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[alsa-devel] [PATCH 0/9] ARM: DaVinci: ASoC: Adds Audio support for TI DM646X
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds Audio support for TI DM646X EVM platform Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch set adds audio support for DM646x This series is dependant on the following 2 patches submitted to the DaVinci git: [PATCH 1/2] ARM: DaVinci: EDMA: Move queue to tc mapping to dm<soc>.c [PATCH 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x Patch 1: Adds ASoC driver support for TI DM646X EVM platform Applies to the DaVinci git. Patch 2: Adds ALSA SoC McASP Audio Layer for TI DM646X processor Applies to the ASoC tree. Patch 3: Adds header file for McASP Audio Layer for TI DM646X processor Applies to the ASoC tree. Patch 4: Adds ALSA PCM interface for the TI DM646X processor Applies to the ASoC tree. Patch 5: Adds Audio Serial Port support for TI DM646X processor Applies to the DaVinci git. Patch 6: Adds McASP clock support for TI DM646X processor Applies to the DaVinci git. Patch 7: Adds ALSA SoC DIT driver support for DM646X Applies to the ASoC tree. Patch 8: ASoC: Adds header file for DM646X DIT driver Applies to the ASoC tree. Patch 9: Adds build support for DM646X audio driver Applies to the ASoC tree. These patches are based on the work done by Steve Chen <schen(a)mvista.com> on Monstavista 2.6.18 kernel. Even without these patches the ASoC tree (http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…) build for DaVinci is broken.

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[alsa-devel] [PATCH 9/9] ARM: DaVinci: ASoC: Adds build support for DM646X audio driver
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds build support for DM646X audio driver Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/Kconfig | 3 +++ sound/soc/codecs/Makefile | 2 ++ sound/soc/davinci/Kconfig | 14 ++++++++++++++ sound/soc/davinci/Makefile | 3 +++ 4 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f33c07..d34b38b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,9 @@ config SND_SOC_TLV320AIC23 tristate depends on I2C +config SND_SOC_CODEC_STUBS + tristate + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9c61037..3d1d270 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-codec_stubs-objs := codec_stubs.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -54,6 +55,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_CODEC_STUBS) += snd-soc-codec_stubs.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index b502741..ec554c9 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM @@ -18,6 +21,17 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM646X_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_CODEC_STUBS + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM646x EVM platform. + + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o -- 1.5.6

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[alsa-devel] [PATCH 7/9] ARM: DaVinci: ASoC: Adds ALSA SoC DIT/DIR driver support for DM646X
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds ALSA SoC DIT/DIR driver support for DM646X Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/codec_stubs.c | 63 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 63 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/codec_stubs.c diff --git a/sound/soc/codecs/codec_stubs.c b/sound/soc/codecs/codec_stubs.c new file mode 100644 index 0000000..60d3446 --- /dev/null +++ b/sound/soc/codecs/codec_stubs.c @@ -0,0 +1,63 @@ +/* + * ALSA SoC DaVinci DIT/DIR driver + * + * TI DaVinci audio controller can operate in DIT/DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai dit_stub_dai[] = { + { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +struct snd_soc_dai dir_stub_dai[] = { + { + .name = "DIR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +static int __init dit_modinit(void) +{ + return snd_soc_register_dai(dit_stub_dai); +} +module_init(dit_modinit); + +static void __exit dit_exit(void) +{ + snd_soc_unregister_dai(dit_stub_dai); +} +module_exit(dit_exit); + -- 1.5.6

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Re: [alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Takashi Iwai 18 Mar '09

18 Mar '09

Hi Giuliano, At Mon, 16 Mar 2009 09:14:20 +0100 (CET), Giuliano Pochini wrote: > > On Mon, 16 Mar 2009, Takashi Iwai wrote: > > >> This patch adds support for IndigoIOx and IndigoDJx. > >> > >> Signed-off-by Giuliano Pochini <pochini(a)shiny.it> > > > > The code looks almost good, but could you do the following? > > > > - split to two patches, one for kernel tree and one for alsa-driver > > tree > > There is a little problem: since about a month ago the git server is > SLOWWW. It takes ages to download the alsa-kernel repository at 5KB/s :( > I'll try from some server at work which stay on 24/7. > > > > - run checkpatch.pl against the patch and fix issues reported there > > Ok ! Any progress on this? I'm asking it because 2.6.29 kernel will likely go out soon, and all patches should be merged before the merge-window is opened. If you have no time, either we skip it or I'll do a quick fix. thanks, Takashi

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[alsa-devel] [PATCH 3/9] ARM: DaVinci: ASoC: Adds header file for McASP Audio Layer for TI DM646X processor
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds header file for McASP Audio Layer for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-i2s-mcasp.h | 312 +++++++++++++++++++++++++++++++++ 1 files changed, 312 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..7d2f7bb --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,312 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 + * published by the Free Software Foundation. + */ + + +#ifndef DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include "davinci-pcm.h" + + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define FREE BIT(0) +#define SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 +#define DISMOD(val) (val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ + /* and Transmit Bad Clock Detect /32 Counter + Reset */ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ -- 1.5.6

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[alsa-devel] Correct macro for Headset widgets
by Lopez Cruz, Misael 18 Mar '09

18 Mar '09

What is the correct way to declare a Headset widget in a machine driver? I originally thought that by using SND_SOC_DAPM_HP macro was enough, but it's not since the headset is both headphones and microphone. In my machine driver I have the following widget and map for headset: SND_SOC_DAPM_HP("Headset Jack", NULL) ... /* Headset: HSMIC (with bias), HSOL, HSOR */ {"Headset Jack", NULL, "HSOL"}, {"Headset Jack", NULL, "HSOR"}, {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Jack"}, but the "Headset Mic Bias" widget never gets powered when the "Headset Jack" pin is enabled. When the audio map is scanned, only valid routes are powered: DAC-Output, Input-DAC, Input-Output, DAC-ADC. And "Headset Jack" (declared as SND_SOC_DAPM_HP) is not a valid input endpoint, then the capture route becomes invalid. Can we have a macro specifically for Headset widgets? Thanks, -Misa

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[alsa-devel] DELL SP2309W Monitor with Integrated microphone
by Jason Holmes 18 Mar '09

18 Mar '09

Hi all, I have a Dell SP2309W monitor which contains an integrated webcam and microphones. I have been unable to get the integrated microphones to record any sound. I ran utils_alsa-info.sh and the link with the results in posted below: http://www.alsa-project.org/db/?f=590e122a466ce09cd1f911ba7146a87df2b5cbeb The integrated microphone seems to be made by OmniVision Technologies. The results of running arecord -l are: **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Webcam [Monitor Webcam], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: default [USB Sound Device ], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 and the results of arecord --device=default:CARD=Webcam test are: ALSA lib pcm_direct.c:866:(snd1_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dsnoop.c:604:(snd_pcm_dsnoop_open) unable to initialize slave arecord: main:590: audio open error: Invalid argument Thanks for your help. -- Jason Holmes

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[alsa-devel] Cleanup patches for snd-usb-caiaq
by Daniel Mack 18 Mar '09

18 Mar '09

Following up to this mail are 3 patches to clean up the snd-usb-caiaq driver. [PATCH 1/3] snd-usb-caiaq: only warn once on streaming errors [PATCH 2/3] snd-usb-caiaq: drop bogus iso packets [PATCH 3/3] snd-usb-caiaq: bump version number

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Re: [alsa-devel] Bad page state in process 'rmmod' cs4281
by "Timo T.Hyrsylä" 18 Mar '09

18 Mar '09

On 13.3.2009 13:04, Takashi Iwai wrote: > At Mon, 09 Mar 2009 12:24:51 +0100, > wrote: >> At Mon, 09 Mar 2009 10:32:02 +0200, >> Timo T.Hyrsylä wrote: >>> Trying to make Alsa work on our embedded arm system with ixdp425 and CS4281. >>> >>> kernel is 2.6.20.3 and the Alsas (driver+lib+utils) latest 1.0.19 >>> >>> Other than the two below errors, the system seems to work ok. >>> >>> When removing the snd_cs4281 module, dmesg reports multiple errors about >>> bad page >>> >>> <0>Bad page state in process 'rmmod' >>> <0>page:c4007300 flags:0x00004000 mapping:00000000 mapcount:0 count:0 >>> <0>Trying to fix it up, but a reboot is needed >>> <0>Backtrace: >>> <0>Bad page state in process 'rmmod' >>> <0>page:c4007320 flags:0x00004000 mapping:00000000 mapcount:0 count:0 >>> <0>Trying to fix it up, but a reboot is needed >>> <0>Backtrace: >>> >>> I've managed to track down the error upto dma_free_coherent. >>> >>> snd_pcm_free > snd_pcm_lib_preallocate_free_for_all > >>> snd_pcm_lib_preallocate_free > snd_pcm_lib_preallocate_dma_free > >>> snd_dma_free_pages > snd_free_dev_pages > dma_free_coherent >>> >>> >>> Another issue, possibly related, is that when running aplay with --mmap >>> it will crash with: >>> >>> Internal error: Oops: f3 [#1] >>> PC is at snd_pcm_mmap_data_nopage+0x10c/0x148 [snd_pcm] >>> LR is at __handle_mm_fault+0x16c/0x8ac >> This is likely the problem with dma_alloc_coherent() and mmap on >> a non-coherent architecture. I have patches for these, but only for >> recent kernels. The patches are found in test/dma-fix branch of sound >> git tree: >> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/dma-fix > > Alternatively, an easier "fix" would be simply disable mmap support. > Remove SNDRV_PCM_INFO_MMAP* from snd_pcm_hardware definitions. > > This may disable the conversion plugin usage in the earlier alsa-lib > versions, though. > > > Takashi Tried to apply the patch changes into my version, but it did not affect the unloading 'bad page' problem at all. Instead it did make all play attempts (even non MMAP) to crash. (no crash details available since I desided to forget about it for now, working on unrelated issues for awhile) I did disable the MMAP so that player wont crash when requesting MMAP. System still dumps the bad pages at unload, play and capture seem to work OK, not tested extensively thou. -- Timo

1 0

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 17 Mar '09

17 Mar '09

The following changes since commit 852fd9e50f62b4ea7afe26eee0710464de4801b8: Mark Brown (1): ASoC: Each PXA AC97 DAI needs a separate ops are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Atsushi Nemoto (1): ASoC: Only deregister AC97 dev if it's name was not "AC97" Daniel Mack (1): ASoC: switch PXA SSP driver from network mode to PSP Mark Brown (3): ASoC: Fix non-networked I2S mode for PXA SSP ASoC: Fix Zylonite for non-networked SSP mode Merge branch 'pxa-ssp' into for-2.6.30 sound/soc/pxa/pxa-ssp.c | 55 ++++++++++++++++++++++++++++++++++++---------- sound/soc/pxa/zylonite.c | 55 +++++++++++++++++++++------------------------- sound/soc/soc-core.c | 3 +- 3 files changed, 70 insertions(+), 43 deletions(-)

2 1

[alsa-devel] Tenx USB audio adapter
by PETER EASTHOPE 17 Mar '09

17 Mar '09

Folk, I was advised to ask about the Tenx USB audio adapter here. lshal claims it is a standard HID device and a standard USB device. Ref. http://carnot.yi.org/lshal1.out Nevertheless, sound in and sound out both fail in my system. Some information is at the bottom of this page. "http://carnot.yi.org/DebianPage.html#USBAudio" Any suggestions? Thanks, ... Peter E. -- http://members.shaw.ca/peasthope/ http://carnot.yi.org/ = http://carnot.pathology.ubc.ca/

1 0

[alsa-devel] Fedora 10 x86_64 alsa problem: Realtek ALC889A Audio Codec(ATI azalia) not detected on alsa
by Emmanuel Chanel 17 Mar '09

17 Mar '09

I wrote this on https://bugzilla.redhat.com/show_bug.cgi?id=489828 I've used the driver of kernel package. And Mr. Jaroslav Kysela replied me that I could try snapshot source. (Thx, Mr. Jaroslav Kysela. And on #alsa on Freenode, Thingol and gnubien advised me much. Thank you!) But I couldn't get sound back none of both: stable(alsa-driver-1.0.19) and snapshot (alsa-driver-1.0.19.34.g70f78.363.g0ec3b). My alsa info is on http://www.alsa-project.org/db/?f=af6a6e787d4bd5212bf766bf5d09fda1ab5e2300 I use desktop. Its mother board is GA-MA69H-S3H . The soundcard is onboard. Realtek ALC889A Audio Codec. # lspci |grep -i "multi\|audio\|040[0-3]" # Audio device: ATI Technologies Inc SB600 Azalia (Intel HDA) When I installed stable release of alsa-driver --with-debug=verbose , dmesg says HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:2218: chipset global capabilities = 0x4401 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:786: codec_mask = 0x8 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:1214: hda_intel: codec #3 probed OK ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/../../alsa-kernel/pci/hda/hda_codec.c:788: hda_codec: no AFG or MFG node found ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:1298: hda-intel: no codecs initialized HDA Intel 0000:00:14.2: PCI INT A disabled -- Emmanuel Chanel <emmanuelchanel(a)gmail.com>

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[alsa-devel] How to tell user level hardware not support a certain format in a certain sample rate
by Richard Zhao 17 Mar '09

17 Mar '09

Hi, runtime->hw.rates and runtime->hw.formats indicate the rate and formats hardware can support, so the user level plugins can auto-convert the audio data. But hardware may not support a certain combin of sample rate and format, and will cause hw_parm fail. In such case, plugin auto-convert will not work. How can I let userlevel know the failure conbin, so let auto-convert work? Thanks Richard

3 6

[alsa-devel] Need help on getting raw audio stream from a dummy sound card
by Santo Chow 17 Mar '09

17 Mar '09

Hi everyone :) I'm currently assigned to a task that requires me to capture raw audio stream from a dummy sound card. Is this even possible? Previously, I have been able to capture raw audio stream from my default sound card, by using the following example I got from http://www.linuxjournal.com/article/6735 /* This example reads from the default PCM device and writes to standard output for 5 seconds of data. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); loops = 5000000 / val; while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); return 0; } The above example allows me to capture 5 seconds of raw audio. I can use aplay to play the recorded sound and it plays nicely. But right now, I'm working on a project that involves the usage of a small development board. Basically, this board does not have a sound card or an actual speaker. So I thought, I can use the dummy soundcard provided by the linux kernel by calling out modprobe snd-dummy. Right now, I'm still testing it in my PC so to simulate the board's environment. I have configured the .asoundrc file and create a new dummy pcm by putting the lines below onto my .asoundrc file. pcm.dummy{ type hw card Dummy } So, I change the above ecample code, so it listens to this new dummy pcm rather than the 'default'. Like this : rc = snd_pcm_open(&handle, "dummy", SND_PCM_STREAM_CAPTURE, 0); Supposedly, the above modification would allow me to record from this dummy pcm, no? I do this by playing a song, using aplay -f cd -D dummy song.wav and at the same time, execute the above example. As expected, no sound was coming out from my speaker. When the program finished recording for 5 seconds, I play back the result, but all i heard was just noise. Of curiousity, I try using file plugins. I modify my .asoundrc file like this: pcm.dummy{ type plug slave{ pcm file format S16_LE channels 2 rate 44100 } } pcm.file{ type file slave{ pcm d } file /home/mydir/out.raw } pcm.d{ type hw card Dummy } Then i called aplay -f cd -D dummy song.wav again, well.. still no sound coming out from the speaker, but it does output the raw audio file into my /home/mydir/out.raw file. I play the out.raw using aplay -f cd /home/mydir/out.raw it's flawless. But I can't use this for my implementation. What I need is actually a way to read raw audio data (or stream) from the dummy sound card, to a buffer inside my program. I need this because I'm going to stream the buffer to my server, so I can listen the sound from my server. I can't afford to use the file plugin approach, basically because later on, in my development board, i won't have that much space. So, the question is: capturing raw audio data from a dummy soundcard, is this possible? I'm pretty sure that if the file plugin works, means that the raw audio data is there. It's just that i'm probably doing a wrong approach to read it. Hence, needs explanation and help.. Please help me.. :( Thank you so much for reading my long request.

2 3

Re: [alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 16 Mar '09

16 Mar '09

Alan Jenkins wrote: > Maxim Levitsky wrote: >> I have just bought an Aspire one A150, XP version, >> as it was the only available here, and installed ubuntu on it. >> >> Bugs I discovered so far: >> >> >> ** 1 - embedded controler works in polling mode, due to this: >> >> [ 0.708571] ACPI: EC: non-query interrupt received, switching to interrupt mode >> [ 1.224043] ACPI: EC: missing confirmations, switch off interrupt mode. >> >> >> Maybe this is the reason for the fact that gnome power manager freezes when I unplug >> the AC, and freezes often when I try to see battery status. >> >> >> (Note: same is seen on my acer aspire 5720) > > That sounds like a known issue. It has been resolved by "ACPI: EC: revert msleep patch". Happily Len submitted it for mainline this week. You will also find it if you try the acpi-test git tree. We're all hoping 2.6.28 will be much improved in terms of reliable operation of different ECs :). > Yes, this almost fixes all issues with that. Almost because, it looks like the EC changes screen brightness on his own when battery is plugged/unplugged, but so does the gnome-power-manager, and thus it still hangs as before on battery removal (but doesn't hang otherwise) I disabled that behaver in gnome-power-manager and now no more hangs. I see that this fix fixes the polled mode. Any chance to make irq mode work? >> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by default in ubuntu... >> wireless more or less works, but kernel log is full of backtraces. > > Well, that doesn't tell us much. Did they still happen after upgrading to 2.6.28-rc3? Can we see them? Fixed completely, looking through the git log I have seen a commit that fixes exactly same backtrace I had. No more errors in kernel log. > >> Was able to connect to my WPA2 access point. >> Sometimes wireless fails completely, especially after suspend to ram. >> Advanced features like monitor/injection work, but when I changed the card's >> mac address it stopped working. >> I also noticed that if I then start airodump, then wireless works with new mac. This still not fixed, I ask about this again in separate mail. >> >> >> >> ** 3 - internal mic doesn't work. >> tried model=acer, model=auto. >> Overall it seems that alsa misprograms O/B realteck >> codec. >> I talk about this later. Surprisingly, this is almost fixed. I can record from both internal mic and external mic (Quality is not so good, but is the same as in windows) Almost, is that to increase capture level, I have to set "one" channel of it. Eg: if I set 100% left and 100% right then capture level is very low. But when I set 100% left and 0% right or vice versa it works fine. Also the "mic boost" doesn't change capture volume for internal mic. also the "mic boost" if set to high value works like analog loopback (I hear what I say in mic in headphones, same happens on my main notebook acer 5720) >> Have same issue on my acer 5720 >> >> >> >> ** 4 - wireless led doesn't work. >> ath5k devs, can you fix this? >> >> >> ** 5 - coretemp doesn't show cpu temperature, >> I have seen somewhere that atom support same thermal diode as core2 >> and only patch to detect it is needed. >> Please include such path in 2.6.28 if exists. >> >> >> ** 6 - both card readers are missing from lspci, is this normal? > > A similar bug has been reported as a regression: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11828> > > so one assumes that it worked on the machines with linux pre-installed. Hopefully without requiring any hacks. > > It seems that for now a workaround may be to pass the option debug_quirks=1 to the sdhci module... > > <http://marc.info/?l=linux-kernel&m=122509648027303&w=2> > > ...or that it may help if you insert an SD card before booting. > > Apparently the reporter also investigated pcie hotplug. Probably the BIOS doesn't provide the normal support. You can try "modprobe pciehp pciehp_force=1", maybe it helps the kernel discover the devices. It worked for something else on my EeePC. But then it will reportedly disappear the ethernet controller. > > However, at the moment pciehp can cause delays of 10s of seconds during resume. I am aware of this, but don't yet have a SD card to test. > >> ** 7 - opengl is broken, I tried running neverball in fullscreen, but it shows wrong/corrupted >> textures (compiz is running, and working well thought) Without compiz neveball works fine (slow, but this is very slow hardware) >> >> Now I installed 2.6.28-rc3 (actually latest -git) >> >> >> Sadly nothing improved. >> I only found few regressions: >> >> >> >> * Wireless now drops connection from my WPA2 access point after few minutes. >> I changed temporarily its wireless protection to WEP and open mode, and both work >> fine, maybe this is related to periodic key exchange WPA does? Now fixed, big thanks. >> >> * System doesn't reboot/shutdown, just hangs. > > Could be this: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11942> > > As a fix, 8fd145917fb62368a9b80db59562c20576238f5a was reverted from the acpi tree. This has also been submitted to mainline. Also fixed big thanks. Also noticed another bug: If I suspend/resume with compiz running, on resume I see wallpaper and mouse cursor, everything hangs for minute or two, and sometimes forever. 2.6.27 worked fine. Also hibernate doesn't work on my main notebook, but it is probably fixed, I update kernel to really latest -git and tell you. Big thanks for bugfixes, you saved me a lot of work and bisecting. Biggest solvable problem now is sound support now I think, datasheets are there, and I take a look at them. Best regards, Maxim Levitsky

3 25

[alsa-devel] [PATCH 6/9] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds McASP clock support for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index af040cf..0aee774 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -185,6 +185,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -253,6 +265,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK("soc-audio.0", NULL, &mcasp0_clk), + CLK("soc-audio.1", NULL, &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

2 1

[alsa-devel] [PATCH 2/9] ARM: DaVinci: ASoC: Adds ALSA SoC McASP Audio Layer for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds ALSA SoC McASP Audio Layer for TI DM646X processor. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-i2s-mcasp.c | 641 +++++++++++++++++++++++++++++++++ 1 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..eb632cd --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,641 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 + * published by the Free Software Foundation. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + mcasp_set_bits(regs, val); + while ((mcasp_get_reg(regs) & val) != val) + ; +} +int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} +EXPORT_SYMBOL(davinci_i2s_startup); + +int davinci_i2s_mcasp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int tmp, link_cnt; + int count = 0; + int backup_count = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai; + struct davinci_audio_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct evm_snd_platform_data *pdata; + struct davinci_pcm_dma_params *dma_data; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * card->num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (card->num_links << 1), GFP_KERNEL); + if (!dma_data) + goto err_release_dev; + + for (link_cnt = 0; link_cnt < card->num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + backup_count = 0; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + backup_count = 1; + goto err_release_data; + } + + + cpu_dai = card->dai_link[link_cnt].cpu_dai; + cpu_dai->private_data = &dev[link_cnt]; + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, NULL); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + backup_count = 2; + goto err_release_data; + } + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].channel = pdata->tx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].channel = pdata->rx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + count++; + + } + return 0; + +err_release_data: + for (tmp = link_cnt; tmp >= 0; tmp--) { + if (backup_count > 2) + clk_disable(dev[tmp].clk); + + if (backup_count > 1) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } + backup_count = 3; + } + kfree(dma_data); + +err_release_dev: + kfree(dev); + + return ret; +} +EXPORT_SYMBOL(davinci_i2s_mcasp_probe); + +void davinci_i2s_mcasp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int i; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct davinci_audio_dev *dev_list = cpu_dai->private_data; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *ptr; + + for (i = 0; i < card->num_links; i++) { + dev = &dev_list[i]; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + ptr = dev_list->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(ptr); + kfree(dev_list); +} +EXPORT_SYMBOL(davinci_i2s_mcasp_remove); + + +void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + /* wait for TX ready */ + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(0)) + & TXSTATE)) + ; + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03 << 4; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05 << 4; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07 << 4; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09 << 4; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B << 4; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D << 4; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F << 4; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, fmt, (0x0F << 4)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, fmt, (0x0F << 4)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} +static void davinci_hw_iis_param(struct snd_pcm_substream *substream) +{ + int i, active_slots; + u32 mask = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + + } else { + /* bit stream is MSB first with no delay */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + FSRDLY(1) | RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_iis_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + }, + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + }, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int __init davinci_i2s_mcasp_init(void) +{ + return snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + return snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + -- 1.5.6

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.29-rc8-519-g236e303
by Takashi Iwai 16 Mar '09

16 Mar '09

At Mon, 16 Mar 2009 15:01:57 +0100 (CET), noreply-git(a)alsa-project.org wrote: > > Hello, > > This is an automated email from the git hooks/update script, it was > generated because a ref change was pushed to the repository. > > Updating branch, master, > via 236e3036199f7ef6817967cb4aee5e6686ef2d52 (commit) > from 1cebbf3bef5ee355894375b92946911068130e22 (commit) > > - Log ----------------------------------------------------------------- > commit 236e3036199f7ef6817967cb4aee5e6686ef2d52 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Mon Mar 16 14:56:58 2009 +0100 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Mon Mar 16 15:00:48 2009 +0100 > > ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index > > ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID. > Confirmed by testing on real hardware. ALC662 has really the index 0x05 although ALC663 and ALC272 have 0x04. Fixed on sound git tree now... Takashi > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> > > ----------------------------------------------------------------------- > > Diffstat: > sound/pci/hda/patch_realtek.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > > Patch -------------- (max 1500 lines) > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index b794cba..672103d 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -16951,7 +16951,7 @@ static int patch_alc662(struct hda_codec *codec) > > if (!spec->cap_mixer) > set_capture_mixer(spec); > - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); > + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); > > spec->vmaster_nid = 0x02; > > ------- End of patch > > hooks/update > --- > Git Source Code Management System > hooks/update refs/heads/master \ > 1cebbf3bef5ee355894375b92946911068130e22 \ > 236e3036199f7ef6817967cb4aee5e6686ef2d52 > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

1 0

[alsa-devel] [PATCH 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3 -- - Joonyoung Shim

4 5

[alsa-devel] [PATCH RESEND 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3

2 1

[alsa-devel] [PATCH RESEND 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3

1 0

[alsa-devel] [PATCH 8/9] ARM: DaVinci: ASoC: Adds header file for DM646X DIT/DIR driver
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds header file for DM646X DIT/DIR driver Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/codec_stubs.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/codec_stubs.h diff --git a/sound/soc/codecs/codec_stubs.h b/sound/soc/codecs/codec_stubs.h new file mode 100644 index 0000000..693797b --- /dev/null +++ b/sound/soc/codecs/codec_stubs.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * + * 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 + * published by the Free Software Foundation. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +#ifdef CONFIG_SND_SOC_CODEC_STUBS +extern struct snd_soc_dai dit_stub_dai[]; +extern struct snd_soc_dai dir_stub_dai[]; +#else +#define dit_stub_dai NULL +#define dir_stub_dai NULL +#endif + +#endif /* CODEC_STUBS_H */ -- 1.5.6

1 0

[alsa-devel] [PATCH 5/9] ARM: DaVinci: ASoC: Adds Audio Serial Port support for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds Audio Serial Port support for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. arch/arm/mach-davinci/include/mach/asp.h | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..670afe9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -6,16 +6,26 @@ #include <mach/irqs.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT -- 1.5.6

1 0

[alsa-devel] [PATCH 4/9] ARM: DaVinci: ASoC: Adds ALSA PCM interface for the TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds ALSA PCM interface for the TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-pcm.h | 22 ++++++++++++++++++---- 1 files changed, 18 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..c7d8711 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,16 +12,30 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; struct evm_snd_platform_data { + char *clk_name; int tx_dma_ch; int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; }; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

1 0

[alsa-devel] Using snd-aloop as default soundcard (need urgent help T.T)
by Santo Chow 16 Mar '09

16 Mar '09

Hi every one.. I'm currently on a project that requires me to capture a raw audio data from a dummy soundcard. I've so far manage to do that, by using snd-aloop. Now the problem i have is.. when i'm running several audio program at the same time, i can only hear the sound from the first program that i ran. i've followed the example os dmix, but this doesn't help either. So, right now, i have a Loopback soundcard as my default. My .asoundrc looks like this: pcm.!default{ type hw card 1 } Card 0 is my actual soundcard (that can actually produce sound). Card 1, when i checked on the /proc/asound/cards actually pointing to Loopback card. So right now, the way i test the card 1, is to call arecord from card 1, and play it to card 0. Right now, i'm still developing this in my ubuntu PC. so i ran this arecord -f cd -D hw:1,1 | aplay -f cd -D hw:0 in one terminal, and in the other terminal i can play mplayer, watching youtube, etc.. and the sound would come out nicely. (Thanks to all of your efforts, nevertheless) But I notice that, when i play mplayer, then i open a web-browser, and surf youtube, the sound from youtube is gone. But if i close the browser, close the mplayer, then reopen the browser, go to youtube and play a video, i can hear the sound nicely. but then, if during the time i start mplayer, it will complain about 'no sound'. Turns out, I have to use dmix. So, i modify my .asoundrc like this: pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:0,0" rate 44100 } bindings { 0 0 1 1 } } i put 'pcm "hw:0,0" ' over there, because i want to make sure that this setting works with my actual sound card first. If it works with my card 0, it should work with my card 1. with the above .asoundrc file, i can play multiple sound from different application at the same time. I even tried to remove the "bindings" tag, and it still able to play multiple sound very well. so i guess, this should work on my card 1. then i change the .asoundrc become like this again: pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:1,1" period_time 0 period_size 881 buffer_size 3524 rate 44100 } }the value for perio_time, period_size, and buffer_size have been adjusted by me to suits my hardware requirement. This is the setting that works on my machine. It's just that, in the end, I'm still ended up with one sound at a time. I can't run mplayer + youtube at the same time. I can only run one of them. And furthermore, if i just paused the mplayer, the audio device is still locks. It will be good enough if someone can hint me a way to unlock the audio device. So, what i want to ask is.. can this Loopback card handle more than 1 sound at a time? Have anyone tried this before? I have tried changing the value of .asoundrc, switching the slave pcm to pcm "hw:1,0" (not working) pcm "hw:1" (works, but only 1 sound too). Any idea guys? -santo-

1 0

[alsa-devel] Problem with Pulse Audio plugin?
by Tom Watson 16 Mar '09

16 Mar '09

I'm doing some experimenting and found that a simple sequential play of a small sound file caused an assertion in the pulse audio plugin. The base system is Fedora 8, and it uses alsa 1.0.16 versions. I looked at the 1.0.19 versions and they still have assertions, so the problem may still be there. It is pretty easy to trip up: $ aplay blah.wav blah.wav Playing WAVE 'blah.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Playing WAVE 'blah.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_pulse.c:524: pulse_hw_params: Assertion `!pcm->stream' failed. Aborted by signal Aborted... (where "blah.wav" is a simple short wav file) The first time through, it plays find, but the second time it doesn't. If this has been solved in 1.0.19, (I haven't installed it), please let me know and I'll go away. If not, it may be a problem in the pulse audio thing. Thanks. -- Tom Watson tsw(a)johana.com

2 2

Re: [alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Takashi Iwai 16 Mar '09

16 Mar '09

At Mon, 16 Mar 2009 09:14:20 +0100 (CET), Giuliano Pochini wrote: > > > > On Mon, 16 Mar 2009, Takashi Iwai wrote: > > >> This patch adds support for IndigoIOx and IndigoDJx. > >> > >> Signed-off-by Giuliano Pochini <pochini(a)shiny.it> > > > > The code looks almost good, but could you do the following? > > > > - split to two patches, one for kernel tree and one for alsa-driver > > tree > > There is a little problem: since about a month ago the git server is > SLOWWW. It takes ages to download the alsa-kernel repository at 5KB/s :( > I'll try from some server at work which stay on 24/7. Better to try my sound git tree on kernel.org. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git The bare alsa-driver tree is: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git The corresponding alsa-driver snapshot is: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Takashi

1 0

[alsa-devel] [PATCH 1/1] echomixer: Fix echomixer to work with the new drivers
by Giuliano Pochini 16 Mar '09

16 Mar '09

There is a long standing bug in the drivers for cards with a vmixer because I overlooked a detail in the c++ generic driver by echoaudio. Those cards do not have a line-out volume control. It is a virtual control provided by the generic driver. The bug is harmless because the DSP just ignores the command to change the volume. Since that control has been removed, echomixer must be updated. With this patch it uses the vmixer to fake the line-out volume. This patch makes echomixer work with the new drivers. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> --- alsa-tools-1.0.19/echomixer/echomixer.c__orig2 2009-03-15 19:03:24.000000000 +0100 +++ alsa-tools-1.0.19/echomixer/echomixer.c 2009-03-15 19:36:23.000000000 +0100 @@ -1178,6 +1178,29 @@ void UpdateMixerVolume(int outchannel) { +// Changes the vmixer volume according to the current Line-out volume for vmixer cards. +void UpdateVMixerVolume(int outchannel) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_value_t *control; + int err, val, ch; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_value_alloca(&control); + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); + + for (ch=0; ch<nPOut; ch++) { + val=Add_dB(vmixerControl.mixer[outchannel][ch].Gain, lineoutControl.Gain[outchannel]); + ClampOutputVolume(&val); + snd_ctl_elem_id_set_numid(id, vmixerControl.mixer[outchannel][ch].id); + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_value_set_integer(control, 0, val); + if ((err = snd_ctl_elem_write(ctlhandle, control)) < 0) + printf("Control %s element write error: %s\n", card, snd_strerror(err)); + } +} + + + void LineOut_volume_changed(GtkWidget *widget, gpointer ch) { char str[16]; int err, channel, val; @@ -1204,6 +1227,9 @@ void LineOut_volume_changed(GtkWidget *w snd_ctl_elem_value_set_integer(control, channel, val); if ((err=snd_ctl_elem_write(ctlhandle, control))<0) printf("Control %s element write error: %s\n", card, snd_strerror(err)); + } else if (vmixerId) { + UpdateVMixerVolume(channel); + UpdateMixerVolume(channel); } else { // Otherwise we have to emulate it. UpdatePCMVolume(channel); UpdateMixerVolume(channel); @@ -1217,10 +1243,11 @@ void LineOut_volume_changed(GtkWidget *w void Vmixer_volume_changed(GtkWidget *widget, gpointer ch) { char str[16]; - int val, channel; + int val, rval, channel; int o, v; channel=(int)(long)ch; + val=INVERT((int)GTK_ADJUSTMENT(widget)->value); #ifdef REVERSE v=channel; @@ -1230,13 +1257,17 @@ void Vmixer_volume_changed(GtkWidget *wi o=channel; #endif - val=INVERT((int)GTK_ADJUSTMENT(widget)->value); + // Emulate the line-out volume if this card can't do it in hw. + if (!lineoutId) { + rval=Add_dB(val, lineoutControl.Gain[o]); + ClampOutputVolume(&rval); + } - SetMixerGain(&vmixerControl.mixer[o][v], val); + SetMixerGain(&vmixerControl.mixer[o][v], rval); vmixerControl.mixer[o][v].Gain=val; if (Gang) { - SetMixerGain(&vmixerControl.mixer[o^1][v^1], val); + SetMixerGain(&vmixerControl.mixer[o^1][v^1], rval); vmixerControl.mixer[o^1][v^1].Gain=val; } -- Giuliano.

