - s16 ch1gain; -}; - -struct omap_mcbsp { - struct device *dev; - struct clk *fclk; - spinlock_t lock; - unsigned long phys_base; - unsigned long phys_dma_base; - void __iomem *io_base; - u8 id; - /* - * Flags indicating is the bus already activated and configured by - * another substream - */ - int active; - int configured; - u8 free; - - int rx_irq; - int tx_irq; - - /* Protect the field .free, while checking if the mcbsp is in use */ - struct omap_mcbsp_platform_data *pdata; - struct omap_mcbsp_st_data *st_data; - struct omap_mcbsp_reg_cfg cfg_regs; - struct omap_pcm_dma_data dma_data[2]; - int dma_op_mode; - u16 max_tx_thres; - u16 max_rx_thres; - void *reg_cache; - int reg_cache_size; - - unsigned int fmt; - unsigned int in_freq; - int clk_div; - int wlen; -}; - -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config); -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp); -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); -int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); -int omap_mcbsp_request(struct omap_mcbsp *mcbsp); -void omap_mcbsp_free(struct omap_mcbsp *mcbsp); -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); - -/* McBSP functional clock source changing function */ -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); - -/* McBSP signal muxing API */ -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux); - -/* Sidetone specific API */ -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); -int omap_st_enable(struct omap_mcbsp *mcbsp); -int omap_st_disable(struct omap_mcbsp *mcbsp); -int omap_st_is_enabled(struct omap_mcbsp *mcbsp); - -int __devinit omap_mcbsp_init(struct platform_device *pdev); -void __devexit omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp); - -#endif /* __ASOC_MCBSP_H */ diff --git a/trunk/sound/soc/omap/n810.c b/trunk/sound/soc/omap/n810.c index abac4b690750..c292bf0fd19c 100644 --- a/trunk/sound/soc/omap/n810.c +++ b/trunk/sound/soc/omap/n810.c @@ -275,7 +275,7 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link n810_dai = { .name = "TLV320AIC33", .stream_name = "AIC33", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", .codec_dai_name = "tlv320aic3x-hifi", diff --git a/trunk/sound/soc/omap/omap-mcbsp.c b/trunk/sound/soc/omap/omap-mcbsp.c index 6912ac7cb625..1287b870f221 100644 --- a/trunk/sound/soc/omap/omap-mcbsp.c +++ b/trunk/sound/soc/omap/omap-mcbsp.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include #include @@ -34,7 +33,6 @@ #include #include -#include "mcbsp.h" #include "omap-mcbsp.h" #include "omap-pcm.h" @@ -48,31 +46,42 @@ .private_value = (unsigned long) &(struct soc_mixer_control) \ {.min = xmin, .max = xmax} } -enum { - OMAP_MCBSP_WORD_8 = 0, - OMAP_MCBSP_WORD_12, - OMAP_MCBSP_WORD_16, - OMAP_MCBSP_WORD_20, - OMAP_MCBSP_WORD_24, - OMAP_MCBSP_WORD_32, +struct omap_mcbsp_data { + unsigned int bus_id; + struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; + /* + * Flags indicating is the bus already activated and configured by + * another substream + */ + int active; + int configured; + unsigned int in_freq; + int clk_div; + int wlen; }; +static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; + /* * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ +static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; + static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); struct omap_pcm_dma_data *dma_data; + int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id); int words; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) + if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or @@ -82,15 +91,15 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) words = dma_data->packet_size; else words = snd_pcm_lib_period_bytes(substream) / - (mcbsp->wlen / 8); + (mcbsp_data->wlen / 8); else words = 1; /* Configure McBSP internal buffer usage */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - omap_mcbsp_set_tx_threshold(mcbsp, words); + omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words); else - omap_mcbsp_set_rx_threshold(mcbsp, words); + omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words); } static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, @@ -100,12 +109,12 @@ static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct omap_mcbsp *mcbsp = rule->private; + struct omap_mcbsp_data *mcbsp_data = rule->private; struct snd_interval frames; int size; snd_interval_any(&frames); - size = mcbsp->pdata->buffer_size; + size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id); frames.min = size / channels->min; frames.integer = 1; @@ -115,11 +124,12 @@ static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + int bus_id = mcbsp_data->bus_id; int err = 0; if (!cpu_dai->active) - err = omap_mcbsp_request(mcbsp); + err = omap_mcbsp_request(bus_id); /* * OMAP3 McBSP FIFO is word structured. @@ -136,16 +146,16 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (mcbsp->pdata->buffer_size) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns */ snd_pcm_hw_rule_add(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + SNDRV_PCM_HW_PARAM_CHANNELS, omap_mcbsp_hwrule_min_buffersize, - mcbsp, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); + mcbsp_data, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -158,33 +168,33 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); if (!cpu_dai->active) { - omap_mcbsp_free(mcbsp); - mcbsp->configured = 0; + omap_mcbsp_free(mcbsp_data->bus_id); + mcbsp_data->configured = 0; } } static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - mcbsp->active++; - omap_mcbsp_start(mcbsp, play, !play); + mcbsp_data->active++; + omap_mcbsp_start(mcbsp_data->bus_id, play, !play); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - omap_mcbsp_stop(mcbsp, play, !play); - mcbsp->active--; + omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + mcbsp_data->active--; break; default: err = -EINVAL; @@ -199,14 +209,14 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); u16 fifo_use; snd_pcm_sframes_t delay; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_use = omap_mcbsp_get_tx_delay(mcbsp); + fifo_use = omap_mcbsp_get_tx_delay(mcbsp_data->bus_id); else - fifo_use = omap_mcbsp_get_rx_delay(mcbsp); + fifo_use = omap_mcbsp_get_rx_delay(mcbsp_data->bus_id); /* * Divide the used locations with the channel count to get the @@ -222,14 +232,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; struct omap_pcm_dma_data *dma_data; + int dma, bus_id = mcbsp_data->bus_id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; int pkt_size = 0; + unsigned long port; unsigned int format, div, framesize, master; - dma_data = &mcbsp->dma_data[substream->stream]; + dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream]; + + dma = omap_mcbsp_dma_ch_params(bus_id, substream->stream); + port = omap_mcbsp_dma_reg_params(bus_id, substream->stream); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -243,17 +258,20 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { + if (omap_mcbsp_get_dma_op_mode(bus_id) == + MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; period_words = params_period_bytes(params) / (wlen / 8); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - max_thrsh = mcbsp->max_tx_thres; + max_thrsh = omap_mcbsp_get_max_tx_threshold( + mcbsp_data->bus_id); else - max_thrsh = mcbsp->max_rx_thres; + max_thrsh = omap_mcbsp_get_max_rx_threshold( + mcbsp_data->bus_id); /* * If the period contains less or equal number of words, * we are using the original threshold mode setup: @@ -286,12 +304,15 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } } + dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback"; + dma_data->dma_req = dma; + dma_data->port_addr = port; dma_data->sync_mode = sync_mode; dma_data->packet_size = pkt_size; snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - if (mcbsp->configured) { + if (mcbsp_data->configured) { /* McBSP already configured by another stream */ return 0; } @@ -300,7 +321,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->xcr2 &= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7)); regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7)); regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7)); - format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || format == SND_SOC_DAIFMT_LEFT_J)) { @@ -338,10 +359,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* In McBSP master modes, FRAME (i.e. sample rate) is generated * by _counting_ BCLKs. Calculate frame size in BCLKs */ - master = mcbsp->fmt & SND_SOC_DAIFMT_MASTER_MASK; + master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK; if (master == SND_SOC_DAIFMT_CBS_CFS) { - div = mcbsp->clk_div ? mcbsp->clk_div : 1; - framesize = (mcbsp->in_freq / div) / params_rate(params); + div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; + framesize = (mcbsp_data->in_freq / div) / params_rate(params); if (framesize < wlen * channels) { printk(KERN_ERR "%s: not enough bandwidth for desired rate and " @@ -367,9 +388,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; } - omap_mcbsp_config(mcbsp, &mcbsp->cfg_regs); - mcbsp->wlen = wlen; - mcbsp->configured = 1; + omap_mcbsp_config(bus_id, &mcbsp_data->regs); + mcbsp_data->wlen = wlen; + mcbsp_data->configured = 1; return 0; } @@ -381,14 +402,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; bool inv_fs = false; - if (mcbsp->configured) + if (mcbsp_data->configured) return 0; - mcbsp->fmt = fmt; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -483,13 +504,13 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; - mcbsp->clk_div = div; + mcbsp_data->clk_div = div; regs->srgr1 &= ~CLKGDV(0xff); regs->srgr1 |= CLKGDV(div - 1); @@ -500,32 +521,28 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; + struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; - if (mcbsp->active) { - if (freq == mcbsp->in_freq) + if (mcbsp_data->active) { + if (freq == mcbsp_data->in_freq) return 0; else return -EBUSY; } - if (clk_id == OMAP_MCBSP_SYSCLK_CLK || - clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK || - clk_id == OMAP_MCBSP_SYSCLK_CLKS_EXT || - clk_id == OMAP_MCBSP_SYSCLK_CLKX_EXT || - clk_id == OMAP_MCBSP_SYSCLK_CLKR_EXT) { - mcbsp->in_freq = freq; - regs->srgr2 &= ~CLKSM; - regs->pcr0 &= ~SCLKME; - } else if (cpu_class_is_omap1()) { - /* - * McBSP CLKR/FSR signal muxing functions are only available on - * OMAP2 or newer versions - */ - return -EINVAL; - } + /* The McBSP signal muxing functions are only available on McBSP1 */ + if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || + clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || + clk_id == OMAP_MCBSP_FSR_SRC_FSR || + clk_id == OMAP_MCBSP_FSR_SRC_FSX) + if (cpu_class_is_omap1() || mcbsp_data->bus_id != 0) + return -EINVAL; + + mcbsp_data->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: @@ -536,7 +553,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, err = -EINVAL; break; } - err = omap2_mcbsp_set_clks_src(mcbsp, + err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, MCBSP_CLKS_PRCM_SRC); break; case OMAP_MCBSP_SYSCLK_CLKS_EXT: @@ -544,7 +561,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, err = 0; break; } - err = omap2_mcbsp_set_clks_src(mcbsp, + err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, MCBSP_CLKS_PAD_SRC); break; @@ -556,16 +573,24 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_CLKR_SRC_CLKR: - err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKR); + if (cpu_class_is_omap1()) + break; + omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR); break; case OMAP_MCBSP_CLKR_SRC_CLKX: - err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKX); + if (cpu_class_is_omap1()) + break; + omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX); break; case OMAP_MCBSP_FSR_SRC_FSR: - err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSR); + if (cpu_class_is_omap1()) + break; + omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR); break; case OMAP_MCBSP_FSR_SRC_FSX: - err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSX); + if (cpu_class_is_omap1()) + break; + omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX); break; default: err = -ENODEV; @@ -585,27 +610,15 @@ static const struct snd_soc_dai_ops mcbsp_dai_ops = { .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, }; -static int omap_mcbsp_probe(struct snd_soc_dai *dai) +static int mcbsp_dai_probe(struct snd_soc_dai *dai) { - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai); - - pm_runtime_enable(mcbsp->dev); - - return 0; -} - -static int omap_mcbsp_remove(struct snd_soc_dai *dai) -{ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai); - - pm_runtime_disable(mcbsp->dev); - + mcbsp_data[dai->id].bus_id = dai->id; + snd_soc_dai_set_drvdata(dai, &mcbsp_data[dai->id].bus_id); return 0; } static struct snd_soc_dai_driver omap_mcbsp_dai = { - .probe = omap_mcbsp_probe, - .remove = omap_mcbsp_remove, + .probe = mcbsp_dai_probe, .playback = { .channels_min = 1, .channels_max = 16, @@ -636,13 +649,11 @@ static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, return 0; } -#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(channel) \ +#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel) \ static int \ -omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ +omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ struct snd_ctl_elem_value *uc) \ { \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ struct soc_mixer_control *mc = \ (struct soc_mixer_control *)kc->private_value; \ int max = mc->max; \ @@ -653,44 +664,46 @@ omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ return -EINVAL; \ \ /* OMAP McBSP implementation uses index values 0..4 */ \ - return omap_st_set_chgain(mcbsp, channel, val); \ + return omap_st_set_chgain((id)-1, channel, val); \ } -#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(channel) \ +#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel) \ static int \ -omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ +omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ struct snd_ctl_elem_value *uc) \ { \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ s16 chgain; \ \ - if (omap_st_get_chgain(mcbsp, channel, &chgain)) \ + if (omap_st_get_chgain((id)-1, channel, &chgain)) \ return -EAGAIN; \ \ uc->value.integer.value[0] = chgain; \ return 0; \ } -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(1) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(1) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1) static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; u8 value = ucontrol->value.integer.value[0]; - if (value == omap_st_is_enabled(mcbsp)) + if (value == omap_st_is_enabled(mc->reg)) return 0; if (value) - omap_st_enable(mcbsp); + omap_st_enable(mc->reg); else - omap_st_disable(mcbsp); + omap_st_disable(mc->reg); return 1; } @@ -698,10 +711,10 @@ static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; - ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp); + ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg); return 0; } @@ -710,12 +723,12 @@ static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = { omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume", -32768, 32767, - omap_mcbsp_get_st_ch0_volume, - omap_mcbsp_set_st_ch0_volume), + omap_mcbsp2_get_st_ch0_volume, + omap_mcbsp2_set_st_ch0_volume), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume", -32768, 32767, - omap_mcbsp_get_st_ch1_volume, - omap_mcbsp_set_st_ch1_volume), + omap_mcbsp2_get_st_ch1_volume, + omap_mcbsp2_set_st_ch1_volume), }; static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { @@ -723,30 +736,25 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume", -32768, 32767, - omap_mcbsp_get_st_ch0_volume, - omap_mcbsp_set_st_ch0_volume), + omap_mcbsp3_get_st_ch0_volume, + omap_mcbsp3_set_st_ch0_volume), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume", -32768, 32767, - omap_mcbsp_get_st_ch1_volume, - omap_mcbsp_set_st_ch1_volume), + omap_mcbsp3_get_st_ch1_volume, + omap_mcbsp3_set_st_ch1_volume), }; -int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) +int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - if (!mcbsp->st_data) + if (!cpu_is_omap34xx()) return -ENODEV; - switch (cpu_dai->id) { - case 2: /* McBSP 2 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp2_st_controls, + switch (dai->id) { + case 1: /* McBSP 2 */ + return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls, ARRAY_SIZE(omap_mcbsp2_st_controls)); - case 3: /* McBSP 3 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp3_st_controls, + case 2: /* McBSP 3 */ + return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls, ARRAY_SIZE(omap_mcbsp3_st_controls)); default: break; @@ -758,51 +766,18 @@ EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); static __devinit int asoc_mcbsp_probe(struct platform_device *pdev) { - struct omap_mcbsp_platform_data *pdata = dev_get_platdata(&pdev->dev); - struct omap_mcbsp *mcbsp; - int ret; - - if (!pdata) { - dev_err(&pdev->dev, "missing platform data.\n"); - return -EINVAL; - } - mcbsp = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcbsp), GFP_KERNEL); - if (!mcbsp) - return -ENOMEM; - - mcbsp->id = pdev->id; - mcbsp->pdata = pdata; - mcbsp->dev = &pdev->dev; - platform_set_drvdata(pdev, mcbsp); - - ret = omap_mcbsp_init(pdev); - if (!ret) - return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai); - - return ret; + return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai); } static int __devexit asoc_mcbsp_remove(struct platform_device *pdev) { - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_dai(&pdev->dev); - - if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id); - - omap_mcbsp_sysfs_remove(mcbsp); - - clk_put(mcbsp->fclk); - - platform_set_drvdata(pdev, NULL); - return 0; } static struct platform_driver asoc_mcbsp_driver = { .driver = { - .name = "omap-mcbsp", + .name = "omap-mcbsp-dai", .owner = THIS_MODULE, }, diff --git a/trunk/sound/soc/omap/omap-mcbsp.h b/trunk/sound/soc/omap/omap-mcbsp.h index f877b16f19c9..476fe2add703 100644 --- a/trunk/sound/soc/omap/omap-mcbsp.h +++ b/trunk/sound/soc/omap/omap-mcbsp.h @@ -59,6 +59,6 @@ enum omap_mcbsp_div { #define NUM_LINKS 5 #endif -int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd); +int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai); #endif diff --git a/trunk/sound/soc/omap/omap-pcm.h b/trunk/sound/soc/omap/omap-pcm.h index b92248cbd47a..f95fe3064172 100644 --- a/trunk/sound/soc/omap/omap-pcm.h +++ b/trunk/sound/soc/omap/omap-pcm.h @@ -25,8 +25,6 @@ #ifndef __OMAP_PCM_H__ #define __OMAP_PCM_H__ -struct snd_pcm_substream; - struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ diff --git a/trunk/sound/soc/omap/omap3beagle.c b/trunk/sound/soc/omap/omap3beagle.c index 2830dfd05661..3357dcc47ed4 100644 --- a/trunk/sound/soc/omap/omap3beagle.c +++ b/trunk/sound/soc/omap/omap3beagle.c @@ -91,7 +91,7 @@ static struct snd_soc_ops omap3beagle_ops = { static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "omap-pcm-audio", .codec_dai_name = "twl4030-hifi", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/omap/omap3evm.c b/trunk/sound/soc/omap/omap3evm.c index 3d468c9179d7..071fcb09b8b2 100644 --- a/trunk/sound/soc/omap/omap3evm.c +++ b/trunk/sound/soc/omap/omap3evm.c @@ -58,7 +58,7 @@ static struct snd_soc_ops omap3evm_ops = { static struct snd_soc_dai_link omap3evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/omap/omap3pandora.c b/trunk/sound/soc/omap/omap3pandora.c index 4c3a0978578a..07794bd10952 100644 --- a/trunk/sound/soc/omap/omap3pandora.c +++ b/trunk/sound/soc/omap/omap3pandora.c @@ -208,7 +208,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { { .name = "PCM1773", .stream_name = "HiFi Out", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", @@ -219,7 +219,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { }, { .name = "TWL4030", .stream_name = "Line/Mic In", - .cpu_dai_name = "omap-mcbsp.4", + .cpu_dai_name = "omap-mcbsp-dai.3", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/omap/osk5912.c b/trunk/sound/soc/omap/osk5912.c index b1a9d64cbc56..d859b597e7ec 100644 --- a/trunk/sound/soc/omap/osk5912.c +++ b/trunk/sound/soc/omap/osk5912.c @@ -96,7 +96,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", - .cpu_dai_name = "omap-mcbsp.1", + .cpu_dai_name = "omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec", diff --git a/trunk/sound/soc/omap/overo.c b/trunk/sound/soc/omap/overo.c index 6ac3e0c3c282..2ee889c50256 100644 --- a/trunk/sound/soc/omap/overo.c +++ b/trunk/sound/soc/omap/overo.c @@ -60,7 +60,7 @@ static struct snd_soc_ops overo_ops = { static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/omap/rx51.c b/trunk/sound/soc/omap/rx51.c index 2712dd232b6d..58936c730a87 100644 --- a/trunk/sound/soc/omap/rx51.c +++ b/trunk/sound/soc/omap/rx51.c @@ -313,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); - err = omap_mcbsp_st_add_controls(rtd); + err = omap_mcbsp_st_add_controls(rtd->cpu_dai); if (err < 0) return err; @@ -353,7 +353,7 @@ static struct snd_soc_dai_link rx51_dai[] = { { .name = "TLV320AIC34", .stream_name = "AIC34", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", diff --git a/trunk/sound/soc/omap/sdp3430.c b/trunk/sound/soc/omap/sdp3430.c index 0e283226e2bf..2c850662ea7e 100644 --- a/trunk/sound/soc/omap/sdp3430.c +++ b/trunk/sound/soc/omap/sdp3430.c @@ -187,7 +187,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", @@ -199,7 +199,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", - .cpu_dai_name = "omap-mcbsp.3", + .