2 1

[alsa-devel] [PATCH 1/2] Echoaudio: remove line-out volume from vmixer cards
by Giuliano Pochini 16 Mar '09

16 Mar '09

There is a long standing bug in the drivers for cards with a vmixer because I overlooked a detail in the c++ generic driver by echoaudio. Those cards do not have a line-out volume control. It is a virtual control provided by the generic driver. The bug is harmless because the DSP just ignores the command to change the volume. *NB:* It breaks alsa-tools/echomixer. A patch for it will follow. This patch removes the line-out volume control from vmixer-equipped cards. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.c 2009-03-15 16:51:17.000000000 +0100 @@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(st Control interface ******************************************************************************/ +#ifndef ECHOCARD_HAS_VMIXER + /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(stru return changed; } -#ifdef ECHOCARD_HAS_VMIXER -/* On Vmixer cards this one controls the line-out volume */ -static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { - .name = "Line Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = snd_echo_output_gain_info, - .get = snd_echo_output_gain_get, - .put = snd_echo_output_gain_put, - .tlv = {.p = db_scale_output_gain}, -}; -#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_ .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; + #endif @@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(stru #ifdef ECHOCARD_HAS_VMIXER snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) - goto ctl_error; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; #else -- Giuliano.

2 1

[alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Giuliano Pochini 16 Mar '09

16 Mar '09

This patch adds support for IndigoIOx and IndigoDJx. I tested IndigoIOx myself, and Carmen Rocco tested the IndigoDJx. It wasn't tested extensively, but I think it stable enough. Maybe you want to mark it as experimental anyway. The soundcard matrix should be updated as well, but the page is locked. These two cards have the same features of their respective CARDBUS versions. The links to the manufacturer's site are: http://www.echoaudio.com/Products/ExpressCard/IndigoDJx/ and http://www.echoaudio.com/Products/ExpressCard/IndigoIOx/ . I want to thank Echoaudio again for their support. This patch adds support for IndigoIOx and IndigoDJx. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.h alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio_dsp.h --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.h 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio_dsp.h 2009-03-15 17:38:47.000000000 +0100 @@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command. #define E3G_ASIC_NOT_LOADED 0xffff #define E3G_BOX_TYPE_MASK 0xf0 -#define EXT_3GBOX_NC 0x01 -#define EXT_3GBOX_NOT_SET 0x02 +/* Indigo express control register values */ +#define INDIGO_EXPRESS_32000 0x02 +#define INDIGO_EXPRESS_44100 0x01 +#define INDIGO_EXPRESS_48000 0x00 +#define INDIGO_EXPRESS_DOUBLE_SPEED 0x10 +#define INDIGO_EXPRESS_QUAD_SPEED 0x04 +#define INDIGO_EXPRESS_CLOCK_MASK 0x17 /* diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.h 2009-03-15 17:14:39.000000000 +0100 @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx.c 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,107 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> + * + * 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; version 2 of the License. + * + * 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. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_DJX +#define ECHOCARD_NAME "Indigo DJx" +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 0 */ +#define PX_DIGITAL_IN 8 /* 0 */ +#define PX_NUM 8 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 4 */ +#define BX_DIGITAL_OUT 4 /* 0 */ +#define BX_ANALOG_IN 4 /* 0 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/io.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_DJX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigodjx_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx_dsp.c 2009-03-15 17:17:50.000000000 +0100 @@ -0,0 +1,65 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo DJx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) + return -ENODEV; + + if ((err = init_dsp_comm_page(chip))) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + if ((err = load_firmware(chip)) < 0) + return err; + chip->bad_board = FALSE; + + if ((err = init_line_levels(chip)) < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_express_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigo_express_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_express_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigo_express_dsp.c 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,119 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int set_sample_rate(struct echoaudio *chip, u32 rate) +{ + u32 clock, control_reg, old_control_reg; + + if (wait_handshake(chip)) + return -EIO; + + old_control_reg = le32_to_cpu(chip->comm_page->control_register); + control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK; + + switch (rate) { + case 32000: + clock = INDIGO_EXPRESS_32000; + break; + case 44100: + clock = INDIGO_EXPRESS_44100; + break; + case 48000: + clock = INDIGO_EXPRESS_48000; + break; + case 64000: + clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 88200: + clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 96000: + clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + default: + return -EINVAL; + } + + control_reg |= clock; + if (control_reg != old_control_reg) { + chip->comm_page->control_register = cpu_to_le32(control_reg); + chip->sample_rate = rate; + clear_handshake(chip); + return send_vector(chip, DSP_VC_UPDATE_CLOCKS); + } + return 0; +} + + + +/* This function routes the sound from a virtual channel to a real output */ +static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, + int gain) +{ + int index; + + if (snd_BUG_ON(pipe >= num_pipes_out(chip) || + output >= num_busses_out(chip))) + return -EINVAL; + + if (wait_handshake(chip)) + return -EIO; + + chip->vmixer_gain[output][pipe] = gain; + index = output * num_pipes_out(chip) + pipe; + chip->comm_page->vmixer[index] = gain; + + DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + return 0; +} + + + +/* Tell the DSP to read and update virtual mixer levels in comm page. */ +static int update_vmixer_level(struct echoaudio *chip) +{ + if (wait_handshake(chip)) + return -EIO; + clear_handshake(chip); + return send_vector(chip, DSP_VC_SET_VMIXER_GAIN); +} + + + +static u32 detect_input_clocks(const struct echoaudio *chip) +{ + return ECHO_CLOCK_BIT_INTERNAL; +} + + + +/* The IndigoIO has no ASIC. Just do nothing */ +static int load_asic(struct echoaudio *chip) +{ + return 0; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox.c 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,109 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> + * + * 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; version 2 of the License. + * + * 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. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_IOX +#define ECHOCARD_NAME "Indigo IOx" +#define ECHOCARD_HAS_MONITOR +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 2 */ +#define PX_DIGITAL_IN 10 /* 0 */ +#define PX_NUM 10 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 2 */ +#define BX_DIGITAL_OUT 2 /* 0 */ +#define BX_ANALOG_IN 2 /* 2 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/io.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_IOX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigoiox_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" + diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox_dsp.c 2009-03-15 17:17:38.000000000 +0100 @@ -0,0 +1,65 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo IOx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) + return -ENODEV; + + if ((err = init_dsp_comm_page(chip))) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + if ((err = load_firmware(chip)) < 0) + return err; + chip->bad_board = FALSE; + + if ((err = init_line_levels(chip)) < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/Makefile alsa-driver-1.0.19/sound/pci/echoaudio/Makefile --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/Makefile 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/Makefile 2009-03-15 17:14:39.000000000 +0100 @@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o snd-indigo-objs := indigo.o snd-indigoio-objs := indigoio.o snd-indigodj-objs := indigodj.o +snd-indigoiox-objs := indigoiox.o +snd-indigodjx-objs := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o @@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o obj-$(CONFIG_SND_INDIGO) += snd-indigo.o obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o +obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o +obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o --- alsa-driver-1.0.19__orig/alsa-kernel/pci/Kconfig 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/Kconfig 2009-03-15 17:14:39.000000000 +0100 @@ -400,6 +400,26 @@ config SND_INDIGODJ To compile this driver as a module, choose M here: the module will be called snd-indigodj +config SND_INDIGOIOX + tristate "(Echoaudio) Indigo IOx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo IOx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigoiox + +config SND_INDIGODJX + tristate "(Echoaudio) Indigo DJx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo DJx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigodjx + config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" select FW_LOADER --- alsa-driver-1.0.19__orig/pci/echoaudio/Makefile 2009-01-19 12:17:17.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/Makefile 2009-03-15 17:14:39.000000000 +0100 @@ -11,7 +11,8 @@ include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf clean-files := echoaudio.c darla20.c darla24.c echo3g.c gina20.c gina24.c \ - indigo.c indigodj.c indigoio.c layla20.c layla24.c mia.c mona.c + indigo.c indigodj.c indigoio.c indigodjx.c indigoiox.c \ + layla20.c layla24.c mia.c mona.c include $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/Makefile @@ -26,6 +27,8 @@ gina24.c: gina24.patch $(SND_TOPDIR)/als indigo.c: indigo.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigo.c echoaudio.c indigodj.c: indigodj.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigodj.c echoaudio.c indigoio.c: indigoio.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigoio.c echoaudio.c +indigodjx.c: indigodjx.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigodjx.c echoaudio.c +indigoiox.c: indigoiox.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigoiox.c echoaudio.c layla20.c: layla20.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/layla20.c echoaudio.c layla24.c: layla24.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/layla24.c echoaudio.c mia.c: mia.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/mia.c echoaudio.c --- alsa-driver-1.0.19__orig/pci/echoaudio/indigoiox.patch 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigoiox.patch 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,18 @@ +--- ../../alsa-kernel/pci/echoaudio/indigoiox.c 2009-03-01 15:26:34.000000000 +0100 ++++ indigoiox.c 2009-03-01 15:32:56.000000000 +0100 +@@ -1,3 +1,9 @@ ++#include "adriver.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) ++#define pci_device(chip) pci_name(chip->pci) ++#endif ++ + /* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> +@@ -107,3 +113,5 @@ static struct snd_pcm_hardware pcm_hardw + #include "echoaudio_dsp.c" + #include "echoaudio.c" + ++ ++EXPORT_NO_SYMBOLS; --- alsa-driver-1.0.19__orig/pci/echoaudio/indigodjx.patch 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigodjx.patch 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,18 @@ +--- ../../alsa-kernel/pci/echoaudio/indigodjx.c 2009-03-01 15:26:27.000000000 +0100 ++++ indigodjx.c 2009-03-01 19:02:55.000000000 +0100 +@@ -1,3 +1,9 @@ ++#include "adriver.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) ++#define pci_device(chip) pci_name(chip->pci) ++#endif ++ + /* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> +@@ -105,3 +111,5 @@ static struct snd_pcm_hardware pcm_hardw + #include "indigo_express_dsp.c" + #include "echoaudio_dsp.c" + #include "echoaudio.c" ++ ++EXPORT_NO_SYMBOLS; --- alsa-driver-1.0.19__orig/pci/echoaudio/indigo_express_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigo_express_dsp.c 2009-03-15 17:21:56.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigo_express_dsp.c" --- alsa-driver-1.0.19__orig/pci/echoaudio/indigoiox_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigoiox_dsp.c 2009-03-15 17:19:36.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigoiox_dsp.c" --- alsa-driver-1.0.19__orig/pci/echoaudio/indigodjx_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigodjx_dsp.c 2009-03-15 17:19:18.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigodjx_dsp.c" -- Giuliano.

2 1

[alsa-devel] When I got sound back by "kernel: hda-intel: Invalid position buffer, using LPIB read method instead."(random fallover)
by Emmanuel Chanel 16 Mar '09

16 Mar '09

When I got sound back by "kernel: hda-intel: Invalid position buffer, using LPIB read method instead."(random fallover) It doesn't mean succeed. Sometimes sound get back and sometimes not, at random.

2 1

[alsa-devel] no sound from headphones with STAC9205
by Thomas Folz-Donahue 16 Mar '09

16 Mar '09

I'm using a Gateway T-1616, which is an RS690 laptop from last year. It uses a STAC9205 codec. Somewhere between 1.0.16 and i suppose as early as 1.0.17, there was a regression wherein when I plug in my headphones, they are correctly detected and activated and the speakers are deactivated but then the headphones turn off. None of the existing model flags work. alsa-info.sh reports this: http://www.alsa-project.org/db/?f=d22d464730ca3ebd485ac0911eebb9eb60986233 and also, I ran alsa-info.sh with everything working: http://www.alsa-project.org/db/?f=b9c86006d49c0ff9ce9df02154db032ae2217786 ~thomas -- cartwheel ^(^_^)> v(.-.)v <(^_^)^

2 1

[alsa-devel] [PATCH 0/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

Hi, I found twl4030 build error. It appears because of changes of following commit: ASoC: make ops a pointer in 'struct snd_soc_dai' --- Joonyoung Shim (1): ASoC: twl4030 - Fix build error sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) -- - Joonyoung Shim

1 0

[alsa-devel] [PATCH 0/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

Hi -- - Joonyoung Shim

1 1

[alsa-devel] broonie/sound-2.6.git rebase?
by Daniel Mack 15 Mar '09

15 Mar '09

Hi Mark, Could you rebase your broonie/sound-2.6.git to Linus' master? When trying to merge them, I currently get: _From git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 * branch pxa-ssp -> FETCH_HEAD Auto-merging arch/arm/mach-shark/include/mach/io.h CONFLICT (content): Merge conflict in arch/arm/mach-shark/include/mach/io.h Auto-merging sound/soc/codecs/cs4270.c Auto-merging sound/soc/codecs/tlv320aic26.c Auto-merging sound/soc/codecs/wm8400.c CONFLICT (add/add): Merge conflict in sound/soc/codecs/wm8400.c Auto-merging sound/soc/codecs/wm8580.c CONFLICT (content): Merge conflict in sound/soc/codecs/wm8580.c Auto-merging sound/soc/pxa/pxa-ssp.c Auto-merging sound/soc/pxa/pxa2xx-ac97.c Auto-merging sound/soc/s3c24xx/s3c24xx-i2s.c Automatic merge failed; fix conflicts and then commit the result. Or is there any mystery about git I still haven't fully understood? Thanks, Daniel

2 2

[alsa-devel] [PATCH 2/2] Echoaudio: Add support for Indigo express cards
by Giuliano Pochini 15 Mar '09

15 Mar '09

This patch adds to alsa-firmware the firmware needed by the IndigoIOx and IndigoDJx cards. It's too long (134K) to post here inline. Please download the patch here: Signed-off-by Giuliano Pochini <pochini(a)shiny.it> http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch or http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch.bz2 -- Giuliano.

1 0

[alsa-devel] [PATCH 2/2] Echoaudio: remove line-out volume from vmixer cards
by Giuliano Pochini 15 Mar '09

15 Mar '09

With this patch the drivers do not set the vmixer volume anymore at startup because it is actually the output volume of the voices and ALSA mandates that the volume must be 0 by default. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodj_dsp.c 2009-03-15 17:24:35.000000000 +0100 @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: vchannels 0-3 and - vchannels 4-7 are routed to real channels 0-4 */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 2, 2, 0); - set_vmixer_gain(chip, 3, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigo_dsp.c 2009-03-15 17:24:14.000000000 +0100 @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoio_dsp.c 2009-03-15 17:24:00.000000000 +0100 @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/mia_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/mia_dsp.c 2009-03-15 17:23:37.000000000 +0100 @@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip))) return err; - /* Default routing of the virtual channels: vchannels 0-3 go to analog - outputs and vchannels 4-7 go to S/PDIF outputs */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 2, 4, 0); - set_vmixer_gain(chip, 3, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } -- Giuliano.

1 0

[alsa-devel] [RFC/PATCH] pxa2xx_ac97 and gpio reset line
by Robert Jarzmik 15 Mar '09

15 Mar '09

Hi Mark, I can't find out if somebody had submitted that kind of patch before. This is a starting point of discussion. I'm not fully convinced by my own code, and so I post it here for advice. What I'd really like is to be able to choose the PXA AC97 gpio reset (GPIO95 or GPIO113) through platform_data, but I couldn't find a way of doing it. I hope you'll have some kind of idea. Cheers. -- Robert

2 12

[alsa-devel] PXA SSP - work in progress
by Mark Brown 13 Mar '09

13 Mar '09

I worked a bit on the PXA SSP code last night and was able to come up with a configuration which uses non-network mode for I2S and works well on the Zylonite. I'll post the current series I have in a followup to this, if you could take a look that'd be great - I haven't yet worked through all the testing I'd like to do. Unfortunately it's going to have broken Daniel's configuration since I inverted the sense of LRCLK as the chip seemed not to generate an LRCLK with a non-zero frame delay; I need to check to see if this is just something I've overlooked. Hopefully Daniel's system should just have inverted the left and right channels. Having worked through non-network mode my feeling is that we should be able to come up with something that can figure out the extra clock cycles needed for Daniel's configuration with less of a special case. Non-network mode does seem like a better default than network mode because it avoids needing to look at the TDM configuration unless you want to use that.

2 4

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 13 Mar '09

13 Mar '09

The following changes since commit 2a9f0ba7a976bc2b1bcf9156c1e57ffbc8f8fb64: Mark Brown (1): Merge branch 's3c-iis-header' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Lopez Cruz, Misael (1): ASoC: Move headset jack registration to device initialization for SDP3430 Mark Brown (1): ASoC: Move WM8580 to normal I2C device probe Philipp Zabel (1): ASoC: Replace remaining uses of snd_soc_cnew with snd_soc_add_controls. sound/soc/codecs/cs4270.c | 23 +-- sound/soc/codecs/tlv320aic26.c | 11 +- sound/soc/codecs/wm8400.c | 12 +- sound/soc/codecs/wm8580.c | 326 +++++++++++++++++------------------ sound/soc/codecs/wm8580.h | 5 - sound/soc/omap/n810.c | 12 +- sound/soc/omap/sdp3430.c | 74 +++++---- sound/soc/pxa/corgi.c | 12 +- sound/soc/pxa/palm27x.c | 13 +- sound/soc/pxa/poodle.c | 12 +- sound/soc/pxa/spitz.c | 12 +- sound/soc/pxa/tosa.c | 12 +- sound/soc/s3c24xx/neo1973_wm8753.c | 13 +- 13 files changed, 243 insertions(+), 294 deletions(-)

2 1

[alsa-devel] [PATCH] SoC: Move headset jack registration to device initialization for SDP3430
by Lopez Cruz, Misael 13 Mar '09

13 Mar '09

Move headset jack registration to the codec/machine specific initialization. Having the jack registration in machine init causes that the jack device gets initialized but not registered since the sound card is registered before the jack. Moving jack registration to device initialization will register the jack device along with all other devices associated to the card when the card is registed. As a consequence of jack device registered properly, the jack is detected as an input device. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 74 ++++++++++++++++++++++++--------------------- 1 files changed, 39 insertions(+), 35 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 715c648..0a41de6 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -39,6 +39,8 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +static struct snd_soc_card snd_soc_sdp3430; + static int sdp3430_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -82,6 +84,27 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + /* SDP3430 machine DAPM */ static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), @@ -141,30 +164,25 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; - return ret; -} + /* Headset jack detection */ + ret = snd_soc_jack_new(&snd_soc_sdp3430, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; -/* Headset jack */ -static struct snd_soc_jack hs_jack; + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Headset Jack", - .mask = SND_JACK_HEADSET, - }, -}; + ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); -/* Headset jack detection gpios */ -static struct snd_soc_jack_gpio hs_jack_gpios[] = { - { - .gpio = (OMAP_MAX_GPIO_LINES + 2), - .name = "hsdet-gpio", - .report = SND_JACK_HEADSET, - .debounce_time = 200, - }, -}; + return ret; +} /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { @@ -216,21 +234,7 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - /* Headset jack detection */ - ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", - SND_JACK_HEADSET, &hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - if (ret) - return ret; - - ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return ret; + return 0; err1: printk(KERN_ERR "Unable to add platform device\n"); -- 1.5.4.3

2 1

[alsa-devel] Audio problems with HP DV7-1025nr
by Jonathan Scruggs 13 Mar '09

13 Mar '09

Hi, I have tried many things to try to get my sound card to work, but no luck. At first with the when I started alsamixer, it would say that I have a generic sound card. My laptop has two audio interfaces: a nVidia HDMI and an IDT High Definition Codec. I read this mailing list and saw that many people had issues with the HP's version of the IDT sound card. Then someone added patches to fix an issue with DV4. The fixes were in the latest snapshots, so I did some searching and pulled the alsa sources from here: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git since I am using the 2.6.28 kernel. I went through the code and confirmed that it was updated to what I saw in the patches and the patches were not compatible with the kernel source I have, hence I was looking for something more compatible. More specifically, the kernel I have is "2.6.28-gentoo-r2 #5 SMP Wed Feb 25 15:37:22 GMT 2009 x86_64 Intel(R) Core(TM)2 Duo CPU P8400 @ 2.26GHz Genuine Intel GNU/Linux". With a little bit of work, I merged in the new alsa source with that kernel and recompiled. Things are detected a bit better, like now alsamixer is saying that I have the chip "Nvidia MCP78 HDMI" instead of a Generic one. However, this is not the chip I want. I want to output to my speakers and headphone ports. I tried setting the option "model=hp-dv5" and then "model=hp-dv7" but that does nothing. I have absolutely no sound at all out of my speakers or headphone port. The drivers for this soundcard are on the HP website if you need them. Also, I tried removing the nvidia option from the list in the kernel configuration and only have IDT selected, but it would say unknow nvidia chip when trying to load the module on boot. I have the output of lspci -nv attached to this email. I hope you can give me a clue as to what to do next. This is what I have in my in my /etc/modprobe.d/alsa file: # Alsa kernel modules' configuration file. # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=1 alias snd-card-0 snd-hda-intel options snd-hda-intel model=hp-dv7 enable=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 I have a program that analyzed my audio in Windows. This is the output from that: Sound, video and game controllers || NVIDIA HDMI Audio Top Property Value NVIDIA HDMI Audio Device ID HDAUDIO\FUNC_01VEN_10DEDEV_0006SUBSYS_10DE0101REV_1000\43451138D00201 Status 0x0180200a Started Problem 0x00000000 (0) Service NVHDA Capabilities 0x00000000 Config Flags 0x00000000 Class MEDIA Manufacturer NVIDIA Hardware IDs HDAUDIO\FUNC_01VEN_10DEDEV_0006SUBSYS_10DE0101REV_1000 HDAUDIO\FUNC_01VEN_10DEDEV_0006SUBSYS_10DE0101 Compatible IDs HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293EVEN_10DEDEV_0006REV_1000 HDAUDIO\FUNC_01CTLR_VEN_8086VEN_10DEDEV_0006REV_1000 HDAUDIO\FUNC_01VEN_10DEDEV_0006REV_1000 HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293EVEN_10DEDEV_0006 HDAUDIO\FUNC_01CTLR_VEN_8086VEN_10DEDEV_0006 HDAUDIO\FUNC_01VEN_10DEDEV_0006 HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293EVEN_10DE HDAUDIO\FUNC_01CTLR_VEN_8086VEN_10DE HDAUDIO\FUNC_01VEN_10DE HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293E HDAUDIO\FUNC_01CTLR_VEN_8086 HDAUDIO\FUNC_01GFVEN_10DEDEV_0006SUBSYS_10DE0101REV_1000 HDAUDIO\FUNC_01 Class GUID {4d36e96c-e325-11ce-bfc1-08002be10318} Location Internal High Definition Audio Bus Bus number 0x00000000 Enumerator name HDAUDIO Description NVIDIA HDMI Audio Driver {4d36e96c-e325-11ce-bfc1-08002be10318}\0000 Physical Object Name \Device\0000008b UI number 0x00000002 Bustype GUID {41203534-2037-3144-2042-422044362041} Legacy bus type 0x00000005 Device Type 0x0000001d Install State 0x00000000 Security 01 00 04 90 00 00 00 00 00 00 00 00 00 00 00 00 14 00 00 00 02 00 5C 00 04 00 00 00 00 00 14 00 00 00 00 10 01 01 00 00 00 00 00 05 12 00 00 00 00 00 18 00 00 00 00 E0 01 02 00 00 00 00 00 05 20 00 00 00 20 02 00 00 00 00 14 00 00 00 00 E0 01 01 00 00 00 00 00 01 00 00 00 00 00 00 14 00 00 00 00 E0 01 01 00 00 00 00 00 05 0C 00 00 00 Security (SDS form) D:P(A;;GA;;;SY)(A;;GXGWGR;;;BA)(A;;GXGWGR;;;WD)(A;;GXGWGR;;;RC) Device Address 0x00000201 Sound, video and game controllers || IDT High Definition Audio CODEC Top Property Value IDT High Definition Audio CODEC Device ID HDAUDIO\FUNC_01VEN_111DDEV_76B2SUBSYS_103C30F4REV_1003\43451138D00001 Status 0x0180200a Started Problem 0x00000000 (0) Service STHDA Capabilities 0x00000000 Config Flags 0x00000000 Class MEDIA Manufacturer IDT Hardware IDs HDAUDIO\FUNC_01VEN_111DDEV_76B2SUBSYS_103C30F4REV_1003 HDAUDIO\FUNC_01VEN_111DDEV_76B2SUBSYS_103C30F4 Compatible IDs HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293EVEN_111DDEV_76B2REV_1003 HDAUDIO\FUNC_01CTLR_VEN_8086VEN_111DDEV_76B2REV_1003 HDAUDIO\FUNC_01VEN_111DDEV_76B2REV_1003 HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293EVEN_111DDEV_76B2 HDAUDIO\FUNC_01CTLR_VEN_8086VEN_111DDEV_76B2 HDAUDIO\FUNC_01VEN_111DDEV_76B2 HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293EVEN_111D HDAUDIO\FUNC_01CTLR_VEN_8086VEN_111D HDAUDIO\FUNC_01VEN_111D HDAUDIO\FUNC_01CTLR_VEN_8086CTLR_DEV_293E HDAUDIO\FUNC_01CTLR_VEN_8086 HDAUDIO\FUNC_01GFVEN_111DDEV_76B2SUBSYS_103C30F4REV_1003 HDAUDIO\FUNC_01 Class GUID {4d36e96c-e325-11ce-bfc1-08002be10318} Location Internal High Definition Audio Bus Bus number 0x00000000 Enumerator name HDAUDIO Description IDT High Definition Audio CODEC Friendly name IDT High Definition Audio CODEC Driver {4d36e96c-e325-11ce-bfc1-08002be10318}\0007 Physical Object Name \Device\00000089 UI number 0x00000000 Bustype GUID {41203534-2037-3144-2042-422044362041} Legacy bus type 0x00000005 Device Type 0x0000001d Install State 0x00000000 Security 01 00 04 90 00 00 00 00 00 00 00 00 00 00 00 00 14 00 00 00 02 00 5C 00 04 00 00 00 00 00 14 00 00 00 00 10 01 01 00 00 00 00 00 05 12 00 00 00 00 00 18 00 00 00 00 E0 01 02 00 00 00 00 00 05 20 00 00 00 20 02 00 00 00 00 14 00 00 00 00 E0 01 01 00 00 00 00 00 01 00 00 00 00 00 00 14 00 00 00 00 E0 01 01 00 00 00 00 00 05 0C 00 00 00 Security (SDS form) D:P(A;;GA;;;SY)(A;;GXGWGR;;;BA)(A;;GXGWGR;;;WD)(A;;GXGWGR;;;RC) Device Address 0x00000001 Thanks, Jon

2 1

[alsa-devel] multiple USB soundcards crashes, please help
by Giovanni Maruzzelli 13 Mar '09

13 Mar '09

Dear ALSA developers, my application needs to use multiple USB soundcards (based on standard cm-108 chipset) for I/O. Seems to me that with recent kernels (after 2.6.21) this is almost impossible: -if you use uhci_hcd it crashes with 4 running soundcards (cannot submit datapipe: not enough bandwidth) -if you use ehci_hcd it crashes way before (device resets) With older kernels, at least uhci_hcd seems to work. This maybe is related to the reworking of the bandwidth management into the uhci_hcd introduced in 2.6.21. I'm available to and I would be glad to help you ALSA developers with all the information possible: just tell me wich infos you need and I'll gather all the infos for you. You can test the problem with: arecord -Dplughw:0 | aplay plughw:0 & arecord -Dplughw:1 | aplay plughw:1 & arecord -Dplughw:2 | aplay plughw:2 & arecord -Dplughw:3 | aplay plughw:3 &

2 2

[alsa-devel] How can PCI resource allocation fail?
by Christian P. Schmidt 13 Mar '09

13 Mar '09

Hi all, I'm toying with the alsa-driver for the Creative Labs X-Fi. I'm working on the PCI Express version of this card, though, which is the same as the PCI version plus a PCIe to PCI bridge but has a different ID: +-1c.2-[0000:03-04]----00.0-[0000:04]----00.0 1102:0009 03:00.0 PCI bridge: Creative Labs Device 7006 04:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Now, after adding the ID, the driver fails: [46833.487579] CA0106 0000:04:00.0: PCI INT A -> GSI 18 (level, low) -> IRQ 18 [46833.487586] CA0106 0000:04:00.0: PCI INT A disabled [46833.487588] cannot allocate the port [46833.487594] CA0106: probe of 0000:04:00.0 failed with error -16 The actual error is occurring in chip->port = pci_resource_start(pci, 0); chip->res_port = request_region(chip->port, pci_resource_len(pci,0), "snd_ca0106"); if (!chip->res_port) { snd_ca0106_free(chip); printk(KERN_ERR "cannot allocate the port\n"); return -EBUSY; } This from /proc/iomem: f9f00000-f9ffffff : PCI Bus 0000:03 f9f00000-f9ffffff : PCI Bus 0000:04 f9ffc000-f9ffffff : 0000:04:00.0 Some additional debug output tells me that the code is actually trying to allocate 16k starting from f9ffc000. What could be potential reasons for the failure? Do I have to do something with the bridge before I can try to request the region? Regards, Christian PS: lcpci -vv output afterwards, can't see anything special here. 03:00.0 PCI bridge: Creative Labs Device 7006 (prog-if 00 [Normal decode]) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Bus: primary=03, secondary=04, subordinate=04, sec-latency=64 Memory behind bridge: f9f00000-f9ffffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity+ SERR+ NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ Capabilities: [60] MSI: Mask- 64bit+ Count=1/16 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [80] Subsystem: Creative Labs Device 0010 Capabilities: [90] Express (v1) PCI/PCI-X Bridge, MSI 00 DevCap: MaxPayload 512 bytes, PhantFunc 0, Latency L0s <4us, L1 <64us ExtTag- AttnBtn- AttnInd- PwrInd- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ BrConfRtry- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- UncorrErr+ FatalErr- UnsuppReq+ AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <512ns, L1 <16us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq+ ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES- TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- AERCap: First Error Pointer: 14, GenCap+ CGenEn- ChkCap+ ChkEn- 04:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin A routed to IRQ 18 Region 0: Memory at f9ffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel modules: snd-ca0106

5 4

[alsa-devel] [PATCH] ALSA: drop outdated and broken sa11xx-uda1341 driver
by Dmitry Artamonow 13 Mar '09

13 Mar '09

It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got merged into mainline. Since there's no way to use it on any of supported machines (iPaq h3100 or h3600), better drop it for now. It can be reimplemented later using ASoC infrastructure (there's already a driver for uda1341 codec in mainline, so only CPU and machine parts need to be written). Signed-off-by: Dmitry Artamonow <mad_soft(a)inbox.ru> Cc: Russell King <linux(a)arm.linux.org.uk> --- include/sound/uda1341.h | 126 ------ sound/arm/Kconfig | 11 - sound/arm/Makefile | 3 - sound/arm/sa11xx-uda1341.c | 983 -------------------------------------------- sound/i2c/Makefile | 2 - sound/i2c/l3/Makefile | 8 - sound/i2c/l3/uda1341.c | 935 ----------------------------------------- 7 files changed, 0 insertions(+), 2068 deletions(-) delete mode 100644 include/sound/uda1341.h delete mode 100644 sound/arm/sa11xx-uda1341.c delete mode 100644 sound/i2c/l3/Makefile delete mode 100644 sound/i2c/l3/uda1341.c diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h deleted file mode 100644 index 110d5dc..0000000 --- a/include/sound/uda1341.h +++ /dev/null @@ -1,126 +0,0 @@ -/* - * linux/include/linux/l3/uda1341.h - * - * Philips UDA1341 mixer device driver for ALSA - * - * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek(a)seznam.cz> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek Initial release - based on uda1341.h from OSS - * 2002-03-30 Tomas Kasparek Proc filesystem support, complete mixer and DSP - * features support - */ - -#define UDA1341_ALSA_NAME "snd-uda1341" - -/* - * Default rate set after inicialization - */ -#define AUDIO_RATE_DEFAULT 44100 - -/* - * UDA1341 L3 address and command types - */ -#define UDA1341_L3ADDR 5 -#define UDA1341_DATA0 (UDA1341_L3ADDR << 2 | 0) -#define UDA1341_DATA1 (UDA1341_L3ADDR << 2 | 1) -#define UDA1341_STATUS (UDA1341_L3ADDR << 2 | 2) - -enum uda1341_onoff { - OFF=0, - ON, -}; - -enum uda1341_format { - I2S=0, - LSB16, - LSB18, - LSB20, - MSB, - LSB16MSB, - LSB18MSB, - LSB20MSB, -}; - -enum uda1341_fs { - F512=0, - F384, - F256, - Funused, -}; - -enum uda1341_peak { - BEFORE=0, - AFTER, -}; - -enum uda1341_filter { - FLAT=0, - MIN, - MIN2, - MAX, -}; - -enum uda1341_mixer { - DOUBLE, - LINE, - MIC, - MIXER, -}; - -enum uda1341_deemp { - NONE, - D32, - D44, - D48, -}; - -enum uda1341_config { - CMD_READ_REG = 0, - CMD_RESET, - CMD_FS, - CMD_FORMAT, - CMD_OGAIN, - CMD_IGAIN, - CMD_DAC, - CMD_ADC, - CMD_VOLUME, - CMD_BASS, - CMD_TREBBLE, - CMD_PEAK, - CMD_DEEMP, - CMD_MUTE, - CMD_FILTER, - CMD_CH1, - CMD_CH2, - CMD_MIC, - CMD_MIXER, - CMD_AGC, - CMD_IG, - CMD_AGC_TIME, - CMD_AGC_LEVEL, -#ifdef CONFIG_PM - CMD_SUSPEND, - CMD_RESUME, -#endif - CMD_LAST, -}; - -enum write_through { - //used in update_bits (write_cfg) to avoid l3_write - just update local copy of regs. - REGS_ONLY=0, - //update local regs and write value to uda1341 - do l3_write - FLUSH, -}; - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clnt); - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de4..885683a 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -11,17 +11,6 @@ menuconfig SND_ARM if SND_ARM -config SND_SA11XX_UDA1341 - tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && L3 - select SND_PCM - help - Say Y here if you have a Compaq iPaq H3x00 handheld computer - and want to use its Philips UDA 1341 audio chip. - - To compile this driver as a module, choose M here: the module - will be called snd-sa11xx-uda1341. - config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de1..5a549ed 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -2,9 +2,6 @@ # Makefile for ALSA # -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o -snd-sa11xx-uda1341-objs := sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o devdma.o diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c deleted file mode 100644 index 1dcd51d..0000000 --- a/sound/arm/sa11xx-uda1341.c +++ /dev/null @@ -1,983 +0,0 @@ -/* - * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard - * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek(a)seznam.cz> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS - * 2002-03-20 Tomas Kasparek playback over ALSA is working - * 2002-03-28 Tomas Kasparek playback over OSS emulation is working - * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA) - * 2002-03-29 Tomas Kasparek capture is working (OSS emulation) - * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates) - * 2003-02-14 Brian Avery fixed full duplex mode, other updates - * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL) - * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel - * working suspend and resume - * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again - * merged HAL layer (patches from Brian) - */ - -/*************************************************************************************************** -* -* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai -* available in the Alsa doc section on the website -* -* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100. -* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated -* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it. -* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the -* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which -* is a mem loc that always decodes to 0's w/ no off chip access. -* -* Some alsa terminology: -* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes -* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte -* buffer and 4 periods in the runtime structure this means we'll get an int every 256 -* bytes or 4 times per buffer. -* A number of the sizes are in frames rather than bytes, use frames_to_bytes and -* bytes_to_frames to convert. The easiest way to tell the units is to look at the -* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t -* -* Notes about the pointer fxn: -* The pointer fxn needs to return the offset into the dma buffer in frames. -* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts. -* -* Notes about pause/resume -* Implementing this would be complicated so it's skipped. The problem case is: -* A full duplex connection is going, then play is paused. At this point you need to start xmitting -* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd -* need to save off the dma info, and restore it properly on a resume. Yeach! -* -* Notes about transfer methods: -* The async write calls fail. I probably need to implement something else to support them? -* -***************************************************************************************************/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/err.h> -#include <linux/platform_device.h> -#include <linux/errno.h> -#include <linux/ioctl.h> -#include <linux/delay.h> -#include <linux/slab.h> - -#ifdef CONFIG_PM -#include <linux/pm.h> -#endif - -#include <mach/hardware.h> -#include <mach/h3600.h> -#include <asm/mach-types.h> -#include <asm/dma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/initval.h> - -#include <linux/l3/l3.h> - -#undef DEBUG_MODE -#undef DEBUG_FUNCTION_NAMES -#include <sound/uda1341.h> - -/* - * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? - * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this - * module for Familiar 0.6.1 - */ - -/* {{{ Type definitions */ - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); - -static char *id; /* ID for this card */ - -module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); - -struct audio_stream { - char *id; /* identification string */ - int stream_id; /* numeric identification */ - dma_device_t dma_dev; /* device identifier for DMA */ -#ifdef HH_VERSION - dmach_t dmach; /* dma channel identification */ -#else - dma_regs_t *dma_regs; /* points to our DMA registers */ -#endif - unsigned int active:1; /* we are using this stream for transfer now */ - int period; /* current transfer period */ - int periods; /* current count of periods registerd in the DMA engine */ - int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ - unsigned int old_offset; - spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ - struct snd_pcm_substream *stream; -}; - -struct sa11xx_uda1341 { - struct snd_card *card; - struct l3_client *uda1341; - struct snd_pcm *pcm; - long samplerate; - struct audio_stream s[2]; /* playback & capture */ -}; - -static unsigned int rates[] = { - 8000, 10666, 10985, 14647, - 16000, 21970, 22050, 24000, - 29400, 32000, 44100, 48000, -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct platform_device *device; - -/* }}} */ - -/* {{{ Clock and sample rate stuff */ - -/* - * Stop-gap solution until rest of hh.org HAL stuff is merged. - */ -#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) -#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) - -#ifdef CONFIG_SA1100_H3XXX -#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) -#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) -#else -#error This driver could serve H3x00 handhelds only! -#endif - -static void sa11xx_uda1341_set_audio_clock(long val) -{ - switch (val) { - case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ - GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - - case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ - GPSR = GPIO_H3600_CLK_SET0; - GPCR = GPIO_H3600_CLK_SET1; - break; - - case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ - GPCR = GPIO_H3600_CLK_SET0; - GPSR = GPIO_H3600_CLK_SET1; - break; - - case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ - GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - } -} - -static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate) -{ - int clk_div = 0; - int clk=0; - - /* We don't want to mess with clocks when frames are in flight */ - Ser4SSCR0 &= ~SSCR0_SSE; - /* wait for any frame to complete */ - udelay(125); - - /* - * We have the following clock sources: - * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz - * Those can be divided either by 256, 384 or 512. - * This makes up 12 combinations for the following samplerates... - */ - if (rate >= 48000) - rate = 48000; - else if (rate >= 44100) - rate = 44100; - else if (rate >= 32000) - rate = 32000; - else if (rate >= 29400) - rate = 29400; - else if (rate >= 24000) - rate = 24000; - else if (rate >= 22050) - rate = 22050; - else if (rate >= 21970) - rate = 21970; - else if (rate >= 16000) - rate = 16000; - else if (rate >= 14647) - rate = 14647; - else if (rate >= 10985) - rate = 10985; - else if (rate >= 10666) - rate = 10666; - else - rate = 8000; - - /* Set the external clock generator */ - - sa11xx_uda1341_set_audio_clock(rate); - - /* Select the clock divisor */ - switch (rate) { - case 8000: - case 10985: - case 22050: - case 24000: - clk = F512; - clk_div = SSCR0_SerClkDiv(16); - break; - case 16000: - case 21970: - case 44100: - case 48000: - clk = F256; - clk_div = SSCR0_SerClkDiv(8); - break; - case 10666: - case 14647: - case 29400: - case 32000: - clk = F384; - clk_div = SSCR0_SerClkDiv(12); - break; - } - - /* FMT setting should be moved away when other FMTs are added (FIXME) */ - l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); - - l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); - Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; - sa11xx_uda1341->samplerate = rate; -} - -/* }}} */ - -/* {{{ HW init and shutdown */ - -static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - unsigned long flags; - - /* Setup DMA stuff */ - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; - - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; - - /* Initialize the UDA1341 internal state */ - - /* Setup the uarts */ - local_irq_save(flags); - GAFR |= (GPIO_SSP_CLK); - GPDR &= ~(GPIO_SSP_CLK); - Ser4SSCR0 = 0; - Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); - Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; - Ser4SSCR0 |= SSCR0_SSE; - local_irq_restore(flags); - - /* Enable the audio power */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* Wait for the UDA1341 to wake up */ - mdelay(1); //FIXME - was removed by Perex - Why? - - /* Initialize the UDA1341 internal state */ - l3_open(sa11xx_uda1341->uda1341); - - /* external clock configuration (after l3_open - regs must be initialized */ - sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); - - /* Wait for the UDA1341 to wake up */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - mdelay(1); - - /* make the left and right channels unswapped (flip the WS latch) */ - Ser4SSDR = 0; - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - /* mute on */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* disable the audio power and all signals leading to the audio chip */ - l3_close(sa11xx_uda1341->uda1341); - Ser4SSCR0 = 0; - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - - /* power off and mute off */ - /* FIXME - is muting off necesary??? */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -/* }}} */ - -/* {{{ DMA staff */ - -/* - * these are the address and sizes used to fill the xmit buffer - * so we can get a clock in record only mode - */ -#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS -#define FORCE_CLOCK_SIZE 4096 // was 2048 - -// FIXME Why this value exactly - wrote comment -#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ - -#ifdef HH_VERSION - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int)) -{ - int ret; - - ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); - if (ret < 0) { - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; - } - sa1100_dma_set_callback(s->dmach, callback); - return 0; -} - -static inline void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dmach); - s->dmach = -1; -} - -#else - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *)) -{ - int ret; - - ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); - if (ret < 0) - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; -} - -static void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dma_regs); - s->dma_regs = 0; -} - -#endif - -static u_int audio_get_dma_pos(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int offset; - unsigned long flags; - dma_addr_t addr; - - // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel - spin_lock_irqsave(&s->dma_lock, flags); -#ifdef HH_VERSION - sa1100_dma_get_current(s->dmach, NULL, &addr); -#else - addr = sa1100_get_dma_pos((s)->dma_regs); -#endif - offset = addr - runtime->dma_addr; - spin_unlock_irqrestore(&s->dma_lock, flags); - - offset = bytes_to_frames(runtime,offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -/* - * this stops the dma and clears the dma ptrs - */ -static void audio_stop_dma(struct audio_stream *s) -{ - unsigned long flags; - - spin_lock_irqsave(&s->dma_lock, flags); - s->active = 0; - s->period = 0; - /* this stops the dma channel and clears the buffer ptrs */ -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - sa1100_clear_dma(s->dma_regs); -#endif - spin_unlock_irqrestore(&s->dma_lock, flags); -} - -static void audio_process_dma(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime; - unsigned int dma_size; - unsigned int offset; - int ret; - - /* we are requested to process synchronization DMA transfer */ - if (s->tx_spin) { - if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK)) - return; - /* fill the xmit dma buffers and return */ -#ifdef HH_VERSION - sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); -#else - while (1) { - ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); - if (ret) - return; - } -#endif - return; - } - - /* must be set here - only valid for running streams, not for forced_clock dma fills */ - runtime = substream->runtime; - while (s->active && s->periods < runtime->periods) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - if (s->old_offset) { - /* a little trick, we need resume from old position */ - offset = frames_to_bytes(runtime, s->old_offset - 1); - s->old_offset = 0; - s->periods = 0; - s->period = offset / dma_size; - offset %= dma_size; - dma_size = dma_size - offset; - if (!dma_size) - continue; /* special case */ - } else { - offset = dma_size * s->period; - snd_BUG_ON(dma_size > DMA_BUF_SIZE); - } -#ifdef HH_VERSION - ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size); - if (ret) - return; //FIXME -#else - ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size); - if (ret) { - printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret); - return; - } -#endif - - s->period++; - s->period %= runtime->periods; - s->periods++; - } -} - -#ifdef HH_VERSION -static void audio_dma_callback(void *data, int size) -#else -static void audio_dma_callback(void *data) -#endif -{ - struct audio_stream *s = data; - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - spin_lock(&s->dma_lock); - if (!s->tx_spin && s->periods > 0) - s->periods--; - audio_process_dma(s); - spin_unlock(&s->dma_lock); -} - -/* }}} */ - -/* {{{ PCM setting */ - -/* {{{ trigger & timer */ - -static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - int stream_id = substream->pstr->stream; - struct audio_stream *s = &chip->s[stream_id]; - struct audio_stream *s1 = &chip->s[stream_id ^ 1]; - int err = 0; - - /* note local interrupts are already disabled in the midlevel code */ - spin_lock(&s->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* now we need to make sure a record only stream has a clock */ - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s1->tx_spin = 1; - audio_process_dma(s1); - } - /* this case is when you were recording then you turn on a - * playback stream so we stop (also clears it) the dma first, - * clear the sync flag and then we let it turned on - */ - else { - s->tx_spin = 0; - } - - /* requested stream startup */ - s->active = 1; - audio_process_dma(s); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - audio_stop_dma(s); - - /* - * now we need to make sure a record only stream has a clock - * so if we're stopping a playback with an active capture - * we need to turn the 0 fill dma on for the xmit side - */ - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s->tx_spin = 1; - audio_process_dma(s); - } - /* - * we killed a capture only stream, so we should also kill - * the zero fill transmit - */ - else { - if (s1->tx_spin) { - s1->tx_spin = 0; - audio_stop_dma(s1); - } - } - - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - s->active = 0; -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - s->periods = 0; - break; - case SNDRV_PCM_TRIGGER_RESUME: - s->active = 1; - s->tx_spin = 0; - audio_process_dma(s); - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->active = 0; - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) { - if (s1->active) { - s->tx_spin = 1; - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - audio_process_dma(s); - } - } else { - if (s1->tx_spin) { - s1->tx_spin = 0; -#ifdef HH_VERSION - sa1100_dma_flush_all(s1->dmach); -#else - //FIXME - DMA API -#endif - } - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - s->active = 1; - if (s->old_offset) { - s->tx_spin = 0; - audio_process_dma(s); - break; - } - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } -#ifdef HH_VERSION - sa1100_dma_resume(s->dmach); -#else - //FIXME - DMA API -#endif - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&s->dma_lock); - return err; -} - -static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audio_stream *s = &chip->s[substream->pstr->stream]; - - /* set requested samplerate */ - sa11xx_uda1341_set_samplerate(chip, runtime->rate); - - /* set requestd format when available */ - /* set FMT here !!! FIXME */ - - s->period = 0; - s->periods = 0; - - return 0; -} - -static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - return audio_get_dma_pos(&chip->s[substream->pstr->stream]); -} - -/* }}} */ - -static struct snd_pcm_hardware snd_sa11xx_uda1341_capture = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_sa11xx_uda1341_playback = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int stream_id = substream->pstr->stream; - int err; - - chip->s[stream_id].stream = substream; - - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = snd_sa11xx_uda1341_playback; - else - runtime->hw = snd_sa11xx_uda1341_capture; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) - return err; - - return 0; -} - -static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - - chip->s[substream->pstr->stream].stream = NULL; - return 0; -} - -/* {{{ HW params & free */ - -static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - -static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -/* }}} */ - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0) - return err; - - /* - * this sets up our initial buffers and sets the dma_type to isa. - * isa works but I'm not sure why (or if) it's the right choice - * this may be too large, trying it for now - */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, 64*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops); - pcm->private_data = sa11xx_uda1341; - pcm->info_flags = 0; - strcpy(pcm->name, "UDA1341 PCM"); - - sa11xx_uda1341_audio_init(sa11xx_uda1341); - - /* setup DMA controller */ - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback); - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback); - - sa11xx_uda1341->pcm = pcm; - - return 0; -} - -/* }}} */ - -/* {{{ module init & exit */ - -#ifdef CONFIG_PM - -static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr, - pm_message_t state) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); -#ifdef HH_VERSION - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - l3_command(chip->uda1341, CMD_SUSPEND, NULL); - sa11xx_uda1341_audio_shutdown(chip); - - return 0; -} - -static int snd_sa11xx_uda1341_resume(struct platform_device *devptr) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - sa11xx_uda1341_audio_init(chip); - l3_command(chip->uda1341, CMD_RESUME, NULL); -#ifdef HH_VERSION - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif /* COMFIG_PM */ - -void snd_sa11xx_uda1341_free(struct snd_card *card) -{ - struct sa11xx_uda1341 *chip = card->private_data; - - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); -} - -static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) -{ - int err; - struct snd_card *card; - struct sa11xx_uda1341 *chip; - - /* register the soundcard */ - card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341)); - if (card == NULL) - return -ENOMEM; - - chip = card->private_data; - spin_lock_init(&chip->s[0].dma_lock); - spin_lock_init(&chip->s[1].dma_lock); - - card->private_free = snd_sa11xx_uda1341_free; - chip->card = card; - chip->samplerate = AUDIO_RATE_DEFAULT; - - // mixer - if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341))) - goto nodev; - - // PCM - if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0) - goto nodev; - - strcpy(card->driver, "UDA1341"); - strcpy(card->shortname, "H3600 UDA1341TS"); - sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); - - snd_card_set_dev(card, &devptr->dev); - - if ((err = snd_card_register(card)) == 0) { - printk( KERN_INFO "iPAQ audio support initialized\n" ); - platform_set_drvdata(devptr, card); - return 0; - } - - nodev: - snd_card_free(card); - return err; -} - -static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr) -{ - snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; -} - -#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341" - -static struct platform_driver sa11xx_uda1341_driver = { - .probe = sa11xx_uda1341_probe, - .remove = __devexit_p(sa11xx_uda1341_remove), -#ifdef CONFIG_PM - .suspend = snd_sa11xx_uda1341_suspend, - .resume = snd_sa11xx_uda1341_resume, -#endif - .driver = { - .name = SA11XX_UDA1341_DRIVER, - }, -}; - -static int __init sa11xx_uda1341_init(void) -{ - int err; - - if (!machine_is_h3xxx()) - return -ENODEV; - if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) - return err; - device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (!IS_ERR(device)) { - if (platform_get_drvdata(device)) - return 0; - platform_device_unregister(device); - err = -ENODEV; - } else - err = PTR_ERR(device); - platform_driver_unregister(&sa11xx_uda1341_driver); - return err; -} - -static void __exit sa11xx_uda1341_exit(void) -{ - platform_device_unregister(device); - platform_driver_unregister(&sa11xx_uda1341_driver); -} - -module_init(sa11xx_uda1341_init); -module_exit(sa11xx_uda1341_exit); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 3797066..36879bf 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -obj-$(CONFIG_L3) += l3/ - obj-$(CONFIG_SND) += other/ # Toplevel Module Dependency diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile deleted file mode 100644 index 49455b8..0000000 --- a/sound/i2c/l3/Makefile +++ /dev/null @@ -1,8 +0,0 @@ -# -# Makefile for ALSA -# - -snd-uda1341-objs := uda1341.o - -# Module Dependency -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c deleted file mode 100644 index 9840eb4..0000000 --- a/sound/i2c/l3/uda1341.c +++ /dev/null @@ -1,935 +0,0 @@ -/* - * Philips UDA1341 mixer device driver - * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek(a)seznam.cz> - * - * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS - * 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble) - * 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP - * features support - * 2002-04-12 Tomas Kasparek proc interface update, code cleanup - * 2002-05-12 Tomas Kasparek another code cleanup - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/types.h> -#include <linux/slab.h> -#include <linux/errno.h> -#include <linux/ioctl.h> - -#include <asm/uaccess.h> - -#include <sound/core.h> -#include <sound/control.h> -#include <sound/initval.h> -#include <sound/info.h> - -#include <linux/l3/l3.h> - -#include <sound/uda1341.h> - -/* {{{ HW regs definition */ - -#define STAT0 0x00 -#define STAT1 0x80 -#define STAT_MASK 0x80 - -#define DATA0_0 0x00 -#define DATA0_1 0x40 -#define DATA0_2 0x80 -#define DATA_MASK 0xc0 - -#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2) -#define IS_DATA1(x) ((x) == data1) -#define IS_STATUS(x) ((x) == stat0 || (x) == stat1) -#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6) - -/* }}} */ - - -static const char *peak_names[] = { - "before", - "after", -}; - -static const char *filter_names[] = { - "flat", - "min", - "min", - "max", -}; - -static const char *mixer_names[] = { - "double differential", - "input channel 1 (line in)", - "input channel 2 (microphone)", - "digital mixer", -}; - -static const char *deemp_names[] = { - "none", - "32 kHz", - "44.1 kHz", - "48 kHz", -}; - -enum uda1341_regs_names { - stat0, - stat1, - data0_0, - data0_1, - data0_2, - data1, - ext0, - ext1, - ext2, - empty, - ext4, - ext5, - ext6, - uda1341_reg_last, -}; - -static const char *uda1341_reg_names[] = { - "stat 0 ", - "stat 1 ", - "data 00", - "data 01", - "data 02", - "data 1 ", - "ext 0", - "ext 1", - "ext 2", - "empty", - "ext 4", - "ext 5", - "ext 6", -}; - -static const int uda1341_enum_items[] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 2, //peak - before/after - 4, //deemp - none/32/44.1/48 - 0, - 4, //filter - flat/min/min/max - 0, 0, 0, - 4, //mixer - differ/line/mic/mixer - 0, 0, 0, 0, 0, -}; - -static const char ** uda1341_enum_names[] = { - NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, - peak_names, //peak - before/after - deemp_names, //deemp - none/32/44.1/48 - NULL, - filter_names, //filter - flat/min/min/max - NULL, NULL, NULL, - mixer_names, //mixer - differ/line/mic/mixer - NULL, NULL, NULL, NULL, NULL, -}; - -typedef int uda1341_cfg[CMD_LAST]; - -struct uda1341 { - int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val); - int (*read) (struct l3_client *uda1341, unsigned short reg); - unsigned char regs[uda1341_reg_last]; - int active; - spinlock_t reg_lock; - struct snd_card *card; - uda1341_cfg cfg; -#ifdef CONFIG_PM - unsigned char suspend_regs[uda1341_reg_last]; - uda1341_cfg suspend_cfg; -#endif -}; - -/* transfer 8bit integer into string with binary representation */ -static void int2str_bin8(uint8_t val, char *buf) -{ - const int size = sizeof(val) * 8; - int i; - - for (i= 0; i < size; i++){ - *(buf++) = (val >> (size - 1)) ? '1' : '0'; - val <<= 1; - } - *buf = '\0'; //end the string with zero -} - -/* {{{ HW manipulation routines */ - -static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val) -{ - struct uda1341 *uda = clnt->driver_data; - unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing - int err = 0; - - uda->regs[reg] = val; - - if (uda->active) { - if (IS_DATA0(reg)) { - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1); - } else if (IS_DATA1(reg)) { - err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1); - } else if (IS_STATUS(reg)) { - err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1); - } else if (IS_EXTEND(reg)) { - buf[0] |= (reg - ext0) & 0x7; //EXT address - buf[1] |= val; //EXT data - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2); - } - } else - printk(KERN_ERR "UDA1341 codec not active!\n"); - return err; -} - -static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg) -{ - unsigned char val; - int err; - - err = l3_read(clnt, reg, &val, 1); - if (err == 1) - // use just 6bits - the rest is address of the reg - return val & 63; - return err < 0 ? err : -EIO; -} - -static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg) -{ - return reg < uda1341_reg_last; -} - -static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg, - unsigned short mask, unsigned short shift, - unsigned short value, int flush) -{ - int change; - unsigned short old, new; - struct uda1341 *uda = clnt->driver_data; - -#if 0 - printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n", - uda1341_reg_names[reg], mask, shift, value); -#endif - - if (!snd_uda1341_valid_reg(clnt, reg)) - return -EINVAL; - spin_lock(&uda->reg_lock); - old = uda->regs[reg]; - new = (old & ~(mask << shift)) | (value << shift); - change = old != new; - if (change) { - if (flush) uda->write(clnt, reg, new); - uda->regs[reg] = new; - } - spin_unlock(&uda->reg_lock); - return change; -} - -static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what, - unsigned short value, int flush) -{ - struct uda1341 *uda = clnt->driver_data; - int ret = 0; -#ifdef CONFIG_PM - int reg; -#endif - -#if 0 - printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value); -#endif - - uda->cfg[what] = value; - - switch(what) { - case CMD_RESET: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE - uda->cfg[CMD_RESET]=0; - break; - case CMD_FS: - ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush); - break; - case CMD_FORMAT: - ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush); - break; - case CMD_OGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush); - break; - case CMD_IGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush); - break; - case CMD_DAC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush); - break; - case CMD_ADC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush); - break; - case CMD_VOLUME: - ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush); - break; - case CMD_BASS: - ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush); - break; - case CMD_TREBBLE: - ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush); - break; - case CMD_PEAK: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush); - break; - case CMD_DEEMP: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush); - break; - case CMD_MUTE: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush); - break; - case CMD_FILTER: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush); - break; - case CMD_CH1: - ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush); - break; - case CMD_CH2: - ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush); - break; - case CMD_MIC: - ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush); - break; - case CMD_MIXER: - ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush); - break; - case CMD_AGC: - ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush); - break; - case CMD_IG: - ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush); - ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush); - break; - case CMD_AGC_TIME: - ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush); - break; - case CMD_AGC_LEVEL: - ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush); - break; -#ifdef CONFIG_PM - case CMD_SUSPEND: - for (reg = stat0; reg < uda1341_reg_last; reg++) - uda->suspend_regs[reg] = uda->regs[reg]; - for (reg = 0; reg < CMD_LAST; reg++) - uda->suspend_cfg[reg] = uda->cfg[reg]; - break; - case CMD_RESUME: - for (reg = stat0; reg < uda1341_reg_last; reg++) - snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]); - for (reg = 0; reg < CMD_LAST; reg++) - uda->cfg[reg] = uda->suspend_cfg[reg]; - break; -#endif - default: - ret = -EINVAL; - break; - } - - if (!uda->active) - printk(KERN_ERR "UDA1341 codec not active!\n"); - return ret; -} - -/* }}} */ - -/* {{{ Proc interface */ -#ifdef CONFIG_PROC_FS - -static const char *format_names[] = { - "I2S-bus", - "LSB 16bits", - "LSB 18bits", - "LSB 20bits", - "MSB", - "in LSB 16bits/out MSB", - "in LSB 18bits/out MSB", - "in LSB 20bits/out MSB", -}; - -static const char *fs_names[] = { - "512*fs", - "384*fs", - "256*fs", - "Unused - bad value!", -}; - -static const char* bass_values[][16] = { - {"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", - "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB", - "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max -}; - -static const char *mic_sens_value[] = { - "-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used", -}; - -static const unsigned short AGC_atime[] = { - 11, 16, 11, 16, 21, 11, 16, 21, -}; - -static const unsigned short AGC_dtime[] = { - 100, 100, 200, 200, 200, 400, 400, 400, -}; - -static const char *AGC_level[] = { - "-9.0", "-11.5", "-15.0", "-17.5", -}; - -static const char *ig_small_value[] = { - "-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5", -}; - -/* - * this was computed as peak_value[i] = pow((63-i)*1.42,1.013) - * - * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2 - * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29 - * [61]=-2.78, [62] = -1.48, [63] = 0.0 - * I tried to compute it, but using but even using logarithm with base either 10 or 2 - * i was'n able to get values in the table from the formula. So I constructed another - * formula (see above) to interpolate the values as good as possible. If there is some - * mistake, please contact me on tomas.kasparek(a)seznam.cz. Thanks. - * UDA1341TS datasheet is available at: - * http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf - */ -static const char *peak_value[] = { - "-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB", - "-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB", - "-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB", - "-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB", - "-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB", - "-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB", - "-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB", - "-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB", - "-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB", -}; - -static void snd_uda1341_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int peak; - - peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1); - if (peak < 0) - peak = 0; - - snd_iprintf(buffer, "%s\n\n", uda->card->longname); - - // for information about computed values see UDA1341TS datasheet pages 15 - 21 - snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off"); - snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off"); - snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]); - snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]); - - snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]); - snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]); - snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]); - snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before"); - snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]); - - snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off"); - snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]); - - snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off"); - - if (uda->cfg[CMD_VOLUME] == 0) - snd_iprintf(buffer, "Volume : 0 dB\n"); - else if (uda->cfg[CMD_VOLUME] < 62) - snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1); - else - snd_iprintf(buffer, "Volume : -INF dB\n"); - snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]); - snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0); - snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]); - - - if(uda->cfg[CMD_CH1] < 31) - snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n", - ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1), - uda->cfg[CMD_CH1] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n"); - if(uda->cfg[CMD_CH2] < 31) - snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n", - ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1), - uda->cfg[CMD_CH2] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n"); - - if(uda->cfg[CMD_IG] > 5) - snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n", - (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]); -} - -static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int reg; - char buf[12]; - - for (reg = 0; reg < uda1341_reg_last; reg ++) { - if (reg == empty) - continue; - int2str_bin8(uda->regs[reg], buf); - snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf); - } - - int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf); - snd_iprintf(buffer, "DATA1 = %s\n", buf); -} -#endif /* CONFIG_PROC_FS */ - -static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt) -{ - struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "uda1341", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read); - if (! snd_card_proc_new(card, "uda1341-regs", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read); -} - -/* }}} */ - -/* {{{ Mixer controls setting */ - -/* {{{ UDA1341 single functions */ - -#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \ - .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 12) & 63; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - - uda->cfg[where] = val; - return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 enum functions */ - -#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \ - .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int where = kcontrol->private_value & 31; - const char **texts; - - // this register we don't handle this way - if (!uda1341_enum_items[where]) - return -EINVAL; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = uda1341_enum_items[where]; - - if (uinfo->value.enumerated.item >= uda1341_enum_items[where]) - uinfo->value.enumerated.item = uda1341_enum_items[where] - 1; - - texts = uda1341_enum_names[where]; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - - ucontrol->value.enumerated.item[0] = uda->cfg[where]; - return 0; -} - -static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - - uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask); - - return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 2regs functions */ - -#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \ - .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \ - .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \ - (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \ -} - - -static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg_1 = (kcontrol->private_value >> 5) & 15; - int reg_2 = (kcontrol->private_value >> 9) & 15; - int shift_1 = (kcontrol->private_value >> 13) & 7; - int shift_2 = (kcontrol->private_value >> 16) & 7; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - unsigned short val1, val2, val; - - val = ucontrol->value.integer.value[0]; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - val1 = val & mask_1; - val2 = (val / (mask_1 + 1)) & mask_2; - - if (invert) { - val1 = mask_1 - val1; - val2 = mask_2 - val2; - } - - uda->cfg[where] = invert ? mask - val : val; - - //FIXME - return value - snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH); - return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH); -} - -/* }}} */ - -static struct snd_kcontrol_new snd_uda1341_controls[] = { - UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1), - UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1), - - UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0), - UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0), - - UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0), - UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0), - - UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1), - UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1), - - UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0), - - UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0), - UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0), - UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0), - - UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0), - UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0), - - UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0), - UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0), - UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0), - UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0), - - UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0), -}; - -static void uda1341_free(struct l3_client *clnt) -{ - l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341) - kfree(clnt); -} - -static int uda1341_dev_free(struct snd_device *device) -{ - struct l3_client *clnt = device->device_data; - uda1341_free(clnt); - return 0; -} - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp) -{ - static struct snd_device_ops ops = { - .dev_free = uda1341_dev_free, - }; - struct l3_client *clnt; - int idx, err; - - if (snd_BUG_ON(!card)) - return -EINVAL; - - clnt = kzalloc(sizeof(*clnt), GFP_KERNEL); - if (clnt == NULL) - return -ENOMEM; - - if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) { - kfree(clnt); - return err; - } - - for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) { - uda1341_free(clnt); - return err; - } - } - - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) { - uda1341_free(clnt); - return err; - } - - *clntp = clnt; - strcpy(card->mixername, "UDA1341TS Mixer"); - ((struct uda1341 *)clnt->driver_data)->card = card; - - snd_uda1341_proc_init(card, clnt); - - return 0; -} - -/* }}} */ - -/* {{{ L3 operations */ - -static int uda1341_attach(struct l3_client *clnt) -{ - struct uda1341 *uda; - - uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL); - if (!uda) - return -ENOMEM; - - /* init fixed parts of my copy of registers */ - uda->regs[stat0] = STAT0; - uda->regs[stat1] = STAT1; - - uda->regs[data0_0] = DATA0_0; - uda->regs[data0_1] = DATA0_1; - uda->regs[data0_2] = DATA0_2; - - uda->write = snd_uda1341_codec_write; - uda->read = snd_uda1341_codec_read; - - spin_lock_init(&uda->reg_lock); - - clnt->driver_data = uda; - return 0; -} - -static void uda1341_detach(struct l3_client *clnt) -{ - kfree(clnt->driver_data); -} - -static int -uda1341_command(struct l3_client *clnt, int cmd, void *arg) -{ - if (cmd != CMD_READ_REG) - return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH); - - return snd_uda1341_codec_read(clnt, (int) arg); -} - -static int uda1341_open(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 1; - - /* init default configuration */ - snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY); - snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset - snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset - //at this moment should be QMUTED by h3600_audio_init - snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset - snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode - snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset - - return 0; -} - -static void uda1341_close(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 0; -} - -/* }}} */ - -/* {{{ Module and L3 initialization */ - -static struct l3_ops uda1341_ops = { - .open = uda1341_open, - .command = uda1341_command, - .close = uda1341_close, -}; - -static struct l3_driver uda1341_driver = { - .name = UDA1341_ALSA_NAME, - .attach_client = uda1341_attach, - .detach_client = uda1341_detach, - .ops = &uda1341_ops, - .owner = THIS_MODULE, -}; - -static int __init uda1341_init(void) -{ - return l3_add_driver(&uda1341_driver); -} - -static void __exit uda1341_exit(void) -{ - l3_del_driver(&uda1341_driver); -} - -module_init(uda1341_init); -module_exit(uda1341_exit); - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); - -EXPORT_SYMBOL(snd_chip_uda1341_mixer_new); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ -- 1.6.1.3

3 5

[alsa-devel] Intel HDA volume low
by Daniel Stodden 13 Mar '09

13 Mar '09

Hi. I've got a Lenovo Thinkpad T400 with snd_hda_intel: 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Lenovo Device 20f2 I think this is the same chip which recently quirks entries to work around docking issues on the Thinkpad x200. The codec is identified as a conexant CX20561 (Hermosa). Running Ubuntu 8.10 with 2.6.27, the Volume is extremely low. Quality seems okay, but setting Master and PCM outputs to 100% is barely audible. There's a bunch of similar reports on the Internet. I could not find any workaround but Window users usually report the volume should be fine. I tried 2.6.28 but to no avail. Will verify against latest git. Could someone help me fix this? I don't actually know much about the hardware in question, but would also willing to poke a little around in patch_conexant.c, if that could help identifying the issue. Provided someone someone around here has advice on what to try out. Cheers, Daniell

2 1

[alsa-devel] [PATCH 1/3][RFC] ASoC: pxa-ssp: Use 16-bit DMA for magician stereo
by Philipp Zabel 13 Mar '09

13 Mar '09

Please advise how this behaviour could be made configurable. I guess the only machines that will ever need this are HTC Magician, Blueangel and Himalaya. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 569c0a6..bc9d306 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -22,6 +22,7 @@ #include <linux/io.h> #include <asm/irq.h> +#include <asm/mach-types.h> #include <sound/core.h> #include <sound/pcm.h> @@ -634,8 +635,14 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* select correct DMA params */ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) dma = 1; /* capture DMA offset is 1,3 */ - if (chn == 2) - dma += 2; /* stereo DMA offset is 2, mono is 0 */ + /* FIXME: Magician needs a way to configure 16-bit DMA for stereo */ + if (machine_is_magician()) { + if (width == 32) + dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ + } else { + if (chn == 2) + dma += 2; /* stereo DMA offset is 2, mono is 0 */ + } cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); -- 1.6.2

6 24

[PATCH] ALSA: drop outdated and broken sa11xx-uda1341 driver
by Dmitry Artamonow 12 Mar '09