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/omap/zoom2.c b/trunk/sound/soc/omap/zoom2.c index 920e0d9e03db..981616d61f67 100644 --- a/trunk/sound/soc/omap/zoom2.c +++ b/trunk/sound/soc/omap/zoom2.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", - .cpu_dai_name = "omap-mcbsp.2", + .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", @@ -143,7 +143,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", - .cpu_dai_name = "omap-mcbsp.3", + .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/trunk/sound/soc/samsung/neo1973_wm8753.c b/trunk/sound/soc/samsung/neo1973_wm8753.c index 321d51134e47..24bdb321269a 100644 --- a/trunk/sound/soc/samsung/neo1973_wm8753.c +++ b/trunk/sound/soc/samsung/neo1973_wm8753.c @@ -367,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753.0-001a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -376,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .stream_name = "Voice", .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753.0-001a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/trunk/sound/soc/soc-dapm.c b/trunk/sound/soc/soc-dapm.c index 6241490fff30..dcd11609f930 100644 --- a/trunk/sound/soc/soc-dapm.c +++ b/trunk/sound/soc/soc-dapm.c @@ -3238,13 +3238,9 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - if (dapm->bias_level == SND_SOC_BIAS_ON) - snd_soc_dapm_set_bias_level(dapm, - SND_SOC_BIAS_PREPARE); + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) - snd_soc_dapm_set_bias_level(dapm, - SND_SOC_BIAS_STANDBY); + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } } @@ -3257,9 +3253,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) - snd_soc_dapm_set_bias_level(&codec->dapm, - SND_SOC_BIAS_OFF); + snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); } } diff --git a/trunk/sound/spi/at73c213.c b/trunk/sound/spi/at73c213.c index c6500d00053b..4dd051bdf4fd 100644 --- a/trunk/sound/spi/at73c213.c +++ b/trunk/sound/spi/at73c213.c @@ -1112,7 +1112,17 @@ static struct spi_driver at73c213_driver = { .remove = __devexit_p(snd_at73c213_remove), }; -module_spi_driver(at73c213_driver); +static int __init at73c213_init(void) +{ + return spi_register_driver(&at73c213_driver); +} +module_init(at73c213_init); + +static void __exit at73c213_exit(void) +{ + spi_unregister_driver(&at73c213_driver); +} +module_exit(at73c213_exit); MODULE_AUTHOR("Hans-Christian Egtvedt "); MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); diff --git a/trunk/sound/usb/6fire/chip.c b/trunk/sound/usb/6fire/chip.c index fc8cc823e438..8af92e3e9c18 100644 --- a/trunk/sound/usb/6fire/chip.c +++ b/trunk/sound/usb/6fire/chip.c @@ -5,6 +5,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -28,7 +29,7 @@ #include MODULE_AUTHOR("Torsten Schenk "); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff --git a/trunk/sound/usb/6fire/chip.h b/trunk/sound/usb/6fire/chip.h index bde02d105a51..d11e5cb520f0 100644 --- a/trunk/sound/usb/6fire/chip.h +++ b/trunk/sound/usb/6fire/chip.h @@ -3,6 +3,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/comm.c b/trunk/sound/usb/6fire/comm.c index 6c3d531a250e..c994daa57af2 100644 --- a/trunk/sound/usb/6fire/comm.c +++ b/trunk/sound/usb/6fire/comm.c @@ -5,6 +5,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/comm.h b/trunk/sound/usb/6fire/comm.h index d2af0a5ddcf3..edc5dc84b888 100644 --- a/trunk/sound/usb/6fire/comm.h +++ b/trunk/sound/usb/6fire/comm.h @@ -3,6 +3,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/common.h b/trunk/sound/usb/6fire/common.h index b6eb03ed1c2c..7dbeb4a37831 100644 --- a/trunk/sound/usb/6fire/common.h +++ b/trunk/sound/usb/6fire/common.h @@ -3,6 +3,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/control.c b/trunk/sound/usb/6fire/control.c index 07ed914d5e71..ac828eff1a63 100644 --- a/trunk/sound/usb/6fire/control.c +++ b/trunk/sound/usb/6fire/control.c @@ -5,12 +5,9 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * - * Thanks to: - * - Holger Ruckdeschel: he found out how to control individual channel - * volumes and introduced mute switch - * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -19,7 +16,6 @@ #include #include -#include #include "control.h" #include "comm.h" @@ -28,6 +24,26 @@ static char *opt_coax_texts[2] = { "Optical", "Coax" }; static char *line_phono_texts[2] = { "Line", "Phono" }; +/* + * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$ + * this is done because the linear values cause rapid degredation + * of volume in the uppermost region. + */ +static const u8 log_volume_table[128] = { + 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34, + 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43, + 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e, + 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56, + 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, + 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62, + 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68, + 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, + 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, + 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, + 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, + 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, + 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f }; + /* * data that needs to be sent to device. sets up card internal stuff. * values dumped from windows driver and filtered by trial'n'error. @@ -43,7 +59,7 @@ init_data[] = { { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -54,47 +70,20 @@ static const int rates_altsetting[] = { 1, 1, 2, 2, 3, 3 }; static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; -static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); -static DECLARE_TLV_DB_MINMAX(tlv_input, -1500, 1500); - enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_output_vol_update(struct control_runtime *rt) +static void usb6fire_control_master_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; - int i; - - if (comm_rt) - for (i = 0; i < 6; i++) - if (!