12 Mar '09

It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got merged into mainline. Since there's no way to use it on any of supported machines (iPaq h3100 or h3600), better drop it for now. It can be reimplemented later using ASoC infrastructure (there's already a driver for uda1341 codec in mainline, so only CPU and machine parts need to be written). Signed-off-by: Dmitry Artamonow <mad_soft(a)inbox.ru> Cc: Russell King <linux(a)arm.linux.org.uk> --- include/sound/uda1341.h | 126 ------ sound/arm/Kconfig | 11 - sound/arm/Makefile | 3 - sound/arm/sa11xx-uda1341.c | 984 -------------------------------------------- sound/i2c/Makefile | 2 - sound/i2c/l3/Makefile | 8 - sound/i2c/l3/uda1341.c | 935 ----------------------------------------- 7 files changed, 0 insertions(+), 2069 deletions(-) delete mode 100644 include/sound/uda1341.h delete mode 100644 sound/arm/sa11xx-uda1341.c delete mode 100644 sound/i2c/l3/Makefile delete mode 100644 sound/i2c/l3/uda1341.c diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h deleted file mode 100644 index 110d5dc..0000000 --- a/include/sound/uda1341.h +++ /dev/null @@ -1,126 +0,0 @@ -/* - * linux/include/linux/l3/uda1341.h - * - * Philips UDA1341 mixer device driver for ALSA - * - * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek(a)seznam.cz> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek Initial release - based on uda1341.h from OSS - * 2002-03-30 Tomas Kasparek Proc filesystem support, complete mixer and DSP - * features support - */ - -#define UDA1341_ALSA_NAME "snd-uda1341" - -/* - * Default rate set after inicialization - */ -#define AUDIO_RATE_DEFAULT 44100 - -/* - * UDA1341 L3 address and command types - */ -#define UDA1341_L3ADDR 5 -#define UDA1341_DATA0 (UDA1341_L3ADDR << 2 | 0) -#define UDA1341_DATA1 (UDA1341_L3ADDR << 2 | 1) -#define UDA1341_STATUS (UDA1341_L3ADDR << 2 | 2) - -enum uda1341_onoff { - OFF=0, - ON, -}; - -enum uda1341_format { - I2S=0, - LSB16, - LSB18, - LSB20, - MSB, - LSB16MSB, - LSB18MSB, - LSB20MSB, -}; - -enum uda1341_fs { - F512=0, - F384, - F256, - Funused, -}; - -enum uda1341_peak { - BEFORE=0, - AFTER, -}; - -enum uda1341_filter { - FLAT=0, - MIN, - MIN2, - MAX, -}; - -enum uda1341_mixer { - DOUBLE, - LINE, - MIC, - MIXER, -}; - -enum uda1341_deemp { - NONE, - D32, - D44, - D48, -}; - -enum uda1341_config { - CMD_READ_REG = 0, - CMD_RESET, - CMD_FS, - CMD_FORMAT, - CMD_OGAIN, - CMD_IGAIN, - CMD_DAC, - CMD_ADC, - CMD_VOLUME, - CMD_BASS, - CMD_TREBBLE, - CMD_PEAK, - CMD_DEEMP, - CMD_MUTE, - CMD_FILTER, - CMD_CH1, - CMD_CH2, - CMD_MIC, - CMD_MIXER, - CMD_AGC, - CMD_IG, - CMD_AGC_TIME, - CMD_AGC_LEVEL, -#ifdef CONFIG_PM - CMD_SUSPEND, - CMD_RESUME, -#endif - CMD_LAST, -}; - -enum write_through { - //used in update_bits (write_cfg) to avoid l3_write - just update local copy of regs. - REGS_ONLY=0, - //update local regs and write value to uda1341 - do l3_write - FLUSH, -}; - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clnt); - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de4..885683a 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -11,17 +11,6 @@ menuconfig SND_ARM if SND_ARM -config SND_SA11XX_UDA1341 - tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && L3 - select SND_PCM - help - Say Y here if you have a Compaq iPaq H3x00 handheld computer - and want to use its Philips UDA 1341 audio chip. - - To compile this driver as a module, choose M here: the module - will be called snd-sa11xx-uda1341. - config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de1..5a549ed 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -2,9 +2,6 @@ # Makefile for ALSA # -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o -snd-sa11xx-uda1341-objs := sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o devdma.o diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c deleted file mode 100644 index 7101d3d..0000000 --- a/sound/arm/sa11xx-uda1341.c +++ /dev/null @@ -1,984 +0,0 @@ -/* - * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard - * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek(a)seznam.cz> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS - * 2002-03-20 Tomas Kasparek playback over ALSA is working - * 2002-03-28 Tomas Kasparek playback over OSS emulation is working - * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA) - * 2002-03-29 Tomas Kasparek capture is working (OSS emulation) - * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates) - * 2003-02-14 Brian Avery fixed full duplex mode, other updates - * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL) - * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel - * working suspend and resume - * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again - * merged HAL layer (patches from Brian) - */ - -/*************************************************************************************************** -* -* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai -* available in the Alsa doc section on the website -* -* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100. -* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated -* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it. -* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the -* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which -* is a mem loc that always decodes to 0's w/ no off chip access. -* -* Some alsa terminology: -* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes -* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte -* buffer and 4 periods in the runtime structure this means we'll get an int every 256 -* bytes or 4 times per buffer. -* A number of the sizes are in frames rather than bytes, use frames_to_bytes and -* bytes_to_frames to convert. The easiest way to tell the units is to look at the -* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t -* -* Notes about the pointer fxn: -* The pointer fxn needs to return the offset into the dma buffer in frames. -* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts. -* -* Notes about pause/resume -* Implementing this would be complicated so it's skipped. The problem case is: -* A full duplex connection is going, then play is paused. At this point you need to start xmitting -* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd -* need to save off the dma info, and restore it properly on a resume. Yeach! -* -* Notes about transfer methods: -* The async write calls fail. I probably need to implement something else to support them? -* -***************************************************************************************************/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/err.h> -#include <linux/platform_device.h> -#include <linux/errno.h> -#include <linux/ioctl.h> -#include <linux/delay.h> -#include <linux/slab.h> - -#ifdef CONFIG_PM -#include <linux/pm.h> -#endif - -#include <mach/hardware.h> -#include <mach/h3600.h> -#include <asm/mach-types.h> -#include <asm/dma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/initval.h> - -#include <linux/l3/l3.h> - -#undef DEBUG_MODE -#undef DEBUG_FUNCTION_NAMES -#include <sound/uda1341.h> - -/* - * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? - * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this - * module for Familiar 0.6.1 - */ - -/* {{{ Type definitions */ - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); - -static char *id; /* ID for this card */ - -module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); - -struct audio_stream { - char *id; /* identification string */ - int stream_id; /* numeric identification */ - dma_device_t dma_dev; /* device identifier for DMA */ -#ifdef HH_VERSION - dmach_t dmach; /* dma channel identification */ -#else - dma_regs_t *dma_regs; /* points to our DMA registers */ -#endif - unsigned int active:1; /* we are using this stream for transfer now */ - int period; /* current transfer period */ - int periods; /* current count of periods registerd in the DMA engine */ - int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ - unsigned int old_offset; - spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ - struct snd_pcm_substream *stream; -}; - -struct sa11xx_uda1341 { - struct snd_card *card; - struct l3_client *uda1341; - struct snd_pcm *pcm; - long samplerate; - struct audio_stream s[2]; /* playback & capture */ -}; - -static unsigned int rates[] = { - 8000, 10666, 10985, 14647, - 16000, 21970, 22050, 24000, - 29400, 32000, 44100, 48000, -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct platform_device *device; - -/* }}} */ - -/* {{{ Clock and sample rate stuff */ - -/* - * Stop-gap solution until rest of hh.org HAL stuff is merged. - */ -#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) -#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) - -#ifdef CONFIG_SA1100_H3XXX -#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) -#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) -#else -#error This driver could serve H3x00 handhelds only! -#endif - -static void sa11xx_uda1341_set_audio_clock(long val) -{ - switch (val) { - case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ - GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - - case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ - GPSR = GPIO_H3600_CLK_SET0; - GPCR = GPIO_H3600_CLK_SET1; - break; - - case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ - GPCR = GPIO_H3600_CLK_SET0; - GPSR = GPIO_H3600_CLK_SET1; - break; - - case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ - GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - } -} - -static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate) -{ - int clk_div = 0; - int clk=0; - - /* We don't want to mess with clocks when frames are in flight */ - Ser4SSCR0 &= ~SSCR0_SSE; - /* wait for any frame to complete */ - udelay(125); - - /* - * We have the following clock sources: - * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz - * Those can be divided either by 256, 384 or 512. - * This makes up 12 combinations for the following samplerates... - */ - if (rate >= 48000) - rate = 48000; - else if (rate >= 44100) - rate = 44100; - else if (rate >= 32000) - rate = 32000; - else if (rate >= 29400) - rate = 29400; - else if (rate >= 24000) - rate = 24000; - else if (rate >= 22050) - rate = 22050; - else if (rate >= 21970) - rate = 21970; - else if (rate >= 16000) - rate = 16000; - else if (rate >= 14647) - rate = 14647; - else if (rate >= 10985) - rate = 10985; - else if (rate >= 10666) - rate = 10666; - else - rate = 8000; - - /* Set the external clock generator */ - - sa11xx_uda1341_set_audio_clock(rate); - - /* Select the clock divisor */ - switch (rate) { - case 8000: - case 10985: - case 22050: - case 24000: - clk = F512; - clk_div = SSCR0_SerClkDiv(16); - break; - case 16000: - case 21970: - case 44100: - case 48000: - clk = F256; - clk_div = SSCR0_SerClkDiv(8); - break; - case 10666: - case 14647: - case 29400: - case 32000: - clk = F384; - clk_div = SSCR0_SerClkDiv(12); - break; - } - - /* FMT setting should be moved away when other FMTs are added (FIXME) */ - l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); - - l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); - Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; - sa11xx_uda1341->samplerate = rate; -} - -/* }}} */ - -/* {{{ HW init and shutdown */ - -static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - unsigned long flags; - - /* Setup DMA stuff */ - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; - - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; - - /* Initialize the UDA1341 internal state */ - - /* Setup the uarts */ - local_irq_save(flags); - GAFR |= (GPIO_SSP_CLK); - GPDR &= ~(GPIO_SSP_CLK); - Ser4SSCR0 = 0; - Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); - Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; - Ser4SSCR0 |= SSCR0_SSE; - local_irq_restore(flags); - - /* Enable the audio power */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* Wait for the UDA1341 to wake up */ - mdelay(1); //FIXME - was removed by Perex - Why? - - /* Initialize the UDA1341 internal state */ - l3_open(sa11xx_uda1341->uda1341); - - /* external clock configuration (after l3_open - regs must be initialized */ - sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); - - /* Wait for the UDA1341 to wake up */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - mdelay(1); - - /* make the left and right channels unswapped (flip the WS latch) */ - Ser4SSDR = 0; - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - /* mute on */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* disable the audio power and all signals leading to the audio chip */ - l3_close(sa11xx_uda1341->uda1341); - Ser4SSCR0 = 0; - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - - /* power off and mute off */ - /* FIXME - is muting off necesary??? */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -/* }}} */ - -/* {{{ DMA staff */ - -/* - * these are the address and sizes used to fill the xmit buffer - * so we can get a clock in record only mode - */ -#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS -#define FORCE_CLOCK_SIZE 4096 // was 2048 - -// FIXME Why this value exactly - wrote comment -#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ - -#ifdef HH_VERSION - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int)) -{ - int ret; - - ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); - if (ret < 0) { - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; - } - sa1100_dma_set_callback(s->dmach, callback); - return 0; -} - -static inline void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dmach); - s->dmach = -1; -} - -#else - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *)) -{ - int ret; - - ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); - if (ret < 0) - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; -} - -static void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dma_regs); - s->dma_regs = 0; -} - -#endif - -static u_int audio_get_dma_pos(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int offset; - unsigned long flags; - dma_addr_t addr; - - // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel - spin_lock_irqsave(&s->dma_lock, flags); -#ifdef HH_VERSION - sa1100_dma_get_current(s->dmach, NULL, &addr); -#else - addr = sa1100_get_dma_pos((s)->dma_regs); -#endif - offset = addr - runtime->dma_addr; - spin_unlock_irqrestore(&s->dma_lock, flags); - - offset = bytes_to_frames(runtime,offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -/* - * this stops the dma and clears the dma ptrs - */ -static void audio_stop_dma(struct audio_stream *s) -{ - unsigned long flags; - - spin_lock_irqsave(&s->dma_lock, flags); - s->active = 0; - s->period = 0; - /* this stops the dma channel and clears the buffer ptrs */ -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - sa1100_clear_dma(s->dma_regs); -#endif - spin_unlock_irqrestore(&s->dma_lock, flags); -} - -static void audio_process_dma(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime; - unsigned int dma_size; - unsigned int offset; - int ret; - - /* we are requested to process synchronization DMA transfer */ - if (s->tx_spin) { - if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK)) - return; - /* fill the xmit dma buffers and return */ -#ifdef HH_VERSION - sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); -#else - while (1) { - ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); - if (ret) - return; - } -#endif - return; - } - - /* must be set here - only valid for running streams, not for forced_clock dma fills */ - runtime = substream->runtime; - while (s->active && s->periods < runtime->periods) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - if (s->old_offset) { - /* a little trick, we need resume from old position */ - offset = frames_to_bytes(runtime, s->old_offset - 1); - s->old_offset = 0; - s->periods = 0; - s->period = offset / dma_size; - offset %= dma_size; - dma_size = dma_size - offset; - if (!dma_size) - continue; /* special case */ - } else { - offset = dma_size * s->period; - snd_BUG_ON(dma_size > DMA_BUF_SIZE); - } -#ifdef HH_VERSION - ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size); - if (ret) - return; //FIXME -#else - ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size); - if (ret) { - printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret); - return; - } -#endif - - s->period++; - s->period %= runtime->periods; - s->periods++; - } -} - -#ifdef HH_VERSION -static void audio_dma_callback(void *data, int size) -#else -static void audio_dma_callback(void *data) -#endif -{ - struct audio_stream *s = data; - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - spin_lock(&s->dma_lock); - if (!s->tx_spin && s->periods > 0) - s->periods--; - audio_process_dma(s); - spin_unlock(&s->dma_lock); -} - -/* }}} */ - -/* {{{ PCM setting */ - -/* {{{ trigger & timer */ - -static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - int stream_id = substream->pstr->stream; - struct audio_stream *s = &chip->s[stream_id]; - struct audio_stream *s1 = &chip->s[stream_id ^ 1]; - int err = 0; - - /* note local interrupts are already disabled in the midlevel code */ - spin_lock(&s->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* now we need to make sure a record only stream has a clock */ - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s1->tx_spin = 1; - audio_process_dma(s1); - } - /* this case is when you were recording then you turn on a - * playback stream so we stop (also clears it) the dma first, - * clear the sync flag and then we let it turned on - */ - else { - s->tx_spin = 0; - } - - /* requested stream startup */ - s->active = 1; - audio_process_dma(s); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - audio_stop_dma(s); - - /* - * now we need to make sure a record only stream has a clock - * so if we're stopping a playback with an active capture - * we need to turn the 0 fill dma on for the xmit side - */ - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s->tx_spin = 1; - audio_process_dma(s); - } - /* - * we killed a capture only stream, so we should also kill - * the zero fill transmit - */ - else { - if (s1->tx_spin) { - s1->tx_spin = 0; - audio_stop_dma(s1); - } - } - - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - s->active = 0; -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - s->periods = 0; - break; - case SNDRV_PCM_TRIGGER_RESUME: - s->active = 1; - s->tx_spin = 0; - audio_process_dma(s); - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->active = 0; - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) { - if (s1->active) { - s->tx_spin = 1; - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - audio_process_dma(s); - } - } else { - if (s1->tx_spin) { - s1->tx_spin = 0; -#ifdef HH_VERSION - sa1100_dma_flush_all(s1->dmach); -#else - //FIXME - DMA API -#endif - } - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - s->active = 1; - if (s->old_offset) { - s->tx_spin = 0; - audio_process_dma(s); - break; - } - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } -#ifdef HH_VERSION - sa1100_dma_resume(s->dmach); -#else - //FIXME - DMA API -#endif - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&s->dma_lock); - return err; -} - -static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audio_stream *s = &chip->s[substream->pstr->stream]; - - /* set requested samplerate */ - sa11xx_uda1341_set_samplerate(chip, runtime->rate); - - /* set requestd format when available */ - /* set FMT here !!! FIXME */ - - s->period = 0; - s->periods = 0; - - return 0; -} - -static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - return audio_get_dma_pos(&chip->s[substream->pstr->stream]); -} - -/* }}} */ - -static struct snd_pcm_hardware snd_sa11xx_uda1341_capture = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_sa11xx_uda1341_playback = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int stream_id = substream->pstr->stream; - int err; - - chip->s[stream_id].stream = substream; - - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = snd_sa11xx_uda1341_playback; - else - runtime->hw = snd_sa11xx_uda1341_capture; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) - return err; - - return 0; -} - -static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - - chip->s[substream->pstr->stream].stream = NULL; - return 0; -} - -/* {{{ HW params & free */ - -static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - -static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -/* }}} */ - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0) - return err; - - /* - * this sets up our initial buffers and sets the dma_type to isa. - * isa works but I'm not sure why (or if) it's the right choice - * this may be too large, trying it for now - */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, 64*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops); - pcm->private_data = sa11xx_uda1341; - pcm->info_flags = 0; - strcpy(pcm->name, "UDA1341 PCM"); - - sa11xx_uda1341_audio_init(sa11xx_uda1341); - - /* setup DMA controller */ - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback); - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback); - - sa11xx_uda1341->pcm = pcm; - - return 0; -} - -/* }}} */ - -/* {{{ module init & exit */ - -#ifdef CONFIG_PM - -static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr, - pm_message_t state) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); -#ifdef HH_VERSION - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - l3_command(chip->uda1341, CMD_SUSPEND, NULL); - sa11xx_uda1341_audio_shutdown(chip); - - return 0; -} - -static int snd_sa11xx_uda1341_resume(struct platform_device *devptr) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - sa11xx_uda1341_audio_init(chip); - l3_command(chip->uda1341, CMD_RESUME, NULL); -#ifdef HH_VERSION - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif /* COMFIG_PM */ - -void snd_sa11xx_uda1341_free(struct snd_card *card) -{ - struct sa11xx_uda1341 *chip = card->private_data; - - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); -} - -static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) -{ - int err; - struct snd_card *card; - struct sa11xx_uda1341 *chip; - - /* register the soundcard */ - err = snd_card_create(-1, id, THIS_MODULE, - sizeof(struct sa11xx_uda1341), &card); - if (err < 0) - return err; - - chip = card->private_data; - spin_lock_init(&chip->s[0].dma_lock); - spin_lock_init(&chip->s[1].dma_lock); - - card->private_free = snd_sa11xx_uda1341_free; - chip->card = card; - chip->samplerate = AUDIO_RATE_DEFAULT; - - // mixer - if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341))) - goto nodev; - - // PCM - if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0) - goto nodev; - - strcpy(card->driver, "UDA1341"); - strcpy(card->shortname, "H3600 UDA1341TS"); - sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); - - snd_card_set_dev(card, &devptr->dev); - - if ((err = snd_card_register(card)) == 0) { - printk(KERN_INFO "iPAQ audio support initialized\n"); - platform_set_drvdata(devptr, card); - return 0; - } - - nodev: - snd_card_free(card); - return err; -} - -static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr) -{ - snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; -} - -#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341" - -static struct platform_driver sa11xx_uda1341_driver = { - .probe = sa11xx_uda1341_probe, - .remove = __devexit_p(sa11xx_uda1341_remove), -#ifdef CONFIG_PM - .suspend = snd_sa11xx_uda1341_suspend, - .resume = snd_sa11xx_uda1341_resume, -#endif - .driver = { - .name = SA11XX_UDA1341_DRIVER, - }, -}; - -static int __init sa11xx_uda1341_init(void) -{ - int err; - - if (!machine_is_h3xxx()) - return -ENODEV; - if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) - return err; - device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (!IS_ERR(device)) { - if (platform_get_drvdata(device)) - return 0; - platform_device_unregister(device); - err = -ENODEV; - } else - err = PTR_ERR(device); - platform_driver_unregister(&sa11xx_uda1341_driver); - return err; -} - -static void __exit sa11xx_uda1341_exit(void) -{ - platform_device_unregister(device); - platform_driver_unregister(&sa11xx_uda1341_driver); -} - -module_init(sa11xx_uda1341_init); -module_exit(sa11xx_uda1341_exit); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 3797066..36879bf 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -obj-$(CONFIG_L3) += l3/ - obj-$(CONFIG_SND) += other/ # Toplevel Module Dependency diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile deleted file mode 100644 index 49455b8..0000000 --- a/sound/i2c/l3/Makefile +++ /dev/null @@ -1,8 +0,0 @@ -# -# Makefile for ALSA -# - -snd-uda1341-objs := uda1341.o - -# Module Dependency -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c deleted file mode 100644 index 9840eb4..0000000 --- a/sound/i2c/l3/uda1341.c +++ /dev/null @@ -1,935 +0,0 @@ -/* - * Philips UDA1341 mixer device driver - * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek(a)seznam.cz> - * - * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS - * 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble) - * 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP - * features support - * 2002-04-12 Tomas Kasparek proc interface update, code cleanup - * 2002-05-12 Tomas Kasparek another code cleanup - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/types.h> -#include <linux/slab.h> -#include <linux/errno.h> -#include <linux/ioctl.h> - -#include <asm/uaccess.h> - -#include <sound/core.h> -#include <sound/control.h> -#include <sound/initval.h> -#include <sound/info.h> - -#include <linux/l3/l3.h> - -#include <sound/uda1341.h> - -/* {{{ HW regs definition */ - -#define STAT0 0x00 -#define STAT1 0x80 -#define STAT_MASK 0x80 - -#define DATA0_0 0x00 -#define DATA0_1 0x40 -#define DATA0_2 0x80 -#define DATA_MASK 0xc0 - -#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2) -#define IS_DATA1(x) ((x) == data1) -#define IS_STATUS(x) ((x) == stat0 || (x) == stat1) -#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6) - -/* }}} */ - - -static const char *peak_names[] = { - "before", - "after", -}; - -static const char *filter_names[] = { - "flat", - "min", - "min", - "max", -}; - -static const char *mixer_names[] = { - "double differential", - "input channel 1 (line in)", - "input channel 2 (microphone)", - "digital mixer", -}; - -static const char *deemp_names[] = { - "none", - "32 kHz", - "44.1 kHz", - "48 kHz", -}; - -enum uda1341_regs_names { - stat0, - stat1, - data0_0, - data0_1, - data0_2, - data1, - ext0, - ext1, - ext2, - empty, - ext4, - ext5, - ext6, - uda1341_reg_last, -}; - -static const char *uda1341_reg_names[] = { - "stat 0 ", - "stat 1 ", - "data 00", - "data 01", - "data 02", - "data 1 ", - "ext 0", - "ext 1", - "ext 2", - "empty", - "ext 4", - "ext 5", - "ext 6", -}; - -static const int uda1341_enum_items[] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 2, //peak - before/after - 4, //deemp - none/32/44.1/48 - 0, - 4, //filter - flat/min/min/max - 0, 0, 0, - 4, //mixer - differ/line/mic/mixer - 0, 0, 0, 0, 0, -}; - -static const char ** uda1341_enum_names[] = { - NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, - peak_names, //peak - before/after - deemp_names, //deemp - none/32/44.1/48 - NULL, - filter_names, //filter - flat/min/min/max - NULL, NULL, NULL, - mixer_names, //mixer - differ/line/mic/mixer - NULL, NULL, NULL, NULL, NULL, -}; - -typedef int uda1341_cfg[CMD_LAST]; - -struct uda1341 { - int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val); - int (*read) (struct l3_client *uda1341, unsigned short reg); - unsigned char regs[uda1341_reg_last]; - int active; - spinlock_t reg_lock; - struct snd_card *card; - uda1341_cfg cfg; -#ifdef CONFIG_PM - unsigned char suspend_regs[uda1341_reg_last]; - uda1341_cfg suspend_cfg; -#endif -}; - -/* transfer 8bit integer into string with binary representation */ -static void int2str_bin8(uint8_t val, char *buf) -{ - const int size = sizeof(val) * 8; - int i; - - for (i= 0; i < size; i++){ - *(buf++) = (val >> (size - 1)) ? '1' : '0'; - val <<= 1; - } - *buf = '\0'; //end the string with zero -} - -/* {{{ HW manipulation routines */ - -static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val) -{ - struct uda1341 *uda = clnt->driver_data; - unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing - int err = 0; - - uda->regs[reg] = val; - - if (uda->active) { - if (IS_DATA0(reg)) { - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1); - } else if (IS_DATA1(reg)) { - err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1); - } else if (IS_STATUS(reg)) { - err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1); - } else if (IS_EXTEND(reg)) { - buf[0] |= (reg - ext0) & 0x7; //EXT address - buf[1] |= val; //EXT data - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2); - } - } else - printk(KERN_ERR "UDA1341 codec not active!\n"); - return err; -} - -static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg) -{ - unsigned char val; - int err; - - err = l3_read(clnt, reg, &val, 1); - if (err == 1) - // use just 6bits - the rest is address of the reg - return val & 63; - return err < 0 ? err : -EIO; -} - -static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg) -{ - return reg < uda1341_reg_last; -} - -static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg, - unsigned short mask, unsigned short shift, - unsigned short value, int flush) -{ - int change; - unsigned short old, new; - struct uda1341 *uda = clnt->driver_data; - -#if 0 - printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n", - uda1341_reg_names[reg], mask, shift, value); -#endif - - if (!snd_uda1341_valid_reg(clnt, reg)) - return -EINVAL; - spin_lock(&uda->reg_lock); - old = uda->regs[reg]; - new = (old & ~(mask << shift)) | (value << shift); - change = old != new; - if (change) { - if (flush) uda->write(clnt, reg, new); - uda->regs[reg] = new; - } - spin_unlock(&uda->reg_lock); - return change; -} - -static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what, - unsigned short value, int flush) -{ - struct uda1341 *uda = clnt->driver_data; - int ret = 0; -#ifdef CONFIG_PM - int reg; -#endif - -#if 0 - printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value); -#endif - - uda->cfg[what] = value; - - switch(what) { - case CMD_RESET: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE - uda->cfg[CMD_RESET]=0; - break; - case CMD_FS: - ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush); - break; - case CMD_FORMAT: - ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush); - break; - case CMD_OGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush); - break; - case CMD_IGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush); - break; - case CMD_DAC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush); - break; - case CMD_ADC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush); - break; - case CMD_VOLUME: - ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush); - break; - case CMD_BASS: - ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush); - break; - case CMD_TREBBLE: - ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush); - break; - case CMD_PEAK: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush); - break; - case CMD_DEEMP: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush); - break; - case CMD_MUTE: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush); - break; - case CMD_FILTER: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush); - break; - case CMD_CH1: - ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush); - break; - case CMD_CH2: - ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush); - break; - case CMD_MIC: - ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush); - break; - case CMD_MIXER: - ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush); - break; - case CMD_AGC: - ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush); - break; - case CMD_IG: - ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush); - ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush); - break; - case CMD_AGC_TIME: - ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush); - break; - case CMD_AGC_LEVEL: - ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush); - break; -#ifdef CONFIG_PM - case CMD_SUSPEND: - for (reg = stat0; reg < uda1341_reg_last; reg++) - uda->suspend_regs[reg] = uda->regs[reg]; - for (reg = 0; reg < CMD_LAST; reg++) - uda->suspend_cfg[reg] = uda->cfg[reg]; - break; - case CMD_RESUME: - for (reg = stat0; reg < uda1341_reg_last; reg++) - snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]); - for (reg = 0; reg < CMD_LAST; reg++) - uda->cfg[reg] = uda->suspend_cfg[reg]; - break; -#endif - default: - ret = -EINVAL; - break; - } - - if (!uda->active) - printk(KERN_ERR "UDA1341 codec not active!\n"); - return ret; -} - -/* }}} */ - -/* {{{ Proc interface */ -#ifdef CONFIG_PROC_FS - -static const char *format_names[] = { - "I2S-bus", - "LSB 16bits", - "LSB 18bits", - "LSB 20bits", - "MSB", - "in LSB 16bits/out MSB", - "in LSB 18bits/out MSB", - "in LSB 20bits/out MSB", -}; - -static const char *fs_names[] = { - "512*fs", - "384*fs", - "256*fs", - "Unused - bad value!", -}; - -static const char* bass_values[][16] = { - {"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", - "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB", - "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max -}; - -static const char *mic_sens_value[] = { - "-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used", -}; - -static const unsigned short AGC_atime[] = { - 11, 16, 11, 16, 21, 11, 16, 21, -}; - -static const unsigned short AGC_dtime[] = { - 100, 100, 200, 200, 200, 400, 400, 400, -}; - -static const char *AGC_level[] = { - "-9.0", "-11.5", "-15.0", "-17.5", -}; - -static const char *ig_small_value[] = { - "-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5", -}; - -/* - * this was computed as peak_value[i] = pow((63-i)*1.42,1.013) - * - * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2 - * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29 - * [61]=-2.78, [62] = -1.48, [63] = 0.0 - * I tried to compute it, but using but even using logarithm with base either 10 or 2 - * i was'n able to get values in the table from the formula. So I constructed another - * formula (see above) to interpolate the values as good as possible. If there is some - * mistake, please contact me on tomas.kasparek(a)seznam.cz. Thanks. - * UDA1341TS datasheet is available at: - * http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf - */ -static const char *peak_value[] = { - "-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB", - "-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB", - "-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB", - "-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB", - "-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB", - "-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB", - "-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB", - "-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB", - "-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB", -}; - -static void snd_uda1341_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int peak; - - peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1); - if (peak < 0) - peak = 0; - - snd_iprintf(buffer, "%s\n\n", uda->card->longname); - - // for information about computed values see UDA1341TS datasheet pages 15 - 21 - snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off"); - snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off"); - snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]); - snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]); - - snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]); - snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]); - snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]); - snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before"); - snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]); - - snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off"); - snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]); - - snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off"); - - if (uda->cfg[CMD_VOLUME] == 0) - snd_iprintf(buffer, "Volume : 0 dB\n"); - else if (uda->cfg[CMD_VOLUME] < 62) - snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1); - else - snd_iprintf(buffer, "Volume : -INF dB\n"); - snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]); - snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0); - snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]); - - - if(uda->cfg[CMD_CH1] < 31) - snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n", - ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1), - uda->cfg[CMD_CH1] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n"); - if(uda->cfg[CMD_CH2] < 31) - snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n", - ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1), - uda->cfg[CMD_CH2] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n"); - - if(uda->cfg[CMD_IG] > 5) - snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n", - (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]); -} - -static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int reg; - char buf[12]; - - for (reg = 0; reg < uda1341_reg_last; reg ++) { - if (reg == empty) - continue; - int2str_bin8(uda->regs[reg], buf); - snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf); - } - - int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf); - snd_iprintf(buffer, "DATA1 = %s\n", buf); -} -#endif /* CONFIG_PROC_FS */ - -static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt) -{ - struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "uda1341", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read); - if (! snd_card_proc_new(card, "uda1341-regs", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read); -} - -/* }}} */ - -/* {{{ Mixer controls setting */ - -/* {{{ UDA1341 single functions */ - -#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \ - .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 12) & 63; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - - uda->cfg[where] = val; - return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 enum functions */ - -#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \ - .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int where = kcontrol->private_value & 31; - const char **texts; - - // this register we don't handle this way - if (!uda1341_enum_items[where]) - return -EINVAL; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = uda1341_enum_items[where]; - - if (uinfo->value.enumerated.item >= uda1341_enum_items[where]) - uinfo->value.enumerated.item = uda1341_enum_items[where] - 1; - - texts = uda1341_enum_names[where]; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - - ucontrol->value.enumerated.item[0] = uda->cfg[where]; - return 0; -} - -static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - - uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask); - - return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 2regs functions */ - -#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \ - .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \ - .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \ - (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \ -} - - -static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg_1 = (kcontrol->private_value >> 5) & 15; - int reg_2 = (kcontrol->private_value >> 9) & 15; - int shift_1 = (kcontrol->private_value >> 13) & 7; - int shift_2 = (kcontrol->private_value >> 16) & 7; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - unsigned short val1, val2, val; - - val = ucontrol->value.integer.value[0]; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - val1 = val & mask_1; - val2 = (val / (mask_1 + 1)) & mask_2; - - if (invert) { - val1 = mask_1 - val1; - val2 = mask_2 - val2; - } - - uda->cfg[where] = invert ? mask - val : val; - - //FIXME - return value - snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH); - return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH); -} - -/* }}} */ - -static struct snd_kcontrol_new snd_uda1341_controls[] = { - UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1), - UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1), - - UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0), - UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0), - - UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0), - UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0), - - UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1), - UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1), - - UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0), - - UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0), - UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0), - UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0), - - UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0), - UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0), - - UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0), - UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0), - UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0), - UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0), - - UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0), -}; - -static void uda1341_free(struct l3_client *clnt) -{ - l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341) - kfree(clnt); -} - -static int uda1341_dev_free(struct snd_device *device) -{ - struct l3_client *clnt = device->device_data; - uda1341_free(clnt); - return 0; -} - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp) -{ - static struct snd_device_ops ops = { - .dev_free = uda1341_dev_free, - }; - struct l3_client *clnt; - int idx, err; - - if (snd_BUG_ON(!card)) - return -EINVAL; - - clnt = kzalloc(sizeof(*clnt), GFP_KERNEL); - if (clnt == NULL) - return -ENOMEM; - - if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) { - kfree(clnt); - return err; - } - - for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) { - uda1341_free(clnt); - return err; - } - } - - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) { - uda1341_free(clnt); - return err; - } - - *clntp = clnt; - strcpy(card->mixername, "UDA1341TS Mixer"); - ((struct uda1341 *)clnt->driver_data)->card = card; - - snd_uda1341_proc_init(card, clnt); - - return 0; -} - -/* }}} */ - -/* {{{ L3 operations */ - -static int uda1341_attach(struct l3_client *clnt) -{ - struct uda1341 *uda; - - uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL); - if (!uda) - return -ENOMEM; - - /* init fixed parts of my copy of registers */ - uda->regs[stat0] = STAT0; - uda->regs[stat1] = STAT1; - - uda->regs[data0_0] = DATA0_0; - uda->regs[data0_1] = DATA0_1; - uda->regs[data0_2] = DATA0_2; - - uda->write = snd_uda1341_codec_write; - uda->read = snd_uda1341_codec_read; - - spin_lock_init(&uda->reg_lock); - - clnt->driver_data = uda; - return 0; -} - -static void uda1341_detach(struct l3_client *clnt) -{ - kfree(clnt->driver_data); -} - -static int -uda1341_command(struct l3_client *clnt, int cmd, void *arg) -{ - if (cmd != CMD_READ_REG) - return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH); - - return snd_uda1341_codec_read(clnt, (int) arg); -} - -static int uda1341_open(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 1; - - /* init default configuration */ - snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY); - snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset - snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset - //at this moment should be QMUTED by h3600_audio_init - snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset - snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode - snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset - - return 0; -} - -static void uda1341_close(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 0; -} - -/* }}} */ - -/* {{{ Module and L3 initialization */ - -static struct l3_ops uda1341_ops = { - .open = uda1341_open, - .command = uda1341_command, - .close = uda1341_close, -}; - -static struct l3_driver uda1341_driver = { - .name = UDA1341_ALSA_NAME, - .attach_client = uda1341_attach, - .detach_client = uda1341_detach, - .ops = &uda1341_ops, - .owner = THIS_MODULE, -}; - -static int __init uda1341_init(void) -{ - return l3_add_driver(&uda1341_driver); -} - -static void __exit uda1341_exit(void) -{ - l3_del_driver(&uda1341_driver); -} - -module_init(uda1341_init); -module_exit(uda1341_exit); - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); - -EXPORT_SYMBOL(snd_chip_uda1341_mixer_new); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ -- 1.6.1.3 -- Best regards, Dmitry "MAD" Artamonow

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[alsa-devel] [PATCH] fix headphone settings and master volume (Conexant CX20551 0x103c30b2)
by Gregorio Guidi 12 Mar '09

12 Mar '09

Hi, I have an Intel HDA card, Conexant CX20551 (Waikiki) model, Subsystem Id: 0x103c30b2 (HP DV2000 series). With these settings, there are 4 volume controls: "PCM-2" and "Speaker" to control speakers, "PCM" and "Headphone" to control headphones. At the moment "PCM-2", "Speaker", "PCM" work fine, but "Headphone" has no effect (also reported inside bug #3091). With the following patch, the "Headphone" volume/switch work as intended, with no effect on other functionalities. --- patch_conexant.c 2009-03-08 21:10:14.000000000 +0100 +++ patch_conexant.c.new 2009-03-08 21:09:51.000000000 +0100 @@ -1357,7 +1357,7 @@ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, /* HP, Speaker */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL,0x1}, + {0x13, AC_VERB_SET_CONNECT_SEL,0x0}, {0x1d, AC_VERB_SET_CONNECT_SEL,0x0}, /* Record selector: Mic */ {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, Also, there is no "Master" control to set the volume for both speakers and headphones. The patch below adds a master volume (and a master switch) that work ok for me. By the way, with the patch the light on the mute button changes between blue and red, which is a nice plus. --- patch_conexant.c 2009-03-08 21:11:14.000000000 +0100 +++ patch_conexant.c.new 2009-03-08 21:11:01.000000000 +0100 @@ -1293,6 +1293,15 @@ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT), + HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5047_hp_master_sw_put, + .private_value = 0x13, + }, {} }; I am no expert of Conexant hardware and I don't know if those changes are correct for all the devices that share the same code (the 103c:30xx), but I'm confident that they are the right ones for the 103c:30b2 device. Can someone take a look? Thanks. Gregorio Guidi

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[alsa-devel] sys/class entry not created for jack in machine driver
by Lopez Cruz, Misael 12 Mar '09

12 Mar '09

Hi, I just noticed that the sys/class entry for the jack is not being created, in my case for headset jack in sdp3430 machine driver. When creating the SoC jack, the soc-jack mechanism creates a new generic jack by calling snd_jack_new function. In that function, a new sound device is created representing the jack device. But in the particular case of jacks not belonging to a codec, they never get registered. A jack device in a codec driver gets registered because codec when initializing the SoC card (snd_soc_init_card) registers the sound card, and the sound card registers all sound devices added to it, among them the jack device. In the machine driver, the soc jack is created after the codec has been initialized and consequently all sound devices attached to the card (till that point) are already registered. Because of that, the jack device is created but not registered. I'm registering the jack device after it's created (in function snd_jack_new) and now I get the sys/class entry for the headset jack in my machine driver. What do you think about that solution? -Misa

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 12 Mar '09

12 Mar '09

The following changes since commit 3c081384007332ab95c8d8753f47719d6646bcb2: Mark Brown (1): Merge branch 's3c-iis-header' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 David Brownell (1): ASoC: buildfix for OSK Eric Miao (1): ASoC: make ops a pointer in 'struct snd_soc_dai' Mark Brown (6): ASoC: Add initial driver for the WM8400 CODEC ASoC: Remove version display from WM8580 driver ASoC: Fix formats for s3c24xx-i2s register prints ASoC: Merge dai_ops factor out [ARM] Revert futher extraneous changes from the S3C header move Merge branch 's3c-iis-header' into for-2.6.30 arch/arm/mach-s3c2410/include/mach/io.h | 2 +- arch/arm/plat-s3c24xx/clock-dclk.c | 1 - include/linux/mfd/wm8400-audio.h | 1 + include/sound/soc-dai.h | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 33 +- sound/soc/au1x/psc-ac97.c | 10 +- sound/soc/au1x/psc-i2s.c | 12 +- sound/soc/blackfin/bf5xx-i2s.c | 14 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ak4104.c | 10 +- sound/soc/codecs/ak4535.c | 14 +- sound/soc/codecs/cs4270.c | 14 +- sound/soc/codecs/ssm2602.c | 20 +- sound/soc/codecs/tlv320aic23.c | 18 +- sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 14 +- sound/soc/codecs/uda134x.c | 18 +- sound/soc/codecs/uda1380.c | 42 +- sound/soc/codecs/wm8350.c | 20 +- sound/soc/codecs/wm8400.c | 1481 +++++++++++++++++++++++++++++++ sound/soc/codecs/wm8400.h | 62 ++ sound/soc/codecs/wm8510.c | 16 +- sound/soc/codecs/wm8580.c | 34 +- sound/soc/codecs/wm8728.c | 12 +- sound/soc/codecs/wm8731.c | 18 +- sound/soc/codecs/wm8750.c | 14 +- sound/soc/codecs/wm8753.c | 90 +- sound/soc/codecs/wm8900.c | 16 +- sound/soc/codecs/wm8903.c | 18 +- sound/soc/codecs/wm8971.c | 14 +- sound/soc/codecs/wm8990.c | 18 +- sound/soc/codecs/wm9705.c | 8 +- sound/soc/codecs/wm9712.c | 14 +- sound/soc/codecs/wm9713.c | 40 +- sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/fsl/fsl_ssi.c | 18 +- sound/soc/fsl/mpc5200_psc_i2s.c | 20 +- sound/soc/omap/omap-mcbsp.c | 20 +- sound/soc/omap/osk5912.c | 12 +- sound/soc/pxa/pxa-ssp.c | 65 +- sound/soc/pxa/pxa2xx-ac97.c | 13 +- sound/soc/pxa/pxa2xx-i2s.c | 18 +- sound/soc/s3c24xx/s3c2412-i2s.c | 8 +- sound/soc/s3c24xx/s3c2443-ac97.c | 18 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 32 +- sound/soc/sh/ssi.c | 30 +- sound/soc/soc-core.c | 102 ++- 49 files changed, 2047 insertions(+), 450 deletions(-) create mode 100644 sound/soc/codecs/wm8400.c create mode 100644 sound/soc/codecs/wm8400.h

2 1

[alsa-devel] Problem compiling soc - imx31 and wm8753
by Marcelo Utikawa da Fonseca 12 Mar '09

12 Mar '09

Hi! I am new here and do not know if I am posting correctly, please understand! :-) I knew about this list in the wolfsonmicro website and I hope someone can help me... I am facing a problem while compiling the sound support for imx31ads board. I got the kernel from wolfsonmicro using the following command: git clone git://opensource.wolfsonmicro.com/linux-2.6-asoc linux-2.6-asoc After this I just copy the arch/arm/configs/mx31ads_defconfig to .config and run: make ARCH=arm CROSS_COMPILE=arm-none-linux-gnueabi- menuconfig I selected Alsa support, SoC, Support for i.MX CPU and Support for i.MX31 - WM8753. After this I run: make ARCH=arm CROSS_COMPILE=arm-none-linux-gnueabi- zImage and I receive this error message: CC sound/soc/imx/imx-ssi.o In file included from sound/soc/imx/imx-ssi.c:29: include/sound/driver.h:1:2: warning: #warning "This file is deprecated" sound/soc/imx/imx-ssi.c:34:26: error: asm/arch/dma.h: No such file or directory sound/soc/imx/imx-ssi.c:35:28: error: asm/arch/clock.h: No such file or directory sound/soc/imx/imx-ssi.c:37:26: error: asm/hardware.h: No such file or directory In file included from sound/soc/imx/imx-ssi.c:39: sound/soc/imx/imx-ssi.h:10:31: error: asm/arch/hardware.h: No such file or directory In file included from sound/soc/imx/imx-ssi.c:39: sound/soc/imx/imx-ssi.h:213: error: array type has incomplete element type In file included from sound/soc/imx/imx-ssi.c:40: sound/soc/imx/imx31-pcm.h:62: error: expected specifier-qualifier-list before 'dma_channel_params' sound/soc/imx/imx31-pcm.h:65: error: array type has incomplete element type sound/soc/imx/imx-ssi.c:72: error: unknown field 'params' specified in initializer sound/soc/imx/imx-ssi.c:72: error: extra brace group at end of initializer sound/soc/imx/imx-ssi.c:72: error: (near initialization for 'imx_ssi1_pcm_stereo_out0') sound/soc/imx/imx-ssi.c:74: error: 'emi_2_per' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:76: error: 'SSI1_BASE_ADDR' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:77: error: 'DMA_REQ_SSI1_TX1' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:78: error: 'SSI' undeclared here (not in a function) sound/soc/imx/imx-ssi.c:79: warning: excess elements in struct initializer sound/soc/imx/imx-ssi.c:79: warning: (near initialization for 'imx_ssi1_pcm_stereo_out0') sound/soc/imx/imx-ssi.c:84: error: unknown field 'params' specified in initializer sound/soc/imx/imx-ssi.c:84: error: extra brace group at end of initializer sound/soc/imx/imx-ssi.c:84: error: (near initialization for 'imx_ssi1_pcm_stereo_out1') ... ... ... sound/soc/imx/imx-ssi.c:836: error: field name not in record or union initializer sound/soc/imx/imx-ssi.c:836: error: (near initialization for 'imx_ssi_pcm_dai') sound/soc/imx/imx-ssi.c:837: error: field name not in record or union initializer sound/soc/imx/imx-ssi.c:837: error: (near initialization for 'imx_ssi_pcm_dai') sound/soc/imx/imx-ssi.c:838: error: field name not in record or union initializer sound/soc/imx/imx-ssi.c:838: error: (near initialization for 'imx_ssi_pcm_dai') sound/soc/imx/imx-ssi.c:841: warning: type defaults to 'int' in declaration of 'imx_ssi_pcm_dai' make[3]: *** [sound/soc/imx/imx-ssi.o] Error 1 make[2]: *** [sound/soc/imx] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 I did not change nothing, just configure and compile... My gcc version is 4.1.2 from Freescale. What can be wrong? Could be this version of gcc a problem? I already tried to correct the missing include files but the other errors are still there after the corrections... Thanks in advance, Marcelo Utikawa da Fonseca --------------------------------------------- Tecnequip Tecnologia em Equipamentos Endereço/Address: Rua Juno, 527 Cidade/City: São Paulo Estado/State: SP País/Country: Brasil CEP/Postal Code: 03445-000 Fone/Phone: 55-11-20937199 FAX: 55-11-29412289

3 5

[alsa-devel] snd-soc-pxa-ssp I2C/FIFO issues
by pHilipp Zabel 11 Mar '09

11 Mar '09

Ok, I'm not sure how to handle this correctly. Magician needs stereo S16_LE input data from alsa, DAIFMT_DSP_A on the wire with a frame size of 16 bit (one frame for each channel) and thus 2-byte DMA transfers to the FIFO (currently called 'mono' DMA params). Zylonite's current code uses 32 bit frame size and 4-byte DMA transfers to emulate I2S (S16_LE,stereo). Network mode seems to work, so it could be changed to use two 16 bit slots instead with 2-byte DMA transfers (which would make Magician happy). But... Daniel's board needs very strange settings because network mode doesn't work, so I guess he can't use 2-byte DMA transfers. Is that correct? If we can't find a way to have all machines use the mono DMA params, there has to be some kind of interface to set the width of the data in the FIFO independently from the Alsa format. How should that interface look like? regards Philipp

4 11

[PATCH] pxa-ssp: switch from network mode to PSP
by Daniel Mack 11 Mar '09

11 Mar '09

This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/pxa-ssp.c | 44 +++++++++++++++++++++++++++++++++----------- 1 files changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 52d97c4..3cde686 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -558,18 +558,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -655,33 +654,56 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && + (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else + sspsp |= SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- 1.6.2

1 0

[PATCH] pxa-ssp: switch from network mode to PSP
by Daniel Mack 11 Mar '09

11 Mar '09

This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/pxa-ssp.c | 44 +++++++++++++++++++++++++++++++++----------- 1 files changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 52d97c4..3cde686 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -558,18 +558,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -655,33 +654,56 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && + (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else + sspsp |= SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- 1.6.2

1 0

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 11 Mar '09

11 Mar '09

The following changes since commit 0a4e1c9069b3cc2a9e5b57898ac9be94314e8efa: Takashi Iwai (1): Merge branch 'for-2.6.30' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Dooks (1): ASoC: Fix Samsung S3C2412_IISMOD_SDF_{MSB,LSB} definitions Daniel Mack (1): ASoC: don't touch pxa-ssp registers when stream is running Hugo Villeneuve (2): ALSA: ASoC: Davinci: Replaced DAI format RIGHT_J by DSP_B for SFFSDR ALSA: ASoC: Davinci: Updated sffsdr_hw_params() function to new format Mark Brown (3): ASoC: Convert PXA AC97 driver to probe with the platform device Merge commit 'takashi/topic/asoc' into for-2.6.30 ASoC: Fix up merge with the ARM tree arch/arm/mach-shark/include/mach/io.h | 1 + arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 4 +- sound/soc/davinci/davinci-sffsdr.c | 20 ++++++----- sound/soc/pxa/pxa-ssp.c | 11 ++++++ sound/soc/pxa/pxa2xx-ac97.c | 38 ++++++++++++++++++-- 5 files changed, 59 insertions(+), 15 deletions(-)

3 3

[alsa-devel] [PATCH] ASoC: Complete Beagleboard support
by Steve Sakoman 11 Mar '09

11 Mar '09

Commit dc06102a0c8b5aa0dd7f9a40ce241e793c252a87 in the asoc tree did not include the necessary Kconfig and Makefile changes. This patch completes the support for Beagleboard Signed-off-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/Kconfig | 10 ++++++++++ sound/soc/omap/Makefile | 2 ++ 2 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4f7f040..ccd8973 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA select SND_SOC_TWL4030 help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_OMAP_SOC_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Beagleboard. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 76fedd9..0c9e4ac 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.5.6.3

4 8

[alsa-devel] [patch 0/9] S3C24XX/S3C64XX updates
by Ben Dooks 11 Mar '09

11 Mar '09

This is a set of patches for 2.6.30 to add more s3c24xx machine support and support for the S3C64XX series of SoC. It would be very useful to get the header moves merged as soon as possible, to help clear out the old ARM include directories (and make it easier for the merges from other s3c branches that need to take place in the merge window). As such, they're only moves and should not affect the actual build. -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

3 23

[alsa-devel] [PATCH] ALSA: hda - Add model quirk for Sony VAIO VGN-NR21E
by Ozan Caglayan 11 Mar '09

11 Mar '09

One of our users reported that its laptop doesn't mute the internal speakers when a jack is inserted into headphones without model=hippo. Here's the alsa-info output for reference: http://www.alsa-project.org/db/?f=8dae188ec2db110b9e6804c60f4659f0cb606c62 Signed-off-by: Ozan Caglayan <ozan(a)pardus.org.tr> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b794cba..6e79770 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10874,6 +10874,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), + SND_PCI_QUIRK(0x104d, 0x902d, "Sony VAIO VGN-NR21E", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", -- 1.6.2

2 4

[alsa-devel] Accuracy of substream->ops->pointer?
by Timur Tabi 11 Mar '09

11 Mar '09

During capture, my hardware can take up to two milliseconds for the DMA to start transferring data to the buffer. This is because the internal FIFO needs to fill up before DMA starts. Today, my .trigger function (during a START request) just waits until the FIFO is full before returning control back to ALSA. This seems to work, except that I wait by using a busy-loop. As you can imagine, a 2ms busy loop is not a good idea. An alternative idea is to have my .pointer function return 0 if DMA has not yet started. This appears to work, but I want to make sure it's a good idea. My concern is that I might confuse ALSA because it is expecting some data in the buffer after capture has started. Is my approach correct? -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] Fix audio for Lyrtech SFFSDR board
by Hugo Villeneuve 10 Mar '09

10 Mar '09

These patches make the audio work again on the SFFSDR

4 18

[alsa-devel] PXA SSP and external clocked I2S again
by Daniel Mack 10 Mar '09

10 Mar '09

Hi, as denoted yesterday, we've spend some more time on the PXA/SSP/I2S issue and would like to share conclusions about out utterly frustrating trial-and-error sessions during the last days, especially as reference for everyone who tries to get a similar setup running. The situation on our board is: we have a tunable master clock generator, a CS4270 codec and an PXA303 connected to each other. In order to use the master clock, the PXA needs to be set to an external clock mode and for the CS4270 to operate properly, we need to provide a full 64 bits I2S stream, even though not all of the data bits (in fact, currently only 16 of them per channel) carry data. The above thing was not possible with the pxa-ssp's current approach as it entirely relyed on the network mode and the time slots mechanism which is - according to the datasheets - supposed to do exactly this, but which simply doesn't work at all. It might work for existing boards without our constrains, but that's more or less due to coincidence. Hence, we switched over to non-network mode and fiddled around with the PSP bits a lot until we found a mode that fits our needs, at least as long as we let the PXA be master in the game (which we did for test purposes). As soon as the clock direction changes (codec takes over control for LRCLK and bitclk), the PSP feature fails badly again. The final solution is now to never ever set the PXA to a real slave mode (DAIFMT_CBM_CFM) but only provide the master clock to its input pin (SSPEXTCLK), set the clock config to SSP_CLK_EXT and let the PXA derive the other clocks from that one internally. Another thing is: ALSA doesn't currently provide a way to configure a DAI format where the I2S protocol has more clock bits than data bits. One of the upcoming series of 4 patches adds them. Best regards, Daniel

4 27

[PATCH 2/2] pxa-ssp: don't touch registers if stream is running
by Daniel Mack 10 Mar '09

10 Mar '09

In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all. If there would be but the SSP port is in use already, bail out. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Philipp Zabel <philipp.zabel(a)gmail.com> Cc: Liam Girdwood <lrg(a)kernel.org> --- sound/soc/pxa/pxa-ssp.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 1b3a81c..569c0a6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -525,6 +525,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, u32 sscr1; u32 sspsp; + /* check if we need to change anything at all */ + if (priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + dev_err(&ssp->pdev->dev, + "can't change hardware dai format: stream is in use"); + return -EINVAL; + } + /* reset port settings */ sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); -- 1.6.2

1 0

[PATCH 1/2] pxa-ssp: switch from network mode to PSP
by Daniel Mack 10 Mar '09

10 Mar '09

This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/pxa-ssp.c | 56 ++++++++++++++++++++++++++++++---------------- 1 files changed, 36 insertions(+), 20 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 7fc13f0..1b3a81c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -45,6 +45,7 @@ struct ssp_priv { struct ssp_dev dev; unsigned int sysclk; + unsigned int scr_div; int dai_fmt; #ifdef CONFIG_PM struct ssp_state state; @@ -281,11 +282,13 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_priv *priv, u32 div) { + struct ssp_dev *dev = &priv->dev; struct ssp_device *ssp = dev->ssp; u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + priv->scr_div = div; ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); } @@ -327,7 +330,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(priv, 1); sscr0 |= SSCR0_ACS; break; default: @@ -388,7 +391,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(priv, div); break; default: return -ENODEV; @@ -547,18 +550,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -644,37 +646,51 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if ((priv->scr_div == 4) && (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else + sspsp |= SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; -- 1.6.2

1 0

[alsa-devel] [PATCH] [SOUND] Fixes recording gain control
by Viral Mehta 10 Mar '09

10 Mar '09

Sorry for resending it again and again. This is since submitting a patch first time. Please never mind. Consider the below thing. Have tested two usb audio mic devices. At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not set for MIC (slot 7). So, the same should not be checked when an application tries to do gain control for audio recording devices. Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently. Verified with a simple application which opens /dev/dsp for recording and /dev/mixer for volume control. Signed-off-by: Viral Mehta <viral.mehta(a)einfochips.com> --- sound/core/oss/mixer_oss.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) --- sound/core/oss/mixer_oss.org.c 2009-03-10 18:37:49.000000000 +0530 +++ sound/core/oss/mixer_oss.c 2009-03-10 18:37:07.000000000 +0530 @@ -692,6 +692,8 @@ static int snd_mixer_oss_put_volume1(str snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) { + snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) { -- _____________________________________________________________________ Disclaimer: This e-mail message and all attachments transmitted with it are intended solely for the use of the addressee and may contain legally privileged and confidential information. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution, copying, or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately by replying to this message and please delete it from your computer. Any views expressed in this message are those of the individual sender unless otherwise stated.Company has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. __________________________________________________________________________

2 1

[alsa-devel] [PATCH] [SOUND] Fixes recording gain control
by Viral Mehta 10 Mar '09

10 Mar '09

Have tested two usb audio mic devices. At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not set for MIC (slot 7). So, the same should not be checked when an application tries to do gain control for audio recording devices. Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently. Verified with a simple application which opens /dev/dsp for recording and /dev/mixer for volume control. Signed-off-by: Viral Mehta <viral.mehta(a)einfochips.com> --- sound/core/oss/mixer_oss.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 4690b8b..919c7eb 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -690,7 +690,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); - if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right); -- 1.6.0.2.GIT -- _____________________________________________________________________ Disclaimer: This e-mail message and all attachments transmitted with it are intended solely for the use of the addressee and may contain legally privileged and confidential information. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution, copying, or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately by replying to this message and please delete it from your computer. Any views expressed in this message are those of the individual sender unless otherwise stated.Company has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. __________________________________________________________________________

1 0

Re: [alsa-devel] [PATCH] [SOUND] recording gain control
by Clemens Ladisch 10 Mar '09

10 Mar '09

(CC alsa-devel) Viral Mehta wrote: > Hi, > > I am trying to do gain control while RECORDING for my Audio Mic device. > > My application opens /dev/mixer device and calls ioctl(fdmixer, > MIXER_WRITE(SOUND_MIXER_MIC), ...) > > But, the call fails. It traces to sound/core/oss/mixer_oss.c file and > snd_mixer_oss_put_volume1() function. > It never falls in "if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)" > as condition "if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) " > comes as false. > > I think there is a bug in kernel and I think it should be like as below, > > --- sound/core/oss/mixer_oss.c 2008-12-03 13:24:02.000000000 +0530 > +++ sound/core/oss/mixer_oss.c 2009-03-09 16:22:06.548766896 +0530 > @@ -688,7 +688,7 @@ static int snd_mixer_oss_put_volume1(str > > if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { > snd_mixer_oss_put_volume1_vol(fmixer, pslot, > slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); > - if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) > + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) { > snd_mixer_oss_put_volume1_vol(fmixer, pslot, > slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); > } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) { > snd_mixer_oss_put_volume1_vol(fmixer, pslot, > slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right); Your mailer mangled the patch; see linux/Documentation/email-clients.txt. Best regards, Clemens

3 3

[alsa-devel] [PATCH 0/5] Blackfin ASoC fixing and updates
by Bryan Wu 10 Mar '09

10 Mar '09

Hi Mark, We have some patches for Blackfin ASoC updates. Please take a look. Thanks a lot -Bryan

4 17

[alsa-devel] [Re: Crash when snd_pcm_open is called from a thread with "pulse" as playback]
by Nicolas.Castagne＠imag.fr 10 Mar '09

10 Mar '09

Hi folks, I'd need the help of some alsa developpers regarding the bug report : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4426 We've made a few progress on it, but we are stuck again. Basically, it seems that calling snd_config_update_free_global() from a pthread causes the program to crash when the thread terminates. Questions are: Should config_update_free_global NOT be called from a thread ? In other words, is the following code valid (though dumb :=)) ? Or is there some reason for it to crash ? ********************************** #include <pthread.h> #include <alsa/asoundlib.h> void * threadWork(void *arg) { snd_pcm_t *playback_handle; if (snd_pcm_open (&playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0) fprintf (stderr, "error\n"; snd_pcm_close(playback_handle); snd_config_update_free_global(); return NULL; } int main() { pthread_attr_t attr; pthread_attr_init(&attr); pthread_t threadid; pthread_create( &threadid, &attr, threadWork, NULL); return EXIT_SUCCESS; } ********************************** I thank you much, Nicolas ---------------------------- Message original ---------------------------- Objet: Crash when snd_pcm_open is called from a thread with "pulse" as playback De: castagne(a)imag.fr Date: Mer 4 mars 2009 8:20 À: alsa-devel(a)alsa-project.org -------------------------------------------------------------------------- Hi folks, I hope I post on the correct mail list... I've read somewhere that I should report bugs on a list, and not only on the Tracker. That's the aim of this message. On all my Fedora 10 machines, whenever snd_pcm_open has been called from a pthread using "pulse" as playback then the program crashes when the thread terminates. Crash in __nptl_deallocate_tsd However, AFAIK, one should be able to call snd_pcm_open from a thread. The full bug report is there, with a very simple sample code : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4426 I think this bug relates directly to alsa. I hope this report is useful. Best- Nicolas

3 5

[alsa-devel] [PATCH] S3C: Fix S3C2412_IISMOD_SDF_{MSB, LSB} definitions
by Ben Dooks 09 Mar '09

09 Mar '09

The definitions of S3C2412_IISMOD_SDF_MSB and S3C2412_IISMOD_SDF_LSB are incorrect, being the same S3C2412_IISMOD_SDF_IIS which is the only correct one in this series. Signed-off-by: Ben Dooks <ben-linux(a)fluff.org> --- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index 25d4058..e5988b4 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -44,8 +44,8 @@ #define S3C2412_IISMOD_LR_LLOW (0 << 7) #define S3C2412_IISMOD_LR_RLOW (1 << 7) #define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (0 << 5) -#define S3C2412_IISMOD_SDF_LSB (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) #define S3C2412_IISMOD_SDF_MASK (3 << 5) #define S3C2412_IISMOD_RCLK_256FS (0 << 3) #define S3C2412_IISMOD_RCLK_512FS (1 << 3) -- 1.5.6.5

2 1

[alsa-devel] Crash when snd_pcm_open is called from a thread with "pulse" as playback
by Nicolas.Castagne＠imag.fr 09 Mar '09

09 Mar '09

Hi folks, I hope I post on the correct mail list... I've read somewhere that I should report bugs on a list, and not only on the Tracker. That's the aim of this message. On all my Fedora 10 machines, whenever snd_pcm_open has been called from a pthread using "pulse" as playback then the program crashes when the thread terminates. Crash in __nptl_deallocate_tsd However, AFAIK, one should be able to call snd_pcm_open from a thread. The full bug report is there, with a very simple sample code : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4426 I think this bug relates directly to alsa. I hope this report is useful. Best- Nicolas

2 2

[alsa-devel] [PATCH 1/1] echomixer, fix vmixer GUI
by Giuliano Pochini 09 Mar '09

09 Mar '09

This patch fixes several bugs in the vmixer GUI and removes the "misc" button from the main window when the card has no global controls. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-tools-1.0.19/echomixer/echomixer.c__orig 2009-02-27 22:51:47.000000000 +0100 +++ alsa-tools-1.0.19/echomixer/echomixer.c 2009-03-01 16:29:42.000000000 +0100 @@ -129,38 +129,8 @@ struct mixel { snd_ctl_t *ctlhandle; -#if __GNUC__ == 3 // gcc 2.x doesn't like unnamed unions inside structures - -struct mixerControl_s { - union { // Currently selected channels - int vchannel; - int input; - }; - union { // Number of channels - int vchannels; - int inputs; - }; - int output, outputs; - int id; - GtkWidget *window; - GtkWidget *volume[ECHO_MAXAUDIOOUTPUTS]; - GtkWidget *label[ECHO_MAXAUDIOOUTPUTS]; - GtkObject *adj[ECHO_MAXAUDIOOUTPUTS]; - GtkWidget *outsel[ECHO_MAXAUDIOOUTPUTS]; - union { - GtkWidget *inpsel[ECHO_MAXAUDIOINPUTS]; - GtkWidget *vchsel[ECHO_MAXAUDIOOUTPUTS]; - }; - struct mixel mixer[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS]; -} mixerControl, vmixerControl; - -#else - struct mixerControl_s { - int vchannel; - int input; - int vchannels; - int inputs; + int input, inputs; int output, outputs; int id; GtkWidget *window; @@ -173,9 +143,6 @@ struct mixerControl_s { struct mixel mixer[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS]; } mixerControl, vmixerControl; -#endif - - struct VolumeControl_s { int input, output; // Currently selected channels int inputs, outputs; @@ -677,8 +644,8 @@ gint DrawMixer(gpointer unused) { gdk_draw_rectangle(Mixpixmap, gc, TRUE, XCELLTOT*(mixerControl.output+1), YCELLTOT*mixerControl.input, XCELLTOT, Mixheight); if (vmixerId) { gdk_gc_set_foreground(gc, &Hilight2); - gdk_draw_rectangle(Mixpixmap, gc, TRUE, 0, YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.vchannel), XCELLTOT*(vmixerControl.output+1), YCELLTOT); - gdk_draw_rectangle(Mixpixmap, gc, TRUE, XCELLTOT*(vmixerControl.output+1), YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.vchannel), XCELLTOT, Mixheight); + gdk_draw_rectangle(Mixpixmap, gc, TRUE, 0, YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.input), XCELLTOT*(vmixerControl.output+1), YCELLTOT); + gdk_draw_rectangle(Mixpixmap, gc, TRUE, XCELLTOT*(vmixerControl.output+1), YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.input), XCELLTOT, Mixheight); } // Draw the grid @@ -720,7 +687,7 @@ gint DrawMixer(gpointer unused) { // Draw vchannels levels and peaks (Vmixer cards only) if (vmixerId) { - for (i=0; i<vmixerControl.vchannels; i++) + for (i=0; i<vmixerControl.inputs; i++) DrawBar(0, i+GMixerSection.VmixerFirst, VirLevel[i], VirPeak[i], DONT_DRAW); } @@ -739,7 +706,7 @@ gint DrawMixer(gpointer unused) { // Draw vmixer elements (Vmixer cards only) if (vmixerId) { for (o=0; o<GMixerSection.Outputs; o++) - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { dB=Add_dB(vmixerControl.mixer[o][i].Gain, VirLevel[i]); DrawBar(o+1, i+GMixerSection.VmixerFirst, dB, DONT_DRAW, vmixerControl.mixer[o][i].Gain); } @@ -945,7 +912,7 @@ static gint Gmixer_button_press(GtkWidge } else if (GMixerRow>=GMixerSection.VmixerFirst && GMixerRow<=GMixerSection.VmixerLast) { if (GMixerColumn!=vmixerControl.output) gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(vmixerControl.outsel[GMixerColumn]), TRUE); - if (GMixerRow!=vmixerControl.vchannel) + if (GMixerRow!=vmixerControl.input) gtk_widget_grab_focus(GTK_WIDGET(vmixerControl.volume[GMixerRow-GMixerSection.VmixerFirst])); } @@ -967,13 +934,13 @@ static gint Gmixer_button_press(GtkWidge return TRUE; // See the note above - if (GMixerRow<GMixerSection.Inputs) { + if (GMixerRow<GMixerSection.VmixerFirst) { if (GMixerRow!=mixerControl.input) gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(mixerControl.inpsel[GMixerRow]), TRUE); if (GMixerColumn!=mixerControl.output) gtk_widget_grab_focus(GTK_WIDGET(mixerControl.volume[GMixerColumn])); } else if (GMixerRow>=GMixerSection.VmixerFirst && GMixerRow<=GMixerSection.VmixerLast) { - if (GMixerRow!=vmixerControl.vchannel) + if (GMixerRow!=vmixerControl.input+GMixerSection.VmixerFirst) gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(vmixerControl.vchsel[GMixerRow-GMixerSection.VmixerFirst]), TRUE); if (GMixerColumn!=vmixerControl.output) gtk_widget_grab_focus(GTK_WIDGET(vmixerControl.volume[GMixerColumn])); @@ -1035,11 +1002,11 @@ static gint Gmixer_motion_notify(GtkWidg gtk_adjustment_set_value(GTK_ADJUSTMENT(mixerControl.adj[mixerControl.output]), (gfloat)val); #endif } else if (GMixerRow>=GMixerSection.VmixerFirst && GMixerRow<=GMixerSection.VmixerLast) { - val=INVERT(vmixerControl.mixer[vmixerControl.output][vmixerControl.vchannel].Gain); + val=INVERT(vmixerControl.mixer[vmixerControl.output][vmixerControl.input].Gain); val+=y-mouseY; mouseY=y; #ifdef REVERSE - gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[vmixerControl.vchannel]), (gfloat)val); + gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[vmixerControl.input]), (gfloat)val); #else gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[vmixerControl.output]), (gfloat)val); #endif @@ -1259,7 +1226,7 @@ void Vmixer_volume_changed(GtkWidget *wi v=channel; o=vmixerControl.output; #else - v=vmixerControl.vchannel; + v=vmixerControl.input; o=channel; #endif @@ -1281,8 +1248,8 @@ void Vmixer_volume_changed(GtkWidget *wi void Vmixer_volume_clicked(GtkWidget *widget, gpointer ch) { #ifdef REVERSE - vmixerControl.vchannel=(int)(long)ch; - UI_DEBUG(("Vmixer_volume_clicked vch=%d\n",vmixerControl.vchannel)); + vmixerControl.input=(int)(long)ch; + UI_DEBUG(("Vmixer_volume_clicked vch=%d\n",vmixerControl.input)); #else vmixerControl.output=(int)(long)ch; UI_DEBUG(("Vmixer_volume_clicked out=%d\n",vmixerControl.output)); @@ -1306,7 +1273,7 @@ void Vmixer_output_selector_clicked(GtkW snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); - for (c=vmixerControl.vchannels-1; c>=0; c--) { + for (c=vmixerControl.inputs-1; c>=0; c--) { val=INVERT(vmixerControl.mixer[vmixerControl.output][c].Gain); gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[c]), (gfloat)val); } @@ -1319,16 +1286,16 @@ void Vmixer_vchannel_selector_clicked(Gt snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *control; - if (vmixerControl.vchannel==(int)(long)ch) + if (vmixerControl.input==(int)(long)ch) return; - vmixerControl.vchannel=(int)(long)ch; + vmixerControl.input=(int)(long)ch; - UI_DEBUG(("Vmixer_selector_clicked vch=%d\n",vmixerControl.vchannel)); + UI_DEBUG(("Vmixer_selector_clicked vch=%d\n",vmixerControl.input)); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); for (c=vmixerControl.outputs-1; c>=0; c--) { - val=INVERT(vmixerControl.mixer[c][vmixerControl.vchannel].Gain); + val=INVERT(vmixerControl.mixer[c][vmixerControl.input].Gain); gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[c]), (gfloat)val); } } @@ -1722,7 +1689,7 @@ int OpenControls(const char *card, const vmixerId=vmixerControl.id=numid; CTLID_DEBUG(("First Vmixer id=%d\n", vmixerId)); vmixerControl.outputs=snd_ctl_elem_info_get_dimension(info, 0); - vmixerControl.vchannels=snd_ctl_elem_info_get_dimension(info, 1); + vmixerControl.inputs=snd_ctl_elem_info_get_dimension(info, 1); } } else if (!strcmp("PCM Playback Volume", snd_ctl_elem_id_get_name(id))) { pcmoutId=pcmoutControl.id=numid; @@ -1808,7 +1775,7 @@ int OpenControls(const char *card, const #ifndef REAL vmixerId=1000; -vmixerControl.vchannels=12; +vmixerControl.inputs=12; vmixerControl.outputs=mixerControl.outputs=nLOut=10; metersStreams=3; metersNumber=16; @@ -1828,8 +1795,8 @@ printf("nIn=%d fdIn=%d nLOut=%d nPOut=%d printf("** Warning - Vmixer cards without LineOut volume control are not supported !\n"); if (vmixerId) { - if (vmixerControl.vchannels!=nPOut || vmixerControl.outputs!=nLOut) { - printf("** Error - vmixer/channels mismatch: vmp=%d npo=%d vmo=%d nlo=%d !!\n", vmixerControl.vchannels, nPOut, vmixerControl.outputs, nLOut); + if (vmixerControl.inputs!=nPOut || vmixerControl.outputs!=nLOut) { + printf("** Error - vmixer/channels mismatch: vmp=%d npo=%d vmo=%d nlo=%d !!\n", vmixerControl.inputs, nPOut, vmixerControl.outputs, nLOut); return(1); } } @@ -1850,7 +1817,7 @@ printf("nIn=%d fdIn=%d nLOut=%d nPOut=%d if (vmixerId) for (o=0, numid=vmixerId; o<vmixerControl.outputs; o++) { - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { vmixerControl.mixer[o][i].id=numid++; } } @@ -1931,10 +1898,10 @@ printf("components = %s\n", snd_ctl_card Mixerw_geom.st=NOPOS; Vmixerw_geom.st=NOPOS; VUwindow=GMwindow=0; - GMixerSection.Inputs=fdIn+2; // The correct value is set by Digital_mode_activate() - GMixerSection.Outputs=fdOut+2; + GMixerSection.Inputs=nIn; // The correct value is set by Digital_mode_activate() + GMixerSection.Outputs=nLOut; GMixerSection.VmixerFirst=nIn; - GMixerSection.VmixerLast=nIn+vmixerControl.vchannels-1; + GMixerSection.VmixerLast=nIn+vmixerControl.inputs-1; GMixerSection.LineOut=GMixerSection.VmixerLast+1; // Read current mixer setting. @@ -2523,7 +2490,7 @@ printf("components = %s\n", snd_ctl_card gtk_widget_show(hbox); gtk_container_add(GTK_CONTAINER(frame), hbox); - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { vbox=gtk_vbox_new(FALSE, 0); gtk_widget_show(vbox); gtk_container_add(GTK_CONTAINER(hbox), vbox); @@ -2582,7 +2549,7 @@ printf("components = %s\n", snd_ctl_card gtk_container_add(GTK_CONTAINER(frame), vbsel); bgroup=0; - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { sprintf(str, "V%d", i); if (i) bgroup=gtk_radio_button_group(GTK_RADIO_BUTTON(vmixerControl.vchsel[i-1])); @@ -2626,7 +2593,7 @@ printf("components = %s\n", snd_ctl_card gtk_box_pack_start(GTK_BOX(vbox), vmixerControl.label[i], FALSE, FALSE, 0); } gtk_widget_set_usize(GTK_WIDGET(vmixerControl.volume[0]), 0, 170); // Set minimum y size - vmixerControl.vchannel=-1; + vmixerControl.input=-1; Vmixer_vchannel_selector_clicked(0, 0); #endif } @@ -2693,13 +2660,15 @@ printf("components = %s\n", snd_ctl_card gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(button), TRUE); // Misc controls button - button=gtk_toggle_button_new_with_label("Misc"); - gtk_widget_show(button); - gtk_box_pack_start(GTK_BOX(hbox), button, TRUE, TRUE, 1); - gtk_signal_connect(GTK_OBJECT(button), "toggled", ToggleWindow, (gpointer)Miscwindow); - Miscw_geom.toggler=button; - if (Miscw_geom.st==1) - gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(button), TRUE); + if (p4InId || p4OutId || phantomId || (dmodeId && ndmodes>1) || (clocksrcId && nclocksrc>1) || (spdifmodeId && nspdifmodes>1)) { + button=gtk_toggle_button_new_with_label("Misc"); + gtk_widget_show(button); + gtk_box_pack_start(GTK_BOX(hbox), button, TRUE, TRUE, 1); + gtk_signal_connect(GTK_OBJECT(button), "toggled", ToggleWindow, (gpointer)Miscwindow); + Miscw_geom.toggler=button; + if (Miscw_geom.st==1) + gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(button), TRUE); + } if (mixerId) { // Graphical mixer button -- Giuliano.