(rt->ovol_updated & (1 << i))) { - comm_rt->write8(comm_rt, 0x12, 0x0f + i, - 180 - rt->output_vol[i]); - rt->ovol_updated |= 1 << i; - } -} - -static void usb6fire_control_output_mute_update(struct control_runtime *rt) -{ - struct comm_runtime *comm_rt = rt->chip->comm; - - if (comm_rt) - comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); -} - -static void usb6fire_control_input_vol_update(struct control_runtime *rt) -{ - struct comm_runtime *comm_rt = rt->chip->comm; - int i; - - if (comm_rt) - for (i = 0; i < 2; i++) - if (!(rt->ivol_updated & (1 << i))) { - comm_rt->write8(comm_rt, 0x12, 0x1c + i, - rt->input_vol[i] & 0x3f); - rt->ivol_updated |= 1 << i; - } + if (comm_rt) { + /* set volume */ + comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - + log_volume_table[rt->master_vol]); + /* unmute */ + comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); + } } static void usb6fire_control_line_phono_update(struct control_runtime *rt) @@ -176,147 +165,34 @@ static int usb6fire_control_streaming_update(struct control_runtime *rt) return -EINVAL; } -static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, +static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 180; - return 0; -} - -static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - unsigned int ch = kcontrol->private_value; - int changed = 0; - - if (ch > 4) { - snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); - return -EINVAL; - } - - if (rt->output_vol[ch] != ucontrol->value.integer.value[0]) { - rt->output_vol[ch] = ucontrol->value.integer.value[0]; - rt->ovol_updated &= ~(1 << ch); - changed = 1; - } - if (rt->output_vol[ch + 1] != ucontrol->value.integer.value[1]) { - rt->output_vol[ch + 1] = ucontrol->value.integer.value[1]; - rt->ovol_updated &= ~(2 << ch); - changed = 1; - } - - if (changed) - usb6fire_control_output_vol_update(rt); - - return changed; -} - -static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - unsigned int ch = kcontrol->private_value; - - if (ch > 4) { - snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); - return -EINVAL; - } - - ucontrol->value.integer.value[0] = rt->output_vol[ch]; - ucontrol->value.integer.value[1] = rt->output_vol[ch + 1]; - return 0; -} - -static int usb6fire_control_output_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - unsigned int ch = kcontrol->private_value; - u8 old = rt->output_mute; - u8 value = 0; - - if (ch > 4) { - snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); - return -EINVAL; - } - - rt->output_mute &= ~(3 << ch); - if (ucontrol->value.integer.value[0]) - value |= 1; - if (ucontrol->value.integer.value[1]) - value |= 2; - rt->output_mute |= value << ch; - - if (rt->output_mute != old) - usb6fire_control_output_mute_update(rt); - - return rt->output_mute != old; -} - -static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - unsigned int ch = kcontrol->private_value; - u8 value = rt->output_mute >> ch; - - if (ch > 4) { - snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); - return -EINVAL; - } - - ucontrol->value.integer.value[0] = 1 & value; - value >>= 1; - ucontrol->value.integer.value[1] = 1 & value; - - return 0; -} - -static int usb6fire_control_input_vol_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; + uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 30; + uinfo->value.integer.max = 127; return 0; } -static int usb6fire_control_input_vol_put(struct snd_kcontrol *kcontrol, +static int usb6fire_control_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); int changed = 0; - - if (rt->input_vol[0] != ucontrol->value.integer.value[0]) { - rt->input_vol[0] = ucontrol->value.integer.value[0] - 15; - rt->ivol_updated &= ~(1 << 0); + if (rt->master_vol != ucontrol->value.integer.value[0]) { + rt->master_vol = ucontrol->value.integer.value[0]; + usb6fire_control_master_vol_update(rt); changed = 1; } - if (rt->input_vol[1] != ucontrol->value.integer.value[1]) { - rt->input_vol[1] = ucontrol->value.integer.value[1] - 15; - rt->ivol_updated &= ~(1 << 1); - changed = 1; - } - - if (changed) - usb6fire_control_input_vol_update(rt); - return changed; } -static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, +static int usb6fire_control_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = rt->input_vol[0] + 15; - ucontrol->value.integer.value[1] = rt->input_vol[1] + 15; - + ucontrol->value.integer.value[0] = rt->master_vol; return 0; } @@ -411,81 +287,16 @@ static int usb6fire_control_digital_thru_get(struct snd_kcontrol *kcontrol, return 0; } -static struct __devinitdata snd_kcontrol_new