2 1

Re: [alsa-devel] Timer instability
by Takashi Iwai 09 Mar '09

09 Mar '09

At Wed, 25 Feb 2009 16:56:47 +0000, Clive Messer wrote: > > On Wednesday 25 Feb 2009 16:39:28 you wrote: > > At Wed, 25 Feb 2009 16:24:18 +0000, > > > > Clive Messer wrote: > > > On Wednesday 25 Feb 2009 15:51:33 you wrote: > > > > At Wed, 25 Feb 2009 15:11:44 +0000, > > > > > > > > Clive Messer wrote: > > > > > 36192 36191 100068 3 0 0 1 2 > > > > > 36195 36194 100090 2 0 0 1 2 > > > > > 36198 36198 100113 1 0 0 1 2 > > > > > 36207 36206 63854 2816 1600 0 1 3 > > > > > 36212 36211 63854 2815 1601 4 1 3 > > > > > 36216 36216 63854 2814 1602 4 1 3 > > > > > 36221 36220 63854 2813 1603 4 1 3 > > > > > 36225 36225 101224 4460 -44 4 1 3 > > > > > > > > This is pretty weird. Is this behavior always reproducible? > > > > > > Nope. I get different results every time I run it. > > > > > > > What shows with the patch below to alsa-lib? > > > > > > Here's one (attached) I just did with the new patch that shows the crazy > > > high numbers again. > > > hwsync: 2080(0), 6496(0) > > > 70758 70758 47142 0 4416 0 0 3 > > > hwsync: 1472(0), 6496(0) > > > 70763 70762 104573379136040 4971973988617026976 -4971973988617022560 4 1 > > >3 > > > > The problem is that hwptr goes backward here. But it didn't happen in > > this hwsync call but sometime before that. > > > > Could you try the patch below instead? This will track all sync_ptr calls. > > > > > > Takashi > > > > --- > > diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c > > index e9ce092..ad8d4a5 100644 > > --- a/src/pcm/pcm_hw.c > > +++ b/src/pcm/pcm_hw.c > > @@ -131,6 +131,10 @@ struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm) > > static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int flags) > > { > > int err; > > + long old_hwptr, new_hwptr; > > + long old_applptr, new_applptr; > > + old_hwptr = *pcm->hw.ptr; > > + old_applptr = *pcm->appl.ptr; > > hw->sync_ptr->flags = flags; > > err = ioctl((hw)->fd, SNDRV_PCM_IOCTL_SYNC_PTR, (hw)->sync_ptr); > > if (err < 0) { > > @@ -138,6 +142,11 @@ static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int > > flags) SYSMSG("SNDRV_PCM_IOCTL_SYNC_PTR failed"); > > return err; > > } > > + new_hwptr = *pcm->hw.ptr; > > + new_applptr = *pcm->appl.ptr; > > + printf("sync_ptr1: %ld(%ld), %ld(%ld)\n", > > + new_hwptr, new_hwptr - old_hwptr, > > + new_applptr, new_applptr - old_applptr); > > return 0; > > } > > Sorry Takashi, > > I'm not a coder so I only know that it wont compile because there isn't a pcm > variable passed into that function - it's snd_pcm_hw_t *hw. Oops, sorry, pasted a wrong version. Here it is. thanks, Takashi --- diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index e9ce092..5211d1c 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -131,6 +131,10 @@ struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm) static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int flags) { int err; + long old_hwptr, new_hwptr; + long old_applptr, new_applptr; + old_hwptr = hw->sync_ptr->s.status.hw_ptr; + old_applptr = hw->sync_ptr->c.control.appl_ptr; hw->sync_ptr->flags = flags; err = ioctl((hw)->fd, SNDRV_PCM_IOCTL_SYNC_PTR, (hw)->sync_ptr); if (err < 0) { @@ -138,6 +142,11 @@ static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int flags) SYSMSG("SNDRV_PCM_IOCTL_SYNC_PTR failed"); return err; } + new_hwptr = hw->sync_ptr->s.status.hw_ptr; + new_applptr = hw->sync_ptr->c.control.appl_ptr; + printf("sync_ptr1: %ld(%ld), %ld(%ld)\n", + new_hwptr, new_hwptr - old_hwptr, + new_applptr, new_applptr - old_applptr); return 0; }

2 9

[alsa-devel] Bad page state in process 'rmmod' cs4281
by "Timo T.Hyrsylä" 09 Mar '09

09 Mar '09

Trying to make Alsa work on our embedded arm system with ixdp425 and CS4281. kernel is 2.6.20.3 and the Alsas (driver+lib+utils) latest 1.0.19 Other than the two below errors, the system seems to work ok. When removing the snd_cs4281 module, dmesg reports multiple errors about bad page <0>Bad page state in process 'rmmod' <0>page:c4007300 flags:0x00004000 mapping:00000000 mapcount:0 count:0 <0>Trying to fix it up, but a reboot is needed <0>Backtrace: <0>Bad page state in process 'rmmod' <0>page:c4007320 flags:0x00004000 mapping:00000000 mapcount:0 count:0 <0>Trying to fix it up, but a reboot is needed <0>Backtrace: I've managed to track down the error upto dma_free_coherent. snd_pcm_free > snd_pcm_lib_preallocate_free_for_all > snd_pcm_lib_preallocate_free > snd_pcm_lib_preallocate_dma_free > snd_dma_free_pages > snd_free_dev_pages > dma_free_coherent Another issue, possibly related, is that when running aplay with --mmap it will crash with: Internal error: Oops: f3 [#1] PC is at snd_pcm_mmap_data_nopage+0x10c/0x148 [snd_pcm] LR is at __handle_mm_fault+0x16c/0x8ac any helping ideas? -- Timo TH.

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 09 Mar '09

09 Mar '09

The following changes since commit f271fa28fbaf947d9c79f188dd149176da727dd5: Takashi Iwai (1): ASoC: Fix Kconfig dependency of CONFIG_SND_S3C24XX_SOC_JIVE_WM8750 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Daniel Mack (2): ASoC: bring cs4270 feature/limitations list in sync ASoC: Add a driver for AK4104 S/PDIF transmitter Hugo Villeneuve (1): ASoC: Davinci: Fix incorrect machine type for SFFSDR board Mark Brown (3): ASoC: Re-remove hand-rolled pr_debug() macros ASoC: Fix logging severity for some S3C error messages ASoC: Remove unneeded forward reference to WM8753 SPI implementation Timur Tabi (1): ASoC: Improve pause/unpause performance in Freescale 8610 drivers sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4104.c | 363 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4104.h | 7 + sound/soc/codecs/cs4270.c | 15 +- sound/soc/codecs/wm8753.c | 5 - sound/soc/davinci/Kconfig | 2 +- sound/soc/fsl/fsl_dma.c | 3 +- sound/soc/fsl/fsl_ssi.c | 23 ++- sound/soc/s3c24xx/neo1973_wm8753.c | 44 ++--- sound/soc/s3c24xx/s3c-i2s-v2.c | 49 ++--- sound/soc/s3c24xx/s3c2412-i2s.c | 12 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 49 ++--- sound/soc/s3c24xx/s3c24xx-pcm.c | 45 ++--- 14 files changed, 482 insertions(+), 141 deletions(-) create mode 100644 sound/soc/codecs/ak4104.c create mode 100644 sound/soc/codecs/ak4104.h

2 1

[alsa-devel] [PATCH] ASoC: Add a driver for AK4104 S/PDIF transmitter
by Daniel Mack 09 Mar '09

09 Mar '09

This adds a driver for the SPI connected AK4104 S/PDIF transmitter device. It's features are fairly simple, but as there is need to set up certain bits in the IEC958 information, this better goes into a real driver. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)sirena.org.uk> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4104.c | 367 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4104.h | 7 + 4 files changed, 380 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4104.c create mode 100644 sound/soc/codecs/ak4104.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 628a591..a1af311 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -14,6 +14,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 if I2C + select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 @@ -60,6 +61,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_AK4104 + tristate + config SND_SOC_AK4535 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3664cdc..4717c3c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,7 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o @@ -30,6 +31,7 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c new file mode 100644 index 0000000..8508a22 --- /dev/null +++ b/sound/soc/codecs/ak4104.c @@ -0,0 +1,367 @@ +/* + * AK4104 ALSA SoC (ASoC) driver + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * Copyright 2007-2009 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + */ + +#include <linux/module.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <linux/spi/spi.h> +#include <sound/asoundef.h> + +#include "ak4104.h" + +/* AK4104 registers addresses */ +#define AK4104_REG_CONTROL1 0x00 +#define AK4104_REG_RESERVED 0x01 +#define AK4104_REG_CONTROL2 0x02 +#define AK4104_REG_TX 0x03 +#define AK4104_REG_CHN_STATUS(x) ((x) + 0x04) +#define AK4104_NUM_REGS 10 + +#define AK4104_REG_MASK 0x1f +#define AK4104_READ 0xc0 +#define AK4104_WRITE 0xe0 +#define AK4104_RESERVED_VAL 0x5b + +/* Bit masks for AK4104 registers */ +#define AK4104_CONTROL1_RSTN (1 << 0) +#define AK4104_CONTROL1_PW (1 << 1) +#define AK4104_CONTROL1_DIF0 (1 << 2) +#define AK4104_CONTROL1_DIF1 (1 << 3) + +#define AK4104_CONTROL2_SEL0 (1 << 0) +#define AK4104_CONTROL2_SEL1 (1 << 1) +#define AK4104_CONTROL2_MODE (1 << 2) + +#define AK4104_TX_TXE (1 << 0) +#define AK4104_TX_V (1 << 1) + +#define DRV_NAME "ak4104" + +struct ak4104_private { + struct snd_soc_codec codec; + u8 reg_cache[AK4104_NUM_REGS]; +}; + +static int ak4104_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *reg_cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + for (i = 0; i < codec->reg_cache_size; i++) { + int ret = spi_w8r8(spi, i | AK4104_READ); + if (ret < 0) { + dev_err(&spi->dev, "SPI write failure\n"); + return ret; + } + + reg_cache[i] = ret; + } + + return 0; +} + +static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + reg &= AK4104_REG_MASK; + reg |= AK4104_WRITE; + + /* only write to the hardware if value has changed */ + if (cache[reg] != value) { + u8 tmp[2] = { reg, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { + dev_err(&spi->dev, "SPI write failed\n"); + return -EIO; + } + + cache[reg] = value; + } + + return 0; +} + +static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val = 0; + + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1); + + /* set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= AK4104_CONTROL1_DIF0; + break; + case SND_SOC_DAIFMT_I2S: + val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1; + break; + default: + dev_err(codec->dev, "invalid dai format\n"); + return -EINVAL; + } + + /* This device can only be slave */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); +} + +static int ak4104_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret, val = 0; + + /* set the IEC958 bits: consumer mode, no copyright bit */ + val |= IEC958_AES0_CON_NOT_COPYRIGHT; + ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val); + + val = 0; + + switch (params_rate(params)) { + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + + ret = ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val); + if (ret < 0) + return ret; + + return 0; +} + +struct snd_soc_dai ak4104_dai = { + .name = DRV_NAME, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_32000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = { + .hw_params = ak4104_hw_params, + .set_fmt = ak4104_set_dai_fmt, + } +}; + +static struct snd_soc_codec *ak4104_codec; + +static int ak4104_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ak4104_private *ak4104; + int ret, val; + + spi->bits_per_word = 8; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + if (!ak4104) { + dev_err(&spi->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + codec = &ak4104->codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &spi->dev; + codec->name = DRV_NAME; + codec->owner = THIS_MODULE; + codec->dai = &ak4104_dai; + codec->num_dai = 1; + codec->private_data = ak4104; + codec->control_data = spi; + codec->reg_cache = ak4104->reg_cache; + codec->reg_cache_size = AK4104_NUM_REGS; + + /* read all regs and fill the cache */ + ret = ak4104_fill_cache(codec); + if (ret < 0) { + dev_err(&spi->dev, "failed to fill register cache\n"); + return ret; + } + + /* read the 'reserved' register - according to the datasheet, it + * should contain 0x5b. Not a good way to verify the presence of + * the device, but there is no hardware ID register. */ + if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) != + AK4104_RESERVED_VAL) { + ret = -ENODEV; + goto error_free_codec; + } + + /* set power-up and non-reset bits */ + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN; + ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + goto error_free_codec; + + /* enable transmitter */ + val = ak4104_read_reg_cache(codec, AK4104_REG_TX); + val |= AK4104_TX_TXE; + ret = ak4104_spi_write(codec, AK4104_REG_TX, val); + if (ret < 0) + goto error_free_codec; + + ak4104_codec = codec; + ret = snd_soc_register_dai(&ak4104_dai); + if (ret < 0) { + dev_err(&spi->dev, "failed to register DAI\n"); + goto error_free_codec; + } + + spi_set_drvdata(spi, ak4104); + dev_info(&spi->dev, "SPI device initialized\n"); + return 0; + +error_free_codec: + kfree(ak4104); + ak4104_dai.dev = NULL; + return ret; +} + +static int __devexit ak4104_spi_remove(struct spi_device *spi) +{ + int ret, val; + struct ak4104_private *ak4104 = spi_get_drvdata(spi); + + val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + /* clear power-up and non-reset bits */ + val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + return ret; + + ak4104_codec = NULL; + kfree(ak4104); + return 0; +} + +static int ak4104_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = ak4104_codec; + int ret; + + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + return ret; + } + + /* Register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return 0; +} + +static int ak4104_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + return 0; +}; + +struct snd_soc_codec_device soc_codec_device_ak4104 = { + .probe = ak4104_probe, + .remove = ak4104_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_ak4104); + +static struct spi_driver ak4104_spi_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = ak4104_spi_probe, + .remove = __devexit_p(ak4104_spi_remove), +}; + +static int __init ak4104_init(void) +{ + pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); + return spi_register_driver(&ak4104_spi_driver); +} +module_init(ak4104_init); + +static void __exit ak4104_exit(void) +{ + spi_unregister_driver(&ak4104_spi_driver); +} +module_exit(ak4104_exit); + +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h new file mode 100644 index 0000000..eb88fe7 --- /dev/null +++ b/sound/soc/codecs/ak4104.h @@ -0,0 +1,7 @@ +#ifndef _AK4104_H +#define _AK4104_H + +extern struct snd_soc_dai ak4104_dai; +extern struct snd_soc_codec_device soc_codec_device_ak4104; + +#endif -- 1.6.1.3

2 2

[alsa-devel] configure script doesn't succesfully end with the latest snapshots
by Ozan Çağlayan 09 Mar '09

09 Mar '09

Latest working snapshots: alsa-driver-20090305.tar.bz2 alsa-driver-20090306.tar.bz2 Affected snapshots: alsa-driver-20090307.tar.bz2 alsa-driver-20090308.tar.bz2 Problem -------- -> 20090307 During the configuration step, configure script complains that it can't find some commands: checking whether gcc needs -traditional... no checking for current directory... /root/devel/kernel/drivers/alsa-driver/alsa-driver-20090307 ./configure: line 3863: ALSA_TOPLEVEL_INIT: command not found checking cross compile... checking for directory with kernel source... /lib/modules/2.6.25.20-114/source checking for directory with kernel build... /lib/modules/2.6.25.20-114/build .. .. checking for CONFIG_HAS_DMA... yes ./configure: line 13325: ALSA_TOPLEVEL_SELECT: command not found ./configure: line 13327: ALSA_PARSE_KCONFIG: command not found ./configure: line 13329: ALSA_TOPLEVEL_DEFINES: command not found ./configure: line 13339: ALSA_TOPLEVEL_OUTPUT: command not found configure: creating ./config.status Eventhough it complains, the configuration step ends succesfully and I can compile the drivers with make. -> 20090308 It's even worse. After complaining about the same ALSA_* stuff, it fails also on the following error: configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf config.status: WARNING: 'Makefile.conf.in' seems to ignore the --datarootdir setting config.status: creating snddevices config.status: creating utils/alsa-driver.spec config.status: creating utils/buildrpm config.status: creating toplevel.config config.status: creating utils/alsasound config.status: creating utils/alsasound.posix config.status: creating include/pci_ids_compat.h config.status: creating include/i2c-id_compat.h config.status: creating include/config.h config.status: error: cannot find input file: include/config1.h.in -- Ozan Çağlayan <ozan_at_pardus.org.tr>

2 1

[alsa-devel] SoC scenarii API
by Robert Jarzmik 08 Mar '09

08 Mar '09

Hi Liam and Mark, As mioa701 submission was stopped due to the need of a generic scenarii API, this a first attempt to design such an API in order to unblock mioa701 submission. This is oversimplified, no convenience macros are provided, just external data structures and namespace polution. I'm expecting a lot of comments from both of you. Cheers. -- Robert enum pin_change { SCEN_PIN_OFF = 0, /* Disable the pin */ SCEN_PIN_ON, /* Enable the pin */ SCEN_PIN_LEAVE /* Leave the pin in previous state */ }; struct setup_mixmux { char *mixname; /* Codec Mixer or Muxer name */ int val; /* Codec Mixer or Muxer value to enforce */ }; /** * struct soc_scenario - describes one sound usecase * @name: Scenario name, value as will be seen in alsa "SoC Mode" alsa control * @pin_setup: Pin configuration to perform on scenario activation * Table of all pins, as they should be configured. Each elements is a * pin_change value, describing how to handle a specific pin. * Table size must be the same as in snd_soc_scenario_init(). * @mixer_cleanup: Mixers/muxes to set up in phase (b) * Table of all mixers/muxes to modify, NULL terminated. * @mixer_setup: Mixers/muxers to set up in phase (c) * Table of all mixers/muxes to modify, NULL terminated * @lvol_master: Left volume aliased to "SoC Volume" * @rvol_master: Right volume aliased to "SoC Volume" * @lvol_mute: Left volume mute aliased to "SoC Volume" mute control * @rvol_mute: Right volume mute aliased to "SoC Volume" mute control * * This structure describes what a scenario change implies. The behaviour is to : * a) enable several pins, disable others, leave others in their state * (understand here snd_soc_dapm_enable_pin) * b) set up some mixers and muxers to prepare the change, and clean up. * This should be normally used to restore a "default state" if need be, before * activating the mixers/muxers into their final state. * Note: table in NULL, or always terminated by NULL pointer * Note: this cleanup may/should be shared amongst all scenarii * c) set up some mixers and muxers to the final state * This should be normally used to restore a "default state" if need be, before * activating the mixers/muxers into their final state. * Note: table in NULL, or always terminated by NULL pointer */ struct soc_scenario { const char *name; /* scenario name */ const unsigned char pin_setup[]; /* pin_change for pins */ const struct setup_mixmux mixes_cleanup[]; /* mix cleanup */ const struct setup_mixmux mixes_setup[]; /* mix setup */ const char *lvol_master; /* left volume master */ const char *rvol_master; /* right volume master */ const char *lvol_mute; /* left volume mute */ const char *rvol_mute; /* right volume mute */ }; /** * snd_soc_scenario_init - initialize soc scenarii * @codec: codec associated to the pins/mixers/muxes/volumes/mutes * @scenario: table of scenarii * @nb_scenarii: number of scenarii * @pin_names: table of pin names * @nb_pins: number of pins handled by scenario * * Initialise the soc scenarii engine. The first scenario (0) will be used. * By default, the user could leave this scenario as non intrusive (not a single * pin changed, no mixers/muxes changes, and volume master inactive). */ int snd_soc_scenario_init(struct snd_soc_codec *codec, struct soc_scenario *scenario, int nb_scenarii, char *pin_names[], int nb_pins); /** * snd_soc_scenario_init - initialize soc scenarii * @codec: codec used for the init */ void snd_soc_scenario_release(struct snd_soc_codec *code);

2 5

[alsa-devel] Soc-scenario API: news
by Robert Jarzmik 08 Mar '09

08 Mar '09

Hi Mark, I send an update of the proposition for the soc-scenario API 2 weeks ago and no news ... I was wondering if you had received that mail. Just in case, I'll make a copy in this one. Cheers. -- Robert From: Robert Jarzmik <robert.jarzmik(a)free.fr> Subject: Re: [alsa-devel] SoC scenarii API To: Mark Brown <broonie(a)sirena.org.uk> Cc: alsa-devel(a)alsa-project.org, lrg(a)kernel.org Date: Fri, 20 Feb 2009 20:04:24 +0100 X-Spam-Status: No, score=0.0 required=5.0 tests=none autolearn=disabled version=3.2.4 Message-ID: <87vdr5orpj.fsf(a)free.fr> User-Agent: Gnus/5.11 (Gnus v5.11) Emacs/22.2 (gnu/linux) Hi Mark, It's been a while since my last post. This is a new proposition for the API. I hope I have not forgotten what has been said previously. As a side note, I'm a bit bothered by your comment about using "snd_ctl_elem_value" instead of integers in structure soc_scen_setup_mix. My concern is about size : snd_ctl_elem_value is huge, and as there will be many mixes/muxes, the memory print will be huge. Can't we deal with only integers for mixers and muxes ? Cheers. -- Robert

1 0

[alsa-devel] [PATCH] CS4270: comments: slave mode is now supported
by Daniel Mack 08 Mar '09

08 Mar '09

Just a minor thing: make the comments reflect the new slave mode feature. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 9 ++++----- 1 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 4c7e546..0558674 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -14,12 +14,11 @@ * * 1) Software mode is supported. Stand-alone mode is not supported. * 2) Only I2C is supported, not SPI - * 3) Only Master mode is supported, not Slave. - * 4) The machine driver's 'startup' function must call + * 3) The machine driver's 'startup' function must call * cs4270_set_dai_sysclk() with the value of MCLK. - * 5) Only I2S and left-justified modes are supported - * 6) Power management is not supported - * 7) The only supported control is volume and hardware mute (if enabled) + * 4) Only I2S and left-justified modes are supported + * 5) Power management is not supported + * 6) The only supported control is volume and hardware mute (if enabled) */ #include <linux/module.h> -- 1.6.1.3

2 4

[PATCH] CS4270: bring feature/limitations list in sync
by Daniel Mack 08 Mar '09

08 Mar '09

Removes numbers from the list of features/limitations and makes it reflect recent changes to the code. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/codecs/cs4270.c | 15 +++++++-------- 1 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 4c7e546..d16aa81 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,14 +12,13 @@ * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is not supported. - * 2) Only I2C is supported, not SPI - * 3) Only Master mode is supported, not Slave. - * 4) The machine driver's 'startup' function must call - * cs4270_set_dai_sysclk() with the value of MCLK. - * 5) Only I2S and left-justified modes are supported - * 6) Power management is not supported - * 7) The only supported control is volume and hardware mute (if enabled) + * - Software mode is supported. Stand-alone mode is not supported. + * - Only I2C is supported, not SPI + * - Support for master and slave mode + * - The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * - Only I2S and left-justified modes are supported + * - Power management is not supported */ #include <linux/module.h> -- 1.6.1.3

1 0

Re: [alsa-devel] [PATCH 1/1] ALSA: ASoC: Davinci: Fix incorrect machine type for SFFSDR board
by Kevin Hilman 08 Mar '09

08 Mar '09

Hugo Villeneuve <hugo(a)hugovil.com> writes: > Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> Signed-off-by: Kevin Hilman <khilman(a)deeprootsystems.com> This has also been pushed to DaVinci git until it get merged into the asoc base. Kevin > --- > sound/soc/davinci/Kconfig | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig > index 7d3a85d..411a710 100644 > --- a/sound/soc/davinci/Kconfig > +++ b/sound/soc/davinci/Kconfig > @@ -21,7 +21,7 @@ config SND_DAVINCI_SOC_EVM > > config SND_DAVINCI_SOC_SFFSDR > tristate "SoC Audio support for SFFSDR" > - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR > + depends on SND_DAVINCI_SOC && MACH_SFFSDR > select SND_DAVINCI_SOC_I2S > select SND_SOC_PCM3008 > select SFFSDR_FPGA > -- > 1.5.4.5 > > > _______________________________________________ > Davinci-linux-open-source mailing list > Davinci-linux-open-source(a)linux.davincidsp.com > http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source

1 0

Re: [alsa-devel] [PATCH 1/1] ALSA: ASoC: Davinci: Fix incorrect machine type for SFFSDR board
by Kevin Hilman 08 Mar '09

08 Mar '09

Hugo Villeneuve <hugo(a)hugovil.com> writes: > Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> > --- > sound/soc/davinci/Kconfig | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig > index 7d3a85d..411a710 100644 > --- a/sound/soc/davinci/Kconfig > +++ b/sound/soc/davinci/Kconfig > @@ -21,7 +21,7 @@ config SND_DAVINCI_SOC_EVM > > config SND_DAVINCI_SOC_SFFSDR > tristate "SoC Audio support for SFFSDR" > - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR > + depends on SND_DAVINCI_SOC && MACH_SFFSDR > select SND_DAVINCI_SOC_I2S > select SND_SOC_PCM3008 > select SFFSDR_FPGA Thanks, pushed. Kevin

1 0

[alsa-devel] [PATCH 2.6.30] ASoC: Improve pause/unpause performance in Freescale 8610 drivers
by Timur Tabi 07 Mar '09

07 Mar '09

Add support for true pause and unpause. Without this, mplayer will drop some audio (less than one second, but still noticeable) when pausing playback. Remove support for PM suspend and resume from the trigger function, since the driver doesn't support PM anyway. Optimize the delay after starting capture. Instead of delaying 1ms, the driver now polls the hardware. The new delay is shorter by over 90% yet still effective. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++++--------- 2 files changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 58a3fa4..b3eb857 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -142,7 +142,8 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_JOINT_DUPLEX, + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_PAUSE, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8cb6bcf..b7733e6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -464,28 +464,33 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + long timeout = jiffies + 10; + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - /* - * I think we need this delay to allow time for the SSI - * to put data into its FIFO. Without it, ALSA starts - * to complain about overruns. + /* Wait until the SSI has filled its FIFO. Without this + * delay, ALSA complains about overruns. When the FIFO + * is full, the DMA controller initiates its first + * transfer. Until then, however, the DMA's DAR + * register is zero, which translates to an + * out-of-bounds pointer. This makes ALSA think an + * overrun has occurred. */ - mdelay(1); + while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && + (jiffies < timeout)); + if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) + return -EIO; } break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); -- 1.6.1.3

2 1

[alsa-devel] No headphone output on HT Omega Claro Halo
by Christopher J. Atherton 07 Mar '09

07 Mar '09

I have an HT Omega Claro Halo (CMI8878) card that is not entirely working. This card has two analog outputs, a pair of RCA connectors and also a separate headphone output (3.5mm jack). The RCA outputs work, but the headphone output, which notably passes a signal from a built in headphone amp, does not. Interestingly, when I turn up the master volume really high, I can hear pops/cracks on the headphone output. Thanks for any assistance. -- Kristoff

2 3

[alsa-devel] [PATCH 1/1] ALSA: ASoC: Davinci: Fix incorrect machine type for SFFSDR board
by Hugo Villeneuve 06 Mar '09

06 Mar '09

Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 7d3a85d..411a710 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -21,7 +21,7 @@ config SND_DAVINCI_SOC_EVM config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + depends on SND_DAVINCI_SOC && MACH_SFFSDR select SND_DAVINCI_SOC_I2S select SND_SOC_PCM3008 select SFFSDR_FPGA -- 1.5.4.5

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 06 Mar '09

06 Mar '09

The following changes since commit 5f2a9384a9291d898b4bf85c4fbf497eef582977: Philipp Zabel (1): ASoC: UDA1380: DATAI is slave only are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Dooks (6): S3C24XX: Move and update IIS headers S3C: Move <mach/audio.h> to <plat/audio.h> ASoC: Add JIVE audio support ASoC: Split s3c2412-i2s.c into core and SoC specific parts ASoC: Add s3c64xx-i2s support ASoC: Select DMA if I2S is configured Daniel Mack (2): ASoC: Fix name of register bit in pxa-ssp ASoC: add two more bitfields for PXA SSP Jonas Andersson (1): ASoC: wm8510 pll settings Lopez Cruz, Misael (3): ASoC: Add GPIO support for jack reporting interface ASoC: Add DAPM machine widgets to SDP3430 driver ASoC: Add headset jack detection for SDP3430 machine driver Mark Brown (5): ASoC: Refresh JIVE driver ASoC: Factor out DAPM widget power check into separate function ASoC: Fix memory allocation for snd_soc_dapm_switch names ASoC: Update Kconfig for Samsung CPUs to reflect S3C64xx support ASoC: Staticise workqueue function for GPIO jack detection Mike Frysinger (2): ASoC: Blackfin: move gpio_err behind the define that is only user of it ASoC: Blackfin: fix typo in MUTE definition Peter Ujfalusi (1): ASoC: TWL4030: Make the HS ramp delay configurable Timur Tabi (1): ASoC: add support for SSI asynchronous mode to the Freescale SSI drivers arch/arm/mach-pxa/include/mach/regs-ssp.h | 7 +- arch/arm/mach-s3c2410/dma.c | 2 +- arch/arm/mach-s3c2410/include/mach/hardware.h | 3 - arch/arm/mach-s3c2410/include/mach/io.h | 2 +- arch/arm/mach-s3c2412/dma.c | 4 +- arch/arm/mach-s3c2440/dma.c | 2 +- arch/arm/mach-s3c2443/dma.c | 2 +- arch/arm/mach-shark/include/mach/io.h | 2 +- .../include/mach => plat-s3c/include/plat}/audio.h | 0 .../arm/plat-s3c/include/plat}/regs-s3c2412-iis.h | 3 + arch/arm/plat-s3c24xx/clock-dclk.c | 1 + .../arm/plat-s3c24xx/include/plat}/regs-iis.h | 0 include/sound/soc.h | 32 + sound/soc/atmel/playpaq_wm8510.c | 24 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/codecs/ad73311.h | 2 +- sound/soc/codecs/twl4030.c | 18 +- sound/soc/codecs/wm8510.c | 4 +- sound/soc/fsl/fsl_ssi.c | 42 +- sound/soc/fsl/fsl_ssi.h | 2 + sound/soc/fsl/mpc8610_hpcd.c | 5 + sound/soc/omap/sdp3430.c | 105 ++++- sound/soc/pxa/pxa-ssp.c | 6 +- sound/soc/s3c24xx/Kconfig | 29 +- sound/soc/s3c24xx/Makefile | 6 + sound/soc/s3c24xx/jive_wm8750.c | 201 ++++++ sound/soc/s3c24xx/neo1973_wm8753.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 645 ++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.h | 90 +++ sound/soc/s3c24xx/s3c2412-i2s.c | 604 +------------------ sound/soc/s3c24xx/s3c2412-i2s.h | 17 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 220 +++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 31 + sound/soc/soc-dapm.c | 265 +++++---- sound/soc/soc-jack.c | 129 ++++ 39 files changed, 1745 insertions(+), 776 deletions(-) rename arch/arm/{mach-s3c2410/include/mach => plat-s3c/include/plat}/audio.h (100%) rename {include/asm-arm/plat-s3c24xx => arch/arm/plat-s3c/include/plat}/regs-s3c2412-iis.h (96%) rename {include/asm-arm/plat-s3c24xx => arch/arm/plat-s3c24xx/include/plat}/regs-iis.h (100%) create mode 100644 sound/soc/s3c24xx/jive_wm8750.c create mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.c create mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.h create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h

3 3

[alsa-devel] [PATCH] ASoC: add support for SSI asynchronous mode to the Freescale SSI drivers
by Timur Tabi 06 Mar '09

06 Mar '09

Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If defined, the SSI is programmed into asynchronous mode, otherwise it is programmed into synchronous mode. In asynchronous mode, pin SRCK must be connected to the same clock source as STFS, and pin SRFS must be connected to the same signal as STFS. Asynchronous mode allows playback and capture to use different sample sizes. It also technically allows different sample rates, but the driver does not support that. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++++++++++++++---------- sound/soc/fsl/fsl_ssi.h | 2 ++ sound/soc/fsl/mpc8610_hpcd.c | 5 +++++ 3 files changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6844009..8cb6bcf 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -72,6 +72,7 @@ * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened + * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -86,6 +87,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; + int asynchronous; struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; @@ -301,9 +303,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, - CCSR_SSI_SCR_TFR_CLK_DIS | - CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + clrsetbits_be32(&ssi->scr, + CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, + CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE + | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -382,10 +385,15 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, first_runtime->rate, first_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); + /* If we're in synchronous mode, then we need to constrain + * the sample size as well. We don't support independent sample + * rates in asynchronous mode. + */ + if (!ssi_private->asynchronous) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); ssi_private->second_stream = substream; } @@ -421,13 +429,18 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); - u32 wl; + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + !ssi_private->asynchronous) + clrsetbits_be32(&ssi->stccr, + CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, + CCSR_SSI_SxCCR_WL_MASK, wl); } return 0; @@ -653,6 +666,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) ssi_private->ssi_phys = ssi_info->ssi_phys; ssi_private->irq = ssi_info->irq; ssi_private->dev = ssi_info->dev; + ssi_private->asynchronous = ssi_info->asynchronous; ssi_private->dev->driver_data = fsl_ssi_dai; @@ -703,6 +717,14 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) } EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); +static int __init fsl_ssi_init(void) +{ + printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); + + return 0; +} +module_init(fsl_ssi_init); + MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 83b44d7..eade01f 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -208,6 +208,7 @@ struct ccsr_ssi { * ssi_phys: physical address of the SSI registers * irq: IRQ of this SSI * dev: struct device, used to create the sysfs statistics file + * asynchronous: 0=synchronous mode, 1=asynchronous mode */ struct fsl_ssi_info { unsigned int id; @@ -215,6 +216,7 @@ struct fsl_ssi_info { dma_addr_t ssi_phys; unsigned int irq; struct device *dev; + int asynchronous; }; struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index acf39a6..ef67d1c 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -353,6 +353,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } ssi_info.irq = machine_data->ssi_irq; + /* Do we want to use asynchronous mode? */ + ssi_info.asynchronous = + of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0; + if (ssi_info.asynchronous) + dev_info(&ofdev->dev, "using asynchronous mode\n"); /* Map the global utilities registers. */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); -- 1.6.1.3

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[alsa-devel] [PATCH 2/2] ASoC: Add headset jack detection for SDP3430 machine driver
by Lopez Cruz, Misael 06 Mar '09

06 Mar '09

Add headset jack detection for SDP3430 boards using SoC jack reporting interface. Headset detection on SDP3430 board is achieved through TWL4030 GPIO_2 pin. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 43 +++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4eab4b4..715c648 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -122,7 +123,7 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); @@ -144,6 +145,27 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", @@ -194,7 +216,21 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - return 0; + /* Headset jack detection */ + ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return ret; err1: printk(KERN_ERR "Unable to add platform device\n"); @@ -206,6 +242,9 @@ module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { + snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + platform_device_unregister(sdp3430_snd_device); } module_exit(sdp3430_soc_exit); -- 1.5.4.3

2 1

[alsa-devel] ALC889A (Gigabyte EX58-UD5) pin support
by Jo Shields 06 Mar '09

06 Mar '09

I've got a pretty new Gigabyte board, using snd-hda-intel and an ALC889A chip. Whilst the chip seems to be detected fine, I have no functional audio out - no sound on any of the ports. dmesg shows that ALSA has no idea about the specific model: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... If I'm reading the source correctly, the default pin layout is 3stack-digout. I tried 6stack-digout, but it didn't help. Linked is output from the diagnostic script: http://www.alsa-project.org/db/?f=3938bc16481f549203e373e303c4b1c1dfb303dd I'm happy enough to try things out in an effort to get the onboard sound to behave

2 2

[alsa-devel] [PATCH] ASoC: Fix long name of control for dapm_switch
by Joonyoung Shim 06 Mar '09

06 Mar '09

The length of long control name doesn't include the length of widget name at present when widget->id is snd_soc_dapm_switch, so the long control name of dapm_switch is shortened. To avoid this problem, the long control name only for dapm_mixer_named_ctl must have simply the kcontrol name. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/soc-dapm.c | 21 +++++++++------------ 1 files changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4b8dbbf..7a1fed1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -331,26 +331,23 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, * for dapm_mixer_named_ctl this is simply the * kcontrol name. */ - name_len = strlen(w->kcontrols[i].name) + 1; - if (w->id == snd_soc_dapm_mixer) - name_len += 1 + strlen(w->name); + if (w->id == snd_soc_dapm_mixer_named_ctl) + name_len = strlen(w->kcontrols[i].name) + 1; + else + name_len = 2 + strlen(w->name) + + strlen(w->kcontrols[i].name); path->long_name = kmalloc(name_len, GFP_KERNEL); if (path->long_name == NULL) return -ENOMEM; - switch (w->id) { - case snd_soc_dapm_mixer: - default: - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); - break; - case snd_soc_dapm_mixer_named_ctl: + if (w->id == snd_soc_dapm_mixer_named_ctl) snprintf(path->long_name, name_len, "%s", w->kcontrols[i].name); - break; - } + else + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); path->long_name[name_len - 1] = '\0'; -- 1.5.6.3 -- - Joonyoung Shim

2 3

[alsa-devel] Fwd: [LAD] kernel 2.6.29 rc6 no working rme multiface 1
by Dmitry Baikov 05 Mar '09

05 Mar '09

---------- Forwarded message ---------- From: Dmitry Baikov <dsbaikov(a)gmail.com> Date: Mon, Mar 2, 2009 at 5:49 AM Subject: Re: [LAD] kernel 2.6.29 rc6 no working rme multiface 1 To: "m.wolkstein(a)gmx.de" <m.wolkstein(a)gmx.de> Cc: The Linux Audio Developers' Mailing List <linux-audio-dev(a)lists.linuxaudio.org> On Mon, Mar 2, 2009 at 4:59 AM, m.wolkstein(a)gmx.de <m.wolkstein(a)gmx.de> wrote: > hiho, > my name is michael wolkstein and i wrote the first time to this list. > sorry about my english. but i am no english native-speaker. > > after booting kernel 2.6.29 rc6 i get a problem with my rme multiface 1 connected true pcmcia cardbusinterface. > > i get this message after insert my card. > RME Hammerfall DSP 0000:16:00.0: enabling device (0000 -> 0002) > RME Hammerfall DSP 0000:16:00.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 > Hammerfall-DSP: no Digiface or Multiface connected! > RME Hammerfall DSP 0000:16:00.0: PCI INT A disabled > RME Hammerfall DSP: probe of 0000:16:00.0 failed with error -5 > > i have compile the kernel included alsa module. > also i have installed the newest alsa-firmware 1.0.19 and newest alsa-tools 1.0.19. > > my old kernel 2.6.24 load both firmwares correct. > > the old one 1.0.16 (debian unstable) > and the new 1.0.19 download from the alsa-project page > > the new kernel produce errors. kernel module snd_hdsp is already loaded with the new kernel. Have the same problem here. 2.6.24 works, 2.6.28 doesn't (tried 2.6.8-ARCH and 2.6.28.7-1-ARCH). I reported the problem to alsa-devel list, but they said I have hw problems (I think I do not). The latest arch kernel gives oops, if the card is inserted before booting. It crashes during udev event processing (i.e. something from userspace triggers the problem) If anybody is really interested, I can try to provide kernel backtrace (either type it in, or try to use serial console). Dmitry.

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[alsa-devel] [PATCH 1/4] ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite
by Philipp Zabel 05 Mar '09

05 Mar '09

For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result in exactly the same behaviour. Now it is possible to use 16-bit single slot transfers in pxa-ssp, which are needed for Magician to get two frame clock pulses per sample (one for each channel). Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 3 +-- sound/soc/pxa/zylonite.c | 7 +++++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4a973ab..c49bb12 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -644,8 +644,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - if (params_channels(params) > 1) - sscr0 |= SSCR0_EDSS; + /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 0140a25..9f6116e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -127,8 +127,11 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* We're not really in network mode but the emulation wants this. */ - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + /* Use network mode for stereo, one slot per channel. */ + if (params_channels(params) > 1) + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 2); + else + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); if (ret < 0) return ret; -- 1.6.1.3

4 12

[alsa-devel] Keyboard volume buttons
by Harsha, Priya 05 Mar '09

05 Mar '09

Hi, Can anyone tell me how keyboard's volume buttons are converted into real volume change? Will there be an application that monitors the keyboard volume button events and call ALSA API to set the volume? Thanks Harsha

4 4

[PATCH] pxa-ssp: add two more bitfields
by Daniel Mack 05 Mar '09

05 Mar '09

Add two more bitfields for the PSP register. As they seem to exist for PXA3xx only, define them conditionally. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index a9ee355..018f6d6 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -109,6 +109,11 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ +#if defined(CONFIG_PXA3xx) +#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ +#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ +#endif + #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ #define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ -- 1.6.1.3

1 0

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Add mixer control for the HS ramp delay
by Peter Ujfalusi 05 Mar '09

05 Mar '09

Hello, The following patch adds user control for the desired ramp delay used on the Headset output - it has been hardwired to use 874/645/437 ms. The actuall delay depends on the MCLK frequency, for example: value 0x0 in RAMP_DELAY means 2^19/f(MCLK), which is: 27 ms when the f(MCLK) = 19.2 MHz 20 ms when the f(MCLK) = 26 MHz 14 ms when the f(MCLK) = 38.4 MHz value 0x5 in RAMP_DELAY means 2^24/f(MCLK), which is: 874 ms when the f(MCLK) = 19.2 MHz 645 ms when the f(MCLK) = 26 MHz 437 ms when the f(MCLK) = 38.4 MHz I think having the actual ms values in the enums are makes more sense than to have the calculation ("2^24/F(MCLK)" for example). If one can suggest more descriptive name, I would love to change the texts. --- Peter Ujfalusi (1): ASoC: TWL4030: Make the HS ramp delay configurable sound/soc/codecs/twl4030.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-)

2 2

[alsa-devel] pcm_oss: Can period_bytes - buffer_used be less than 0?
by Roel Kluin 05 Mar '09

05 Mar '09

vi sound/core/oss/pcm_oss.c +1572 (in linus' git tree) and note: static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) { ... size_t size; ... :1608 size = runtime->oss.period_bytes - runtime->oss.period_ptr; can this period_bytes - buffer_used become less than 0? Roel

2 1

Re: [alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Timur Tabi 05 Mar '09

05 Mar '09

Eric Miao wrote: > Updated as below, (gzip version attached due to mailing list > mail size limitation). > > From 74a2b6dfff2948d9d04b33d0a02f645f8c12caf7 Mon Sep 17 00:00:00 2001 > From: Eric Miao <eric.miao(a)marvell.com> > Date: Mon, 2 Mar 2009 22:38:31 +0800 > Subject: [PATCH] ASoC: make ops a pointer of 'struct snd_soc_dai' > > Considering the fact that most cpu_dai or codec_dai are using a same > 'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better > made a pointer instead, to make sharing easier and code a bit cleaner. > > The patch below is rather preliminary since the asoc tree is being > actively developed, and this touches almost every piece of code, > (and possibly many others in development need to be changed as > well). Building of all codecs are OK, yet to every SoC, I didn't test > that. > > Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Tested on an MPC8610 HPCD, and it works. That's all I can vouch for, though. Acked-by: Timur Tabi <timur(a)freescale.com> -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] [PATCH 2/2] ASoC: Add headset jack detection for SDP3430 machine driver
by Lopez Cruz, Misael 04 Mar '09

04 Mar '09

Add headset jack detection for SDP3430 boards using SoC jack reporting interface. Headset detection on SDP3430 board is achieved through TWL4030 GPIO_2 pin. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 49 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4eab4b4..b595aa4 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -122,7 +123,7 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); @@ -144,6 +145,29 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +/* Headset jack */ +struct snd_soc_jack *hs_jack; + +/* Headset jack DAPM pins */ +struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + .invert = 0, + }, +}; + +/* Headset jack gpios */ +struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .invert = 0, + .debounce_time = 200, + }, +}; + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", @@ -194,8 +218,25 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - return 0; + /* Headset jack detection */ + hs_jack = kzalloc(sizeof(struct snd_soc_jack), GFP_KERNEL); + if (!hs_jack) + return -ENOMEM; + + ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", + SND_JACK_HEADSET, hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + return ret; err1: printk(KERN_ERR "Unable to add platform device\n"); platform_device_put(sdp3430_snd_device); @@ -206,6 +247,10 @@ module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { + snd_soc_jack_free_gpios(hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + kfree(hs_jack); + platform_device_unregister(sdp3430_snd_device); } module_exit(sdp3430_soc_exit); -- 1.5.4.3

2 2

[alsa-devel] [PATCH 1/3] ALSA: hda - Add 4 channel mode for 3stack-hp model (ALC888)
by Herton Ronaldo Krzesinski 04 Mar '09

04 Mar '09

Add additional 4 channel mode for 3stack-hp models. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++--- 1 files changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c02f78..3696ff3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8068,24 +8068,45 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; +/* + * 2ch mode + */ static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } + { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc888_3st_hp_4ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 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, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, { 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, 0x01 }, + { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[2] = { +static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, + { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, }; -- 1.6.1.3

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[alsa-devel] [PATCH 1/2] ASoC: Add DAPM machine widgets to SDP3430 driver
by Lopez Cruz, Misael 04 Mar '09

04 Mar '09

Add DAPM machine domain widgets to SDP3430 machine driver. Interconnection: * Ext Mic: MAINMIC, SUBMIC * Ext Spk: HFL, HFR * Headset Jack: HSMIC, HSOL, HSOR Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 64 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e226fa7..4eab4b4 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -81,12 +81,76 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +/* SDP3430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset: HSMIC (with bias), HSOL, HSOR */ + {"Headset Jack", NULL, "HSOL"}, + {"Headset Jack", NULL, "HSOR"}, + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Jack"}, +}; + +static int sdp3430_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add SDP3430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets, + ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP3430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP3430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AUXL"); + snd_soc_dapm_nc_pin(codec, "AUXR"); + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai, + .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- 1.5.4.3

2 1

[alsa-devel] [PATCH 3/3] ALSA: hda - Map 3stack-hp model (ALC888) for HP Educ.ar
by Herton Ronaldo Krzesinski 04 Mar '09

04 Mar '09

Added model=3stack-hp for HP Educ.ar desktop machine (103c:2a72). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 251647d..91ef9f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8673,6 +8673,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), -- 1.6.1.3

1 0

[alsa-devel] [PATCH 2/3] ALSA: hda - Add headphone automute support for 3stack-hp model (ALC888)
by Herton Ronaldo Krzesinski 04 Mar '09

04 Mar '09

Mute speaker outputs on headphone insertion for machines that use 3stack-hp model. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 38 +++++++++++++++++++++++++++++++++----- 1 files changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3696ff3..251647d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8056,16 +8056,42 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_6st_dell_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc888_3st_hp_front_automute(struct hda_codec *codec) +{ + unsigned int present, bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_3st_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc888_3st_hp_front_automute(codec); + break; + } +} + static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ - { } -}; - -static struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - { } + { } /* end */ }; /* @@ -8950,6 +8976,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, + .unsol_event = alc888_3st_hp_unsol_event, + .init_hook = alc888_3st_hp_front_automute, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, -- 1.6.1.3

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Re: [alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Mark Brown 04 Mar '09

04 Mar '09

On Tue, Mar 03, 2009 at 09:41:00AM +0800, Eric Miao wrote: > Updated as below, (gzip version attached due to mailing list > mail size limitation). Only doing the attachment should put you under the limit and works well with git am so might've been worth trying. I've applied it so no need to worry, though - thanks!

1 0

[alsa-devel] [PATCH] ASoC: Add GPIO support for jack reporting interface
by Lopez Cruz, Misael 04 Mar '09

04 Mar '09

Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc.h | 32 ++++++++++++ sound/soc/soc-jack.c | 129 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 161 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 68d8149..e5b4fd0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include <linux/platform_device.h> #include <linux/types.h> #include <linux/workqueue.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> @@ -168,6 +170,9 @@ struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; +#ifdef CONFIG_GPIOLIB +struct snd_soc_jack_gpio; +#endif typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); @@ -194,6 +199,12 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); +#ifdef CONFIG_GPIOLIB +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios); +#endif /* codec IO */ #define snd_soc_read(codec, reg) codec->read(codec, reg) @@ -262,6 +273,27 @@ struct snd_soc_jack_pin { bool invert; }; +/** + * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection + * + * @gpio: gpio number + * @name: gpio name + * @report: value to report when jack detected + * @invert: report presence in low state + * @debouce_time: debouce time in ms + */ +#ifdef CONFIG_GPIOLIB +struct snd_soc_jack_gpio { + unsigned int gpio; + const char *name; + int report; + int invert; + int debounce_time; + struct snd_soc_jack *jack; + struct work_struct work; +}; +#endif + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3..15bb80f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,10 @@ #include <sound/jack.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> +#include <linux/delay.h> /** * snd_soc_jack_new - Create a new jack @@ -136,3 +140,128 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return 0; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); + +#ifdef CONFIG_GPIOLIB +/* gpio detect */ +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +{ + struct snd_soc_jack *jack = gpio->jack; + int enable; + int report; + + if (gpio->debounce_time > 0) + mdelay(gpio->debounce_time); + + enable = gpio_get_value(gpio->gpio); + if (gpio->invert) + enable = !enable; + + if (enable) + report = gpio->report; + else + report = 0; + + snd_soc_jack_report(jack, report, gpio->report); +} + +/* irq handler for gpio pin */ +static irqreturn_t gpio_handler(int irq, void *data) +{ + struct snd_soc_jack_gpio *gpio = data; + + schedule_work(&gpio->work); + + return IRQ_HANDLED; +} + +/* gpio work */ +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio; + + gpio = container_of(work, struct snd_soc_jack_gpio, work); + snd_soc_jack_gpio_detect(gpio); +} + +/** + * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * This function will request gpio, set data direction and request irq + * for each gpio in the array. + */ +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i, ret; + + for (i = 0; i < count; i++) { + if (!gpio_is_valid(gpios[i].gpio)) { + printk(KERN_ERR "Invalid gpio %d\n", + gpios[i].gpio); + ret = -EINVAL; + goto undo; + } + if (!gpios[i].name) { + printk(KERN_ERR "No name for gpio %d\n", + gpios[i].gpio); + ret = -EINVAL; + goto undo; + } + + ret = gpio_request(gpios[i].gpio, gpios[i].name); + if (ret) + goto undo; + + ret = gpio_direction_input(gpios[i].gpio); + if (ret) + goto err; + + ret = request_irq(gpio_to_irq(gpios[i].gpio), + gpio_handler, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + jack->card->dev->driver->name, + &gpios[i]); + if (ret) + goto err; + + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + } + + return 0; + +err: + gpio_free(gpios[i].gpio); +undo: + snd_soc_jack_free_gpios(jack, i, gpios); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios); + +/** + * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * Release gpio and irq resources for gpio pins associated with an ASoC jack. + */ +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i; + + for (i = 0; i < count; i++) { + free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpio_free(gpios[i].gpio); + gpios[i].jack = NULL; + } +} +EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios); +#endif /* CONFIG_GPIOLIB */ -- 1.5.4.3

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Re: [alsa-devel] [PATCH] ASoC: wm8510 pll settings
by Mark Brown 04 Mar '09

04 Mar '09

On Wed, Mar 04, 2009 at 08:24:26AM +0100, Jonas Andersson wrote: > When setting WM8510_MCLKDIV the pll was turned off. > > When setting pll frequency you got twice the expected freq, because > the code calculated with postscaler of 8, but the hardware divide by 4. > > Signed-off-by: Jonas Andersson <jonas(a)microbit.se> > Applied, thanks.

1 0

Re: [alsa-devel] No sound for HP Pavilion DV4z
by Takashi Iwai 04 Mar '09

04 Mar '09

At Thu, 26 Feb 2009 00:58:37 +0000 (UTC), aeleneski(a)comcast.net wrote: > > Takashi, Thank you so much. > > Ok, as stated before, this seemed to work. But since then, I have upgraded to > KDE 4.2 and built the latest > alsa snapshot as of yesterday. Things seem to be a little better. > > Now initially just the headphones work, before I could not get them to work at > all. But once I do the hda-verb commands, > headphones do not work at all, but speakers do. With the very latest version, you basically don't need any hda-verb commands. If still it doesn't work as is, give alsa-info.sh output again. (But double-check the mixer status beforehand.) thanks, Takashi

1 0

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 04 Mar '09

04 Mar '09

The following changes since commit ff09d49ad0176a5f52a398c137a7ff5f669d6be4: Daniel Mack (1): ASoC: fix typo and removed unneeded switch case for cs4270 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Philipp Zabel (4): ASoC: Remove version display from the UDA1380 driver ASoC: UDA1380: change decimator/interpolator register handling ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite ASoC: UDA1380: DATAI is slave only sound/soc/codecs/uda1380.c | 113 +++++++++++++++++++++----------------------- sound/soc/pxa/pxa-ssp.c | 3 +- sound/soc/pxa/zylonite.c | 7 ++- 3 files changed, 60 insertions(+), 63 deletions(-)

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Re: [alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Takashi Iwai 04 Mar '09

04 Mar '09

At Tue, 3 Mar 2009 15:18:13 -0200, Emilio López wrote: > > Hi, > > 'Side' is the subwoofer on my laptop (my laptop has an integrated subwoofer, > believe it or not. Hmm, then it's a wrong assignment. "Side" shouldn't be LFE. > Acer calls it "Tuba cinebass" or something like that). > Apart from that, the PC has a headphones/SPDIF jack, so I believe it supports > 7.1? No, it's totally irrelevant. (SPDIF is 2-channel only, BTW. Multi-channel is done via compressed formats, but the physical channels are only two.) Takashi > > 2009/3/3 Takashi Iwai <tiwai(a)suse.de> > > At Mon, 2 Mar 2009 21:46:04 -0200, > Emilio López wrote: > > > > 2009/3/2 Takashi Iwai <tiwai(a)suse.de> > > > > At Sat, 28 Feb 2009 14:38:10 -0200, > > Emilio López wrote: > > > > > > Hello, > > > > > > This is my first mail to the list, I hope you can help me. I have > been > > > talking with people on #alsa, but they couldn't help me, so that's > why > > > I'm writing to you. > > > > > > Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel > > > Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". > With > > > this card, and without adding any mode=... to > > > /etc/modprobe.d/alsa-base, playback works perfectly, but when I > plug > > > in headphones, the integrated speakers do not mute. I can mute the > > > speakers muting "front" and "side" and use the headphones, but it > > > should be automatic. > > > > > > Setting mode=auto makes the headphones, when plugged, mute the > > > speakers. But the problem with this is, that when using the > speakers, > > > I only get sound from the speaker that was "front" in the other > mode. > > > The microphone does not work in this mode either. > > > > > > I also tried other modes, including (but not limited to) acer, > > > acer-aspire. I ran alsa-info.sh, here is the URL: > > > http://www.alsa-project.org/db/?f= > > 7910da439e8e190a8c2a6f18167d4879c74881c9 > > > > > > And before I forget, regarding the microphone, if I plug an > external > > > one on the microphone plug, I can record. And I tried the internal > one > > > on windows, and it works, so I believe it's an alsa problem. > > > > > > Looking forward to hearing from you! > > > > Could you check the very latest sound git tree or alsa-driver > snapshot > > below, first without model option? > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > alsa-driver-snapshot.tar.gz > > > > Hi, > > > > I tried with the alsa-driver snapshot without any model option, and now > the > > headphones mute the speakers. But the internal microphone doesn't still > work, > > and 'Side' doesn't output sound like it did before, even if it's unmuted > and > > with volume at 100% (with the old version and without any model option, > it > > worked) > > What "Side" volume would do exactly? It's for 7.1 output. > Does the machine have 7.1 output? > > Takashi > >

2 4

[alsa-devel] S3C64XX: Add s3c64xx-i2s support
by Ben Dooks 04 Mar '09

04 Mar '09

From: Ben Dooks <ben(a)simtec.co.uk> Add the initial code to support the S3C64XX I2S hardware using the s3c-i2s-v2 core code. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> Index: linux.git/sound/soc/s3c24xx/Makefile =================================================================== --- linux.git.orig/sound/soc/s3c24xx/Makefile 2009-03-04 00:44:28.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/Makefile 2009-03-04 00:56:40.000000000 +0000 @@ -2,6 +2,7 @@ snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o @@ -9,6 +10,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o # S3C24XX Machine Support Index: linux.git/sound/soc/s3c24xx/s3c64xx-i2s.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/s3c64xx-i2s.c 2009-03-04 00:58:09.000000000 +0000 @@ -0,0 +1,223 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * Based on s3c24xx-i2s.c and s3c-i2c-v2.c + * (c) 2006 Wolfson Microelectronics PLC. + * + * 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 + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/regs-s3c2412-iis.h> +#include <plat/gpio-bank-d.h> +#include <plat/gpio-bank-e.h> +#include <plat/gpio-cfg.h> +#include <plat/audio.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { + [0] = { + .channel = DMACH_I2S0_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, + .dma_size = 4, + }, +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { + [0] = { + .channel = DMACH_I2S0_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, + .dma_size = 4, + }, +}; + +static struct s3c_i2sv2_info s3c64xx_i2s[2]; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + switch (clk_id) { + case S3C64XX_CLKSRC_PCLK: + iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + break; + + case S3C64XX_CLKSRC_MUX: + iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + + return 0; +} + + +unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + return clk_get_rate(i2s->iis_cclk); +} +EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate); + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct device *dev = &pdev->dev; + struct s3c_i2sv2_info *i2s; + int ret; + + dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); + + if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { + dev_err(dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + + ret = s3c_i2sv2_probe(pdev, dai, i2s, + pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); + if (ret) + return ret; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + i2s->iis_cclk = clk_get(dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(dev, "failed to get audio-bus"); + iounmap(i2s->regs); + return -ENODEV; + } + + /* configure GPIO for i2s port */ + switch (pdev->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +#define S3C64XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai s3c64xx_i2s_dai = { + .name = "s3c64xx-i2s", + .id = 0, + .probe = s3c64xx_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .ops = { + .set_sysclk = s3c64xx_i2s_set_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static int __init s3c64xx_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c64xx_i2s_dai); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); + + + Index: linux.git/sound/soc/s3c24xx/s3c64xx-i2s.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/s3c64xx-i2s.h 2009-03-04 00:44:31.000000000 +0000 @@ -0,0 +1,31 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * 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 + * published by the Free Software Foundation. + */ + +#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK (0) +#define S3C64XX_CLKSRC_MUX (1) + +extern struct snd_soc_dai s3c64xx_i2s_dai; + +extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai); + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ Index: linux.git/sound/soc/s3c24xx/Kconfig =================================================================== --- linux.git.orig/sound/soc/s3c24xx/Kconfig 2009-03-04 00:44:28.000000000 +0000 +++ linux.git/sound/soc/s3c24xx/Kconfig 2009-03-04 00:56:40.000000000 +0000 @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 + depends on ARCH_S3C2410 || ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,6 +16,10 @@ config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + config SND_S3C2443_SOC_AC97 tristate select AC97_BUS Index: linux.git/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h =================================================================== --- linux.git.orig/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h 2009-03-04 00:44:10.000000000 +0000 +++ linux.git/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h 2009-03-04 00:44:31.000000000 +0000 @@ -33,6 +33,9 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) +#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) + #define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) #define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) #define S3C2412_IISMOD_SLAVE (2 << 10) -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

1 0

[alsa-devel] [PATCH] ASoC: Add GPIO support for jack reporting interface
by Lopez Cruz, Misael 03 Mar '09

03 Mar '09

Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpio, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins for a particular jack can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc.h | 27 +++++++++++ sound/soc/soc-jack.c | 123 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 150 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 68d8149..ad0466b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include <linux/platform_device.h> #include <linux/types.h> #include <linux/workqueue.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> @@ -168,6 +170,7 @@ struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; +struct snd_soc_jack_gpio; typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); @@ -194,6 +197,9 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack); /* codec IO */ #define snd_soc_read(codec, reg) codec->read(codec, reg) @@ -262,10 +268,31 @@ struct snd_soc_jack_pin { bool invert; }; +/** + * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection + * + * @gpio: gpio number + * @name: gpio name + * @report: value to report when jack detected + * @invert: report presence in low state + * @debouce_time: debouce time in ms + */ +struct snd_soc_jack_gpio { + unsigned int gpio; + const char *name; + int report; + int invert; + int debounce_time; + struct snd_soc_jack *jack; + struct work_struct work; +}; + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; struct list_head pins; + struct snd_soc_jack_gpio *gpios; + int gpio_count; int status; }; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3..0e9ee28 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,10 @@ #include <sound/jack.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> +#include <linux/delay.h> /** * snd_soc_jack_new - Create a new jack @@ -136,3 +140,122 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return 0; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); + +/* gpio detect */ +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +{ + struct snd_soc_jack *jack = gpio->jack; + int enable; + int report; + + if (gpio->debounce_time > 0) + mdelay(gpio->debounce_time); + + enable = gpio_get_value(gpio->gpio); + if (gpio->invert) + enable != enable; + + if (enable) + report = gpio->report; + else + report = 0; + + snd_soc_jack_report(jack, report, gpio->report); +} + +/* irq handler for gpio pin */ +static irqreturn_t gpio_handler(int irq, void *data) +{ + struct snd_soc_jack_gpio *gpio = data; + + return IRQ_RETVAL(schedule_work(&gpio->work)); +} + +/* gpio work */ +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio; + + gpio = container_of(work, struct snd_soc_jack_gpio, work); + snd_soc_jack_gpio_detect(gpio); +} + +/** + * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * This function will request gpio, set data direction and request irq + * for each gpio in the array. Valid gpio pins references will be saved. + */ +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i, ret = 0; + + /* Link gpio array with the soc_jack */ + if (count > 0) + jack->gpios = gpios; + + for (i = 0; i < count; i++) { + if (!gpio_is_valid(gpios[i].gpio)) { + printk(KERN_ERR "Invalid gpio %d\n", + gpios[i].gpio); + return -EINVAL; + } + if (!gpios[i].name) { + printk(KERN_ERR "No name for gpio %d\n", + gpios[i].gpio); + return -EINVAL; + } + + ret = gpio_request(gpios[i].gpio, gpios[i].name); + if (ret) + return ret; + + ret = gpio_direction_input(gpios[i].gpio); + if (ret) + break; + + ret = request_irq(gpio_to_irq(gpios[i].gpio), + gpio_handler, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + jack->card->dev->driver->name, + &gpios[i]); + if (ret) + break; + + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + } + + if (ret) + gpio_free(gpios[i].gpio); + + jack->gpio_count = i; /* gpios allocated properly */ + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios); + +/** + * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * + * Release gpio and irq resources for gpio pins associated with an ASoC jack. + */ +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack) +{ + struct snd_soc_jack_gpio *gpios = jack->gpios; + int i; + + for (i = 0; i < jack->gpio_count; i++) { + free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpio_free(gpios[i].gpio); + } +} +EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios); -- 1.5.4.3

3 7

[alsa-devel] PCM File plugin - minor fix
by Pavel Hofman 03 Mar '09

03 Mar '09

Hi, After some consideration I decided to ask for changing the meaning of the %b key from bytes to bits as it provides more convenience. Thanks for commiting the patch. Pavel Hofman.

2 1

[PATCH] ALSA: Rewrite hw_ptr updaters
by Takashi Iwai 03 Mar '09

03 Mar '09

Clean up and improve snd_pcm_update_hw_ptr*() functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 125 +++++++++++++++++++++++++++++++++----------------- 1 files changed, 82 insertions(+), 43 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9216910..7d13eee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -125,19 +125,23 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG +#define xrun_debug(substream) ((substream)->pstr->xrun_debug) +#else +#define xrun_debug(substream) 0 +#endif + static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (substream->pstr->xrun_debug) { + if (xrun_debug(substream)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p'); - if (substream->pstr->xrun_debug > 1) + if (xrun_debug(substream) > 1) dump_stack(); } -#endif } static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, @@ -182,11 +186,26 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream return 0; } +static void hw_ptr_error(struct snd_pcm_substream *substream, + const char *fmt, ...) +{ + if (xrun_debug(substream)) { + if (printk_ratelimit()) { + va_list args; + va_start(args, fmt); + vprintk(fmt, args); + va_end(args); + if (xrun_debug(substream) > 1) + dump_stack(); + } + } +} + static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; + snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt, hw_base; snd_pcm_sframes_t delta; pos = snd_pcm_update_hw_ptr_pos(substream, runtime); @@ -194,36 +213,47 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs xrun(substream); return -EPIPE; } - if (runtime->period_size == runtime->buffer_size) - goto __next_buf; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - - delta = hw_ptr_interrupt - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 1 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif - return 0; + delta = new_hw_ptr - hw_ptr_interrupt; + if (hw_ptr_interrupt == runtime->boundary) + hw_ptr_interrupt = 0; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value " + "(stream=%i, pos=%ld, intr_ptr=%ld)\n", + substream->stream, (long)pos, + (long)hw_ptr_interrupt); + /* rebase to interrupt position */ + hw_base = new_hw_ptr = hw_ptr_interrupt; + delta = 0; + } else { + hw_base += runtime->buffer_size; + if (hw_base == runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; } - __next_buf: - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; } - + if (delta > runtime->period_size) { + hw_ptr_error(substream, + "Lost interrupts? " + "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + substream->stream, (long)delta, + (long)hw_ptr_interrupt); + /* rebase hw_ptr_interrupt */ + hw_ptr_interrupt = + new_hw_ptr - new_hw_ptr % runtime->period_size; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size; + runtime->hw_ptr_interrupt = hw_ptr_interrupt; return snd_pcm_update_hw_ptr_post(substream, runtime); } @@ -233,7 +263,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t delta; old_hw_ptr = runtime->status->hw_ptr; @@ -242,29 +272,38 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - new_hw_ptr = runtime->hw_ptr_base + pos; - - delta = old_hw_ptr - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 2 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; + + delta = new_hw_ptr - old_hw_ptr; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value [2] " + "(stream=%i, pos=%ld, old_ptr=%ld)\n", + substream->stream, (long)pos, + (long)old_hw_ptr); return 0; } - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base += runtime->buffer_size; + if (hw_base == runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + if (delta > runtime->period_size && runtime->periods > 1) { + hw_ptr_error(substream, + "hw_ptr skipping! " + "(pos=%ld, delta=%ld, period=%ld)\n", + (long)pos, (long)delta, + (long)runtime->period_size); + return 0; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; return snd_pcm_update_hw_ptr_post(substream, runtime); -- 1.6.1.3

1 0

Re: [alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Takashi Iwai 03 Mar '09

03 Mar '09

At Mon, 2 Mar 2009 21:46:04 -0200, Emilio López wrote: > > 2009/3/2 Takashi Iwai <tiwai(a)suse.de> > > At Sat, 28 Feb 2009 14:38:10 -0200, > Emilio López wrote: > > > > Hello, > > > > This is my first mail to the list, I hope you can help me. I have been > > talking with people on #alsa, but they couldn't help me, so that's why > > I'm writing to you. > > > > Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel > > Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". With > > this card, and without adding any mode=... to > > /etc/modprobe.d/alsa-base, playback works perfectly, but when I plug > > in headphones, the integrated speakers do not mute. I can mute the > > speakers muting "front" and "side" and use the headphones, but it > > should be automatic. > > > > Setting mode=auto makes the headphones, when plugged, mute the > > speakers. But the problem with this is, that when using the speakers, > > I only get sound from the speaker that was "front" in the other mode. > > The microphone does not work in this mode either. > > > > I also tried other modes, including (but not limited to) acer, > > acer-aspire. I ran alsa-info.sh, here is the URL: > > http://www.alsa-project.org/db/?f= > 7910da439e8e190a8c2a6f18167d4879c74881c9 > > > > And before I forget, regarding the microphone, if I plug an external > > one on the microphone plug, I can record. And I tried the internal one > > on windows, and it works, so I believe it's an alsa problem. > > > > Looking forward to hearing from you! > > Could you check the very latest sound git tree or alsa-driver snapshot > below, first without model option? > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > alsa-driver-snapshot.tar.gz > > Hi, > > I tried with the alsa-driver snapshot without any model option, and now the > headphones mute the speakers. But the internal microphone doesn't still work, > and 'Side' doesn't output sound like it did before, even if it's unmuted and > with volume at 100% (with the old version and without any model option, it > worked) What "Side" volume would do exactly? It's for 7.1 output. Does the machine have 7.1 output? Takashi

1 0

[alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Eric Miao 03 Mar '09

03 Mar '09

Considering the fact that most cpu_dai or codec_dai are using a same 'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better made a pointer instead, to make sharing easier and code a bit cleaner. The patch below is rather preliminary since the asoc tree is being actively developed, and this touches almost every piece of code, (and possibly many others in development need to be changed as well). Building of all codecs are OK, yet to every SoC, I didn't test that. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 24247f7..1367647 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -203,7 +203,7 @@ struct snd_soc_dai { int (*resume)(struct snd_soc_dai *dai); /* ops */ - struct snd_soc_dai_ops ops; + struct snd_soc_dai_ops *ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ff0054b..e588e63 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops atmel_ssc_dai_ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv, +}; + struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { { .name = "atmel-ssc0", .id = 0, @@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[0], }, #if NUM_SSC_DEVICES == 3 @@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[1], }, { .name = "atmel-ssc2", @@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[2], }, #endif diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index f0e30ae..479d7bd 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", .ac97_control = 1, @@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, - }, + .ops = &au1xpsc_ac97_dai_ops, }; EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index f916de4..bb58932 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", .probe = au1xpsc_i2s_probe, @@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .channels_min = 2, .channels_max = 8, /* 2 without external help */ }, - .ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, - }, + .ops = &au1xpsc_i2s_dai_ops, }; EXPORT_SYMBOL(au1xpsc_i2s_dai); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index d1d95d2..9648244 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { + .startup = bf5xx_i2s_startup, + .shutdown = bf5xx_i2s_shutdown, + .hw_params = bf5xx_i2s_hw_params, + .set_fmt = bf5xx_i2s_set_dai_fmt, +}; + struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, @@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = { .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, - .ops = { - .startup = bf5xx_i2s_startup, - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, - }, + .ops = &bf5xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(bf5xx_i2s_dai); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index fb53e65..7c0f412 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ac97_dai_ops = { + .prepare = ac97_prepare, +}; + struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .ac97_control = 1, @@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = { .channels_max = 2, .rates = STD_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &ac97_dai_ops, }; EXPORT_SYMBOL_GPL(ac97_dai); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 81300d8..e81526a 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -436,6 +436,13 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ak4535_dai_ops = { + .hw_params = ak4535_hw_params, + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, +}; + struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { @@ -450,12 +457,7 @@ struct snd_soc_dai ak4535_dai = { .channels_max = 2, .rates = AK4535_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = ak4535_hw_params, - .set_fmt = ak4535_set_dai_fmt, - .digital_mute = ak4535_mute, - .set_sysclk = ak4535_set_dai_sysclk, - }, + .ops = &ak4535_dai_ops, }; EXPORT_SYMBOL_GPL(ak4535_dai); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1aa0c3..c6e155c 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -635,6 +635,17 @@ error: #endif /* USE_I2C*/ +#ifdef USE_I2C +static struct snd_soc_dai_ops cs4270_dai_ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + .digital_mute = cs4270_mute, +#endif +}; +#endif + struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { @@ -708,15 +719,10 @@ static int cs4270_probe(struct platform_device *pdev) } /* Did we find a CS4270 on the I2C bus? */ - if (codec->control_data) { + if (codec->control_data) /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.ops.digital_mute = cs4270_mute; -#endif - } else + cs4270_dai.ops = &cs4270_dai_ops; + else printk(KERN_INFO "cs4270: no I2C device found, " "using stand-alone mode\n"); #else diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cac3736..a577ea1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -521,6 +521,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops ssm2602_dai_ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, +}; + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -535,15 +545,7 @@ struct snd_soc_dai ssm2602_dai = { .channels_max = 2, .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, - .ops = { - .startup = ssm2602_startup, - .prepare = ssm2602_pcm_prepare, - .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, - .digital_mute = ssm2602_mute, - .set_sysclk = ssm2602_set_dai_sysclk, - .set_fmt = ssm2602_set_dai_fmt, - } + .ops = &ssm2602_dai_ops, }; EXPORT_SYMBOL_GPL(ssm2602_dai); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cfdea00..daa5b88 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -598,6 +598,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, #define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops tlv320aic23_dai_ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, +}; + struct snd_soc_dai tlv320aic23_dai = { .name = "tlv320aic23", .playback = { @@ -612,14 +621,7 @@ struct snd_soc_dai tlv320aic23_dai = { .channels_max = 2, .rates = AIC23_RATES, .formats = AIC23_FORMATS,}, - .ops = { - .prepare = tlv320aic23_pcm_prepare, - .hw_params = tlv320aic23_hw_params, - .shutdown = tlv320aic23_shutdown, - .digital_mute = tlv320aic23_mute, - .set_fmt = tlv320aic23_set_dai_fmt, - .set_sysclk = tlv320aic23_set_dai_sysclk, - } + .ops = &tlv320aic23_dai_ops, }; EXPORT_SYMBOL_GPL(tlv320aic23_dai); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 29f2f1a..8c6b15b 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) +static struct snd_soc_dai_ops aic26_dai_ops = { + .hw_params = aic26_hw_params, + .digital_mute = aic26_mute, + .set_sysclk = aic26_set_sysclk, + .set_fmt = aic26_set_fmt, +}; + struct snd_soc_dai aic26_dai = { .name = "tlv320aic26", .playback = { @@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = { .rates = AIC26_RATES, .formats = AIC26_FORMATS, }, - .ops = { - .hw_params = aic26_hw_params, - .digital_mute = aic26_mute, - .set_sysclk = aic26_set_sysclk, - .set_fmt = aic26_set_fmt, - }, + .ops = &aic26_dai_ops, }; EXPORT_SYMBOL_GPL(aic26_dai); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749..e24170a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1069,6 +1069,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops aic3x_dai_ops = { + .hw_params = aic3x_hw_params, + .digital_mute = aic3x_mute, + .set_sysclk = aic3x_set_dai_sysclk, + .set_fmt = aic3x_set_dai_fmt, +}; + struct snd_soc_dai aic3x_dai = { .name = "tlv320aic3x", .playback = { @@ -1083,12 +1090,7 @@ struct snd_soc_dai aic3x_dai = { .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, - .ops = { - .hw_params = aic3x_hw_params, - .digital_mute = aic3x_mute, - .set_sysclk = aic3x_set_dai_sysclk, - .set_fmt = aic3x_set_dai_fmt, - } + .ops = &aic3x_dai_ops, }; EXPORT_SYMBOL_GPL(aic3x_dai); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a2c5064..e75a2f9 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -464,6 +464,15 @@ static int uda134x_add_controls(struct snd_soc_codec *codec) return 0; } +static struct snd_soc_dai_ops uda134x_dai_ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, +}; + struct snd_soc_dai uda134x_dai = { .name = "UDA134X", /* playback capabilities */ @@ -483,14 +492,7 @@ struct snd_soc_dai uda134x_dai = { .formats = UDA134X_FORMATS, }, /* pcm operations */ - .ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, - .hw_params = uda134x_hw_params, - .digital_mute = uda134x_mute, - .set_sysclk = uda134x_set_dai_sysclk, - .set_fmt = uda134x_set_dai_fmt, - } + .ops = &uda134x_dai_ops, }; EXPORT_SYMBOL(uda134x_dai); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e6bf084..bdd830f 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -544,6 +544,21 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops uda1380_dai_ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, +}; + +static struct snd_soc_dai_ops uda1380_dai_ops_capture = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .set_fmt = uda1380_set_dai_fmt, +}; + struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", @@ -559,13 +574,7 @@ struct snd_soc_dai uda1380_dai[] = { .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops, }, { /* playback only - dual interface */ .name = "UDA1380", @@ -576,13 +585,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops, }, { /* capture only - dual interface*/ .name = "UDA1380", @@ -593,12 +596,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(uda1380_dai); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 35d9975..8b9e64b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,6 +1436,16 @@ static int wm8350_remove(struct platform_device *pdev) SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8350_dai_ops = { + .hw_params = wm8350_pcm_hw_params, + .digital_mute = wm8350_mute, + .trigger = wm8350_pcm_trigger, + .set_fmt = wm8350_set_dai_fmt, + .set_sysclk = wm8350_set_dai_sysclk, + .set_pll = wm8350_set_fll, + .set_clkdiv = wm8350_set_clkdiv, +}; + struct snd_soc_dai wm8350_dai = { .name = "WM8350", .playback = { @@ -1452,15 +1462,7 @@ struct snd_soc_dai wm8350_dai = { .rates = WM8350_RATES, .formats = WM8350_FORMATS, }, - .ops = { - .hw_params = wm8350_pcm_hw_params, - .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, - .set_fmt = wm8350_set_dai_fmt, - .set_sysclk = wm8350_set_dai_sysclk, - .set_pll = wm8350_set_fll, - .set_clkdiv = wm8350_set_clkdiv, - }, + .ops = &wm8350_dai_ops, }; EXPORT_SYMBOL_GPL(wm8350_dai); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 40f8238..85e7dd7 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -570,6 +570,14 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops wm8510_dai_ops = { + .hw_params = wm8510_pcm_hw_params, + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, +}; + struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { @@ -584,13 +592,7 @@ struct snd_soc_dai wm8510_dai = { .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, - .ops = { - .hw_params = wm8510_pcm_hw_params, - .digital_mute = wm8510_mute, - .set_fmt = wm8510_set_dai_fmt, - .set_clkdiv = wm8510_set_dai_clkdiv, - .set_pll = wm8510_set_dai_pll, - }, + .ops = &wm8510_dai_ops, }; EXPORT_SYMBOL_GPL(wm8510_dai); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d004e58..4e343db 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -785,6 +785,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops wm8580_dai_ops_playback = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, + .digital_mute = wm8580_digital_mute, +}; + +static struct snd_soc_dai_ops wm8580_dai_ops_capture = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, +}; + struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", @@ -796,13 +811,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - .digital_mute = wm8580_digital_mute, - }, + .ops = &wm8580_dai_ops_playback, }, { .name = "WM8580 PAIFTX", @@ -814,12 +823,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - }, + .ops = &wm8580_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(wm8580_dai); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 80b1198..d22ee46 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -259,6 +259,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8728_dai_ops = { + .hw_params = wm8728_hw_params, + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, +}; + struct snd_soc_dai wm8728_dai = { .name = "WM8728", .playback = { @@ -268,11 +274,7 @@ struct snd_soc_dai wm8728_dai = { .rates = WM8728_RATES, .formats = WM8728_FORMATS, }, - .ops = { - .hw_params = wm8728_hw_params, - .digital_mute = wm8728_mute, - .set_fmt = wm8728_set_dai_fmt, - } + .ops = &wm8728_dai_ops, }; EXPORT_SYMBOL_GPL(wm8728_dai); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c444b9f..97f0851 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -446,6 +446,15 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { @@ -460,14 +469,7 @@ struct snd_soc_dai wm8731_dai = { .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, }; EXPORT_SYMBOL_GPL(wm8731_dai); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 5997fa6..35de8e6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -694,6 +694,13 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8750_dai_ops = { + .hw_params = wm8750_pcm_hw_params, + .digital_mute = wm8750_mute, + .set_fmt = wm8750_set_dai_fmt, + .set_sysclk = wm8750_set_dai_sysclk, +}; + struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { @@ -708,12 +715,7 @@ struct snd_soc_dai wm8750_dai = { .channels_max = 2, .rates = WM8750_RATES, .formats = WM8750_FORMATS,}, - .ops = { - .hw_params = wm8750_pcm_hw_params, - .digital_mute = wm8750_mute, - .set_fmt = wm8750_set_dai_fmt, - .set_sysclk = wm8750_set_dai_sysclk, - }, + .ops = &wm8750_dai_ops, }; EXPORT_SYMBOL_GPL(wm8750_dai); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 77620ab..2f291c5 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1316,6 +1316,51 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1h_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode2_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", @@ -1332,14 +1377,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode1, }, /* DAI Voice mode 1 */ { .name = "WM8753 Voice", @@ -1356,14 +1394,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode1, }, /* DAI HiFi mode 2 - dummy */ { .name = "WM8753 HiFi", @@ -1384,14 +1415,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode2, }, /* DAI HiFi mode 3 */ { .name = "WM8753 HiFi", @@ -1408,14 +1432,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode3, }, /* DAI Voice mode 3 - dummy */ { .name = "WM8753 Voice", @@ -1436,14 +1453,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode4, }, /* DAI Voice mode 4 - dummy */ { .name = "WM8753 Voice", diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6767de1..589924d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1104,6 +1104,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm8900_dai_ops = { + .hw_params = wm8900_hw_params, + .set_clkdiv = wm8900_set_dai_clkdiv, + .set_pll = wm8900_set_dai_pll, + .set_fmt = wm8900_set_dai_fmt, + .digital_mute = wm8900_digital_mute, +}; + struct snd_soc_dai wm8900_dai = { .name = "WM8900 HiFi", .playback = { @@ -1120,13 +1128,7 @@ struct snd_soc_dai wm8900_dai = { .rates = WM8900_RATES, .formats = WM8900_PCM_FORMATS, }, - .ops = { - .hw_params = wm8900_hw_params, - .set_clkdiv = wm8900_set_dai_clkdiv, - .set_pll = wm8900_set_dai_pll, - .set_fmt = wm8900_set_dai_fmt, - .digital_mute = wm8900_digital_mute, - }, + .ops = &wm8900_dai_ops, }; EXPORT_SYMBOL_GPL(wm8900_dai); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bde7454..f8f19d8 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1512,6 +1512,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8903_dai_ops = { + .startup = wm8903_startup, + .shutdown = wm8903_shutdown, + .hw_params = wm8903_hw_params, + .digital_mute = wm8903_digital_mute, + .set_fmt = wm8903_set_dai_fmt, + .set_sysclk = wm8903_set_dai_sysclk, +}; + struct snd_soc_dai wm8903_dai = { .name = "WM8903", .playback = { @@ -1528,14 +1537,7 @@ struct snd_soc_dai wm8903_dai = { .rates = WM8903_CAPTURE_RATES, .formats = WM8903_FORMATS, }, - .ops = { - .startup = wm8903_startup, - .shutdown = wm8903_shutdown, - .hw_params = wm8903_hw_params, - .digital_mute = wm8903_digital_mute, - .set_fmt = wm8903_set_dai_fmt, - .set_sysclk = wm8903_set_dai_sysclk - } + .ops = &wm8903_dai_ops, }; EXPORT_SYMBOL_GPL(wm8903_dai); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 88ead7f..826d1c9 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -619,6 +619,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8971_dai_ops = { + .hw_params = wm8971_pcm_hw_params, + .digital_mute = wm8971_mute, + .set_fmt = wm8971_set_dai_fmt, + .set_sysclk = wm8971_set_dai_sysclk, +}; + struct snd_soc_dai wm8971_dai = { .name = "WM8971", .playback = { @@ -633,12 +640,7 @@ struct snd_soc_dai wm8971_dai = { .channels_max = 2, .rates = WM8971_RATES, .formats = WM8971_FORMATS,}, - .ops = { - .hw_params = wm8971_pcm_hw_params, - .digital_mute = wm8971_mute, - .set_fmt = wm8971_set_dai_fmt, - .set_sysclk = wm8971_set_dai_sysclk, - }, + .ops = &wm8971_dai_ops, }; EXPORT_SYMBOL_GPL(wm8971_dai); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 1cbb7b9..e0141d8 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1345,6 +1345,15 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ +static struct snd_soc_dai_ops wm8990_dai_ops = { + .hw_params = wm8990_hw_params, + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, +}; + struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", @@ -1361,14 +1370,7 @@ struct snd_soc_dai wm8990_dai = { .channels_max = 2, .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, - .ops = { - .hw_params = wm8990_hw_params, - .digital_mute = wm8990_mute, - .set_fmt = wm8990_set_dai_fmt, - .set_clkdiv = wm8990_set_dai_clkdiv, - .set_pll = wm8990_set_dai_pll, - .set_sysclk = wm8990_set_dai_sysclk, - }, + .ops = &wm8990_dai_ops, }; EXPORT_SYMBOL_GPL(wm8990_dai); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index af83d62..684e94e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -532,6 +532,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops wm9712_dai_ops_hifi = { + .prepare = ac97_prepare, +}; + +static struct snd_soc_dai_ops wm9712_dai_ops_aux = { + .prepare = ac97_aux_prepare, +}; + struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", @@ -548,8 +556,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &wm9712_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -559,8 +566,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 1, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare,}, + .ops = &wm9712_dai_ops_aux, } }; EXPORT_SYMBOL_GPL(wm9712_dai); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f3ca8aa..ec8f303 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1021,6 +1021,27 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm9713_dai_ops_hifi = { + .prepare = ac97_hifi_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_aux = { + .prepare = ac97_aux_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_voice = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, +}; + struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", @@ -1037,10 +1058,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_hifi_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -1050,10 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 1, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_aux, }, { .name = "WM9713 Voice", @@ -1069,14 +1084,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, - .ops = { - .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll, - .set_fmt = wm9713_set_dai_fmt, - .set_tristate = wm9713_set_dai_tristate, - }, + .ops = &wm9713_dai_ops_voice, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 0fee779..ffdb943 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, +}; + struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, @@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = { .channels_max = 2, .rates = DAVINCI_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, - }, + .ops = &davinci_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(davinci_i2s_dai); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6d6eb7..104d5f9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -563,6 +563,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ +static struct snd_soc_dai_ops fsl_ssi_dai_ops = { + .startup = fsl_ssi_startup, + .prepare = fsl_ssi_prepare, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, +}; + static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ @@ -577,14 +586,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, - .ops = { - .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, - .shutdown = fsl_ssi_shutdown, - .trigger = fsl_ssi_trigger, - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, - }, + .ops = &fsl_ssi_dai_ops, }; /** diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 05dd5ab..d6882be 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } +static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { + .startup = omap_mcbsp_dai_startup, + .shutdown = omap_mcbsp_dai_shutdown, + .trigger = omap_mcbsp_dai_trigger, + .hw_params = omap_mcbsp_dai_hw_params, + .set_fmt = omap_mcbsp_dai_set_dai_fmt, + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, +}; + #define OMAP_MCBSP_DAI_BUILDER(link_id) \ { \ .name = "omap-mcbsp-dai-"#link_id, \ @@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ - .ops = { \ - .startup = omap_mcbsp_dai_startup, \ - .shutdown = omap_mcbsp_dai_shutdown, \ - .trigger = omap_mcbsp_dai_trigger, \ - .hw_params = omap_mcbsp_dai_hw_params, \ - .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ - }, \ + .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 724e8fb..1242d68 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -766,6 +766,19 @@ static void pxa_ssp_remove(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops pxa_ssp_dai_ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, +}; + struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp1", @@ -786,18 +799,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp2", .id = 1, @@ -817,18 +819,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp3", @@ -849,18 +840,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp4", @@ -881,18 +861,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, }; EXPORT_SYMBOL_GPL(pxa_ssp_dai); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a4a655f..a7b8a3f 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -164,6 +164,10 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops pxa_ac97_dai_ops = { + .hw_params = pxa2xx_ac97_hw_params, +}; + /* * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. @@ -189,8 +193,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 2, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_params,}, + .ops = &pxa_ac97_dai_ops, }, { .name = "pxa2xx-ac97-aux", @@ -208,8 +211,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_aux_params,}, + .ops = &pxa_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -221,8 +223,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_mic_params,}, + .ops = &pxa_ac97_dai_ops, }, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 223de89..bd63a86 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -335,6 +335,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .startup = pxa2xx_i2s_startup, + .shutdown = pxa2xx_i2s_shutdown, + .trigger = pxa2xx_i2s_trigger, + .hw_params = pxa2xx_i2s_hw_params, + .set_fmt = pxa2xx_i2s_set_dai_fmt, + .set_sysclk = pxa2xx_i2s_set_dai_sysclk, +}; + struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, @@ -350,14 +359,7 @@ struct snd_soc_dai pxa_i2s_dai = { .channels_max = 2, .rates = PXA2XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = pxa2xx_i2s_startup, - .shutdown = pxa2xx_i2s_shutdown, - .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params, - .set_fmt = pxa2xx_i2s_set_dai_fmt, - .set_sysclk = pxa2xx_i2s_set_dai_sysclk, - }, + .ops = &pxa_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(pxa_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3fc0ab..382d7ee 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -708,6 +708,14 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .trigger = s3c2412_i2s_trigger, + .hw_params = s3c2412_i2s_hw_params, + .set_fmt = s3c2412_i2s_set_fmt, + .set_clkdiv = s3c2412_i2s_set_clkdiv, + .set_sysclk = s3c2412_i2s_set_sysclk, +}; + struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, @@ -726,13 +734,7 @@ struct snd_soc_dai s3c2412_i2s_dai = { .rates = S3C2412_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .trigger = s3c2412_i2s_trigger, - .hw_params = s3c2412_i2s_hw_params, - .set_fmt = s3c2412_i2s_set_fmt, - .set_clkdiv = s3c2412_i2s_set_clkdiv, - .set_sysclk = s3c2412_i2s_set_sysclk, - }, + .ops = &s3c2412_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 5822d2d..83ea623 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -355,6 +355,11 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { + .hw_params = s3c2443_ac97_hw_params, + .trigger = s3c2443_ac97_trigger, +}; + struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", @@ -374,9 +379,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 2, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger}, + .ops = &s3c2443_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -388,9 +391,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 1, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger,}, + .ops = &s3c2443_ac97_dai_ops, }, }; EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6f4d439..82d8c6e 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -456,6 +456,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, @@ -472,13 +480,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .channels_max = 2, .rates = S3C24XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, - }, + .ops = &s3c24xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index d1e5390..56fa087 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) +static struct snd_soc_dai_ops ssi_dai_ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, +}; + struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", @@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #ifdef CONFIG_CPU_SUBTYPE_SH7760 { @@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #endif }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55fdb4a..a0113f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream, cpu_dai); + if (cpu_dai->ops->startup) { + ret = cpu_dai->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream, codec_dai); + if (codec_dai->ops->startup) { + ret = codec_dai->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -247,8 +247,8 @@ codec_dai_err: platform->pcm_ops->close(substream); platform_err: - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream, codec_dai); + if (codec_dai->ops->shutdown) + codec_dai->ops->shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream, codec_dai); + if (codec_dai->ops->prepare) { + ret = codec_dai->ops->prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; } } - if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream, cpu_dai); + if (cpu_dai->ops->prepare) { + ret = cpu_dai->ops->prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params, codec_dai); + if (codec_dai->ops->hw_params) { + ret = codec_dai->ops->hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); + if (cpu_dai->ops->hw_params) { + ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -526,12 +526,12 @@ out: return ret; platform_err: - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->pcm_ops->hw_free(substream); /* now free hw params for the DAI's */ - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; - if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd, codec_dai); + if (codec_dai->ops->trigger) { + ret = codec_dai->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); + if (cpu_dai->ops->trigger) { + ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 1); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 1); } /* suspend all pcms */ @@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 0); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 0); } for (i = 0; i < card->num_links; i++) { @@ -2020,8 +2020,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops.set_sysclk) - return dai->ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops->set_sysclk) + return dai->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -2040,8 +2040,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops.set_clkdiv) - return dai->ops.set_clkdiv(dai, div_id, div); + if (dai->ops->set_clkdiv) + return dai->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -2059,8 +2059,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops.set_pll) - return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops->set_pll) + return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -2075,8 +2075,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops.set_fmt) - return dai->ops.set_fmt(dai, fmt); + if (dai->ops->set_fmt) + return dai->ops->set_fmt(dai, fmt); else return -EINVAL; } @@ -2094,8 +2094,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->ops.set_sysclk) - return dai->ops.set_tdm_slot(dai, mask, slots); + if (dai->ops->set_sysclk) + return dai->ops->set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -2110,8 +2110,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops.set_sysclk) - return dai->ops.set_tristate(dai, tristate); + if (dai->ops->set_sysclk) + return dai->ops->set_tristate(dai, tristate); else return -EINVAL; } @@ -2126,8 +2126,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops.digital_mute) - return dai->ops.digital_mute(dai, mute); + if (dai->ops->digital_mute) + return dai->ops->digital_mute(dai, mute); else return -EINVAL; } @@ -2180,6 +2180,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +static struct snd_soc_dai_ops null_dai_ops = { +}; + /** * snd_soc_register_dai - Register a DAI with the ASoC core * @@ -2194,6 +2197,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai) if (!dai->dev) printk(KERN_WARNING "No device for DAI %s\n", dai->name); + if (!dai->ops) + dai->ops = &null_dai_ops; + INIT_LIST_HEAD(&dai->list); mutex_lock(&client_mutex);

5 6

[alsa-devel] [PATCH 1/3] ASoC: Add GPIO support for jack reporting interface
by Lopez Cruz, Misael 03 Mar '09

03 Mar '09

Add GPIO support for jack reporting framework in ASoC, by using gpiolib calls. It's only required to append GPIO information (gpio number, irq handler and flags) to standard jack_pin declaration. GPIO request, data direction configuration and irq request are handled by the utility. The minimal GPIO information that can be provided is the gpio number, in that case default handler/flags will be used. Default handler queues a work that reads the current value in the gpio pin and informs to the jack framework. If the GPIO support is not required, the "gpio" field ot jack_pin structure must be set to NO_JACK_PIN_GPIO. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc.h | 15 ++++++++++ sound/soc/soc-jack.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 83 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 68d8149..846e2c1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include <linux/platform_device.h> #include <linux/types.h> #include <linux/workqueue.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> @@ -254,14 +256,27 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @gpio: gpio number associated to the pin (gpiolib calls will be used) + * @irqflags IRQ flags + * @handler: handler for servicing interrupt events on gpio pin */ struct snd_soc_jack_pin { + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio *gpio_pin; struct list_head list; const char *pin; int mask; bool invert; + /* GPIO */ + unsigned int gpio; + unsigned int irq; + unsigned long irqflags; + irq_handler_t handler; + struct work_struct work; }; +#define NO_JACK_PIN_GPIO UINT_MAX + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3..0d048b2 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,9 @@ #include <sound/jack.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> /** * snd_soc_jack_new - Create a new jack @@ -96,6 +99,32 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +/* Default IRQ handler for a GPIO jack pin, it will queue a + * work that reads current value in GPIO pin and reports it + * to the jack framework. + */ +static irqreturn_t gpio_interrupt(int irq, void *data) +{ + struct snd_soc_jack_pin *pin = data; + + return IRQ_RETVAL(schedule_work(&pin->work)); +} + +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_pin *pin; + int report; + + pin = container_of(work, struct snd_soc_jack_pin, work); + report = pin->jack->status & pin->mask; + if (gpio_get_value(pin->gpio)) + report |= pin->mask; + else + report &= ~pin->mask; + + snd_soc_jack_report(pin->jack, report, pin->jack->jack->type); +} + /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * @@ -106,11 +135,18 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * After this function has been called the DAPM pins specified in the * pins array will have their status updated to reflect the current * state of the jack whenever the jack status is updated. + * + * A GPIO pin (using gpiolib) can be used to detect events. It requieres + * an IRQ handler and flags to be set in jack_pin structure; if they are + * not provided, default handler/flags will be used instead. If this + * feature is not desired, "gpio" field of jack_pin structure must be + * set to NO_JACK_PIN_GPIO. */ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins) { - int i; + unsigned int gpio = 0; + int i, ret = 0; for (i = 0; i < count; i++) { if (!pins[i].pin) { @@ -123,6 +159,32 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return -EINVAL; } + if (pins[i].gpio != NO_JACK_PIN_GPIO) { + pins[i].jack = jack; + gpio = pins[i].gpio; + ret = gpio_request(gpio, pins[i].pin); + if (ret) + return ret; + + ret = gpio_direction_input(gpio); + if (ret) + goto out; + pins[i].irq = gpio_to_irq(gpio); + /* If none set, use the default handler */ + if (!pins[i].handler) { + pins[i].handler = gpio_interrupt; + pins[i].irqflags = IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING; + INIT_WORK(&pins[i].work, gpio_work); + } + ret = request_irq(pins[i].irq, + pins[i].handler, + pins[i].irqflags, + jack->card->dev->driver->name, + &pins[i]); + if (ret) + goto out; + } INIT_LIST_HEAD(&pins[i].list); list_add(&(pins[i].list), &jack->pins); } @@ -133,6 +195,10 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, */ snd_soc_jack_report(jack, 0, 0); - return 0; +out: + if (ret) + gpio_free(gpio); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); -- 1.5.4.3

3 11

[alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Emilio López 03 Mar '09

03 Mar '09

Hello, This is my first mail to the list, I hope you can help me. I have been talking with people on #alsa, but they couldn't help me, so that's why I'm writing to you. Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". With this card, and without adding any mode=... to /etc/modprobe.d/alsa-base, playback works perfectly, but when I plug in headphones, the integrated speakers do not mute. I can mute the speakers muting "front" and "side" and use the headphones, but it should be automatic. Setting mode=auto makes the headphones, when plugged, mute the speakers. But the problem with this is, that when using the speakers, I only get sound from the speaker that was "front" in the other mode. The microphone does not work in this mode either. I also tried other modes, including (but not limited to) acer, acer-aspire. I ran alsa-info.sh, here is the URL: http://www.alsa-project.org/db/?f=7910da439e8e190a8c2a6f18167d4879c74881c9 And before I forget, regarding the microphone, if I plug an external one on the microphone plug, I can record. And I tried the internal one on windows, and it works, so I believe it's an alsa problem. Looking forward to hearing from you! Emilio

2 2

Re: [alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Timur Tabi 02 Mar '09

02 Mar '09

Eric Miao wrote: > sound/soc/fsl/fsl_ssi.c | 18 ++++--- I think you forgot sound/soc/fsl/mpc5200_psc_i2s.c -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] [PATCH 0/4] ASoC updates for 2.6.30
by Mark Brown 02 Mar '09

02 Mar '09

The following changes since commit 6d5643455ced9ee45a4aa7403fe8056d826bde85: Takashi Iwai (1): ASoC: wm8753 - Fix build error are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Dooks (1): ASoC: Fix copyright statements on Simtec files Daniel Mack (2): ASoC: Add cs4270 support for slave mode configurations ASoC: fix typo and removed unneeded switch case for cs4270 Mark Brown