vol_elements[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Playback Volume", - .index = 0, - .private_value = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_output_vol_info, - .get = usb6fire_control_output_vol_get, - .put = usb6fire_control_output_vol_put, - .tlv = { .p = tlv_output } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Playback Volume", - .index = 1, - .private_value = 2, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_output_vol_info, - .get = usb6fire_control_output_vol_get, - .put = usb6fire_control_output_vol_put, - .tlv = { .p = tlv_output } - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Playback Volume", - .index = 2, - .private_value = 4, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_output_vol_info, - .get = usb6fire_control_output_vol_get, - .put = usb6fire_control_output_vol_put, - .tlv = { .p = tlv_output } - }, - {} -}; - -static struct __devinitdata snd_kcontrol_new mute_elements[] = { +static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Playback Switch", + .name = "Master Playback Volume", .index = 0, - .private_value = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_ctl_boolean_stereo_info, - .get = usb6fire_control_output_mute_get, - .put = usb6fire_control_output_mute_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Playback Switch", - .index = 1, - .private_value = 2, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_ctl_boolean_stereo_info, - .get = usb6fire_control_output_mute_get, - .put = usb6fire_control_output_mute_put, + .info = usb6fire_control_master_vol_info, + .get = usb6fire_control_master_vol_get, + .put = usb6fire_control_master_vol_put }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Playback Switch", - .index = 2, - .private_value = 4, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_ctl_boolean_stereo_info, - .get = usb6fire_control_output_mute_get, - .put = usb6fire_control_output_mute_put, - }, - {} -}; - -static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", @@ -513,54 +324,9 @@ static struct __devinitdata snd_kcontrol_new elements[] = { .get = usb6fire_control_digital_thru_get, .put = usb6fire_control_digital_thru_put }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Capture Volume", - .index = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_input_vol_info, - .get = usb6fire_control_input_vol_get, - .put = usb6fire_control_input_vol_put, - .tlv = { .p = tlv_input } - }, {} }; -static int usb6fire_control_add_virtual( - struct control_runtime *rt, - struct snd_card *card, - char *name, - struct snd_kcontrol_new *elems) -{ - int ret; - int i; - struct snd_kcontrol *vmaster = - snd_ctl_make_virtual_master(name, tlv_output); - struct snd_kcontrol *control; - - if (!vmaster) - return -ENOMEM; - ret = snd_ctl_add(card, vmaster); - if (ret < 0) - return ret; - - i = 0; - while (elems[i].name) { - control = snd_ctl_new1(&elems[i], rt); - if (!control) - return -ENOMEM; - ret = snd_ctl_add(card, control); - if (ret < 0) - return ret; - ret = snd_ctl_add_slave(vmaster, control); - if (ret < 0) - return ret; - i++; - } - return 0; -} - int __devinit usb6fire_control_init(struct sfire_chip *chip) { int i; @@ -586,26 +352,9 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip) usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_output_vol_update(rt); - usb6fire_control_output_mute_update(rt); - usb6fire_control_input_vol_update(rt); + usb6fire_control_master_vol_update(rt); usb6fire_control_streaming_update(rt); - ret = usb6fire_control_add_virtual(rt, chip->card, - "Master Playback Volume", vol_elements); - if (ret) { - snd_printk(KERN_ERR PREFIX "cannot add control.\n"); - kfree(rt); - return ret; - } - ret = usb6fire_control_add_virtual(rt, chip->card, - "Master Playback Switch", mute_elements); - if (ret) { - snd_printk(KERN_ERR PREFIX "cannot add control.\n"); - kfree(rt); - return ret; - } - i = 0; while (elements[i].name) { ret = snd_ctl_add(chip->card, snd_ctl_new1(&elements[i], rt)); diff --git a/trunk/sound/usb/6fire/control.h b/trunk/sound/usb/6fire/control.h index 9a596d95474a..8f5aeead2e3d 100644 --- a/trunk/sound/usb/6fire/control.h +++ b/trunk/sound/usb/6fire/control.h @@ -3,6 +3,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -43,11 +44,7 @@ struct control_runtime { bool line_phono_switch; bool digital_thru_switch; bool usb_streaming; - u8 output_vol[6]; - u8 ovol_updated; - u8 output_mute; - s8 input_vol[2]; - u8 ivol_updated; + u8 master_vol; }; int __devinit usb6fire_control_init(struct sfire_chip *chip); diff --git a/trunk/sound/usb/6fire/firmware.c b/trunk/sound/usb/6fire/firmware.c index 6f9715ab32fe..3b5f517a3972 100644 --- a/trunk/sound/usb/6fire/firmware.c +++ b/trunk/sound/usb/6fire/firmware.c @@ -5,6 +5,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/midi.c b/trunk/sound/usb/6fire/midi.c index f0e5179b242b..13f4509dce2b 100644 --- a/trunk/sound/usb/6fire/midi.c +++ b/trunk/sound/usb/6fire/midi.c @@ -5,6 +5,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/midi.h b/trunk/sound/usb/6fire/midi.h index 5114eccc1d8e..97a7bf669135 100644 --- a/trunk/sound/usb/6fire/midi.h +++ b/trunk/sound/usb/6fire/midi.h @@ -3,6 +3,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/pcm.c b/trunk/sound/usb/6fire/pcm.c index c97d05f0e966..d144cdb2f159 100644 --- a/trunk/sound/usb/6fire/pcm.c +++ b/trunk/sound/usb/6fire/pcm.c @@ -5,6 +5,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/6fire/pcm.h b/trunk/sound/usb/6fire/pcm.h index 3104301b257d..2bee81374002 100644 --- a/trunk/sound/usb/6fire/pcm.h +++ b/trunk/sound/usb/6fire/pcm.h @@ -3,6 +3,7 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 + * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/trunk/sound/usb/Kconfig b/trunk/sound/usb/Kconfig index ff77b28f3da1..3efc21c3d67c 100644 --- a/trunk/sound/usb/Kconfig +++ b/trunk/sound/usb/Kconfig @@ -106,7 +106,6 @@ config SND_USB_6FIRE select BITREVERSE select SND_RAWMIDI select SND_PCM - select SND_VMASTER help Say Y here to include support for TerraTec 6fire DMX USB interface. diff --git a/trunk/sound/usb/caiaq/audio.c b/trunk/sound/usb/caiaq/audio.c index fde9a7a29cb6..2cf87f5afed4 100644 --- a/trunk/sound/usb/caiaq/audio.c +++ b/trunk/sound/usb/caiaq/audio.c @@ -311,10 +311,8 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) spin_lock(&dev->spinlock); - if (dev->input_panic || dev->output_panic) { + if (dev->input_panic || dev->output_panic) ptr = SNDRV_PCM_POS_XRUN; - goto unlock; - } if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ptr = bytes_to_frames(sub->runtime, @@ -323,7 +321,6 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); -unlock: spin_unlock(&dev->spinlock); return ptr; } diff --git a/trunk/sound/usb/card.h b/trunk/sound/usb/card.h index da5fa1ac4eda..a39edcc32a93 100644 --- a/trunk/sound/usb/card.h +++ b/trunk/sound/usb/card.h @@ -1,7 +1,6 @@ #ifndef __USBAUDIO_CARD_H #define __USBAUDIO_CARD_H -#define MAX_NR_RATES 1024 #define MAX_PACKS 20 #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 diff --git a/trunk/sound/usb/format.c b/trunk/sound/usb/format.c index ddfef57c4c9f..e09aba19375c 100644 --- a/trunk/sound/usb/format.c +++ b/trunk/sound/usb/format.c @@ -209,6 +209,8 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } +#define MAX_UAC2_NR_RATES 1024 + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -253,7 +255,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; - if (nr_rates >= MAX_NR_RATES) { + if (nr_rates >= MAX_UAC2_NR_RATES) { snd_printk(KERN_ERR "invalid uac2 rates\n"); break; } diff --git a/trunk/sound/usb/pcm.c b/trunk/sound/usb/pcm.c index 0eed6115c2d4..0220b0f335b9 100644 --- a/trunk/sound/usb/pcm.c +++ b/trunk/sound/usb/pcm.c @@ -695,7 +695,6 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { struct audioformat *fp; - int *rate_list; int count = 0, needs_knot = 0; int err; @@ -709,8 +708,7 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, if (!needs_knot) return 0; - subs->rate_list.list = rate_list = - kmalloc(sizeof(int) * count, GFP_KERNEL); + subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL); if (!subs->rate_list.list) return -ENOMEM; subs->rate_list.count = count; @@ -719,7 +717,7 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, list_for_each_entry(fp, &subs->fmt_list, list) { int i; for (i = 0; i < fp->nr_rates; i++) - rate_list[count++] = fp->rate_table[i]; + subs->rate_list.list[count++] = fp->rate_table[i]; } err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &subs->rate_list); diff --git a/trunk/sound/usb/quirks.c b/trunk/sound/usb/quirks.c index 27817266867a..a3ddac0deffd 100644 --- a/trunk/sound/usb/quirks.c +++ b/trunk/sound/usb/quirks.c @@ -132,14 +132,10 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (!fp) { + if (! fp) { snd_printk(KERN_ERR "cannot memdup\n"); return -ENOMEM; } - if (fp->nr_rates > MAX_NR_RATES) { - kfree(fp); - return -EINVAL; - } if (fp->nr_rates > 0) { rate_table = kmemdup(fp->rate_table, sizeof(int) * fp->nr_rates, GFP_KERNEL); diff --git a/trunk/sound/usb/usx2y/usbusx2yaudio.c b/trunk/sound/usb/usx2y/usbusx2yaudio.c index 520ef96d7c75..6ffb3713b60c 100644 --- a/trunk/sound/usb/usx2y/usbusx2yaudio.c +++ b/trunk/sound/usb/usx2y/usbusx2yaudio.c @@ -80,7 +80,7 @@ static int usX2Y_urb_capt_retire(struct snd_usX2Y_substream *subs) cp = (unsigned char*)urb->transfer_buffer + urb->iso_frame_desc[i].offset; if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ snd_printk(KERN_ERR "active frame status %i. " - "Most probably some hardware problem.\n", + "Most propably some hardware problem.\n", urb->iso_frame_desc[i].status); return urb->iso_frame_desc[i].status; } @@ -300,7 +300,7 @@ static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, { snd_printk(KERN_ERR "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" +"Most propably some urb of usb-frame %i is still missing.\n" "Cause could be too long delays in usb-hcd interrupt handling.\n", usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", diff --git a/trunk/sound/usb/usx2y/usx2yhwdeppcm.c b/trunk/sound/usb/usx2y/usx2yhwdeppcm.c index 8e40b6e67e9e..a51340f6f2db 100644 --- a/trunk/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/trunk/sound/usb/usx2y/usx2yhwdeppcm.c @@ -74,7 +74,7 @@ static int usX2Y_usbpcm_urb_capt_retire(struct snd_usX2Y_substream *subs) } for (i = 0; i < nr_of_packs(); i++) { if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ - snd_printk(KERN_ERR "active frame status %i. Most probably some hardware problem.\n", urb->iso_frame_desc[i].status); + snd_printk(KERN_ERR "activ frame status %i. Most propably some hardware problem.\n", urb->iso_frame_desc[i].status); return urb->iso_frame_desc[i].status; } lens += urb->iso_frame_desc[i].actual_length / usX2Y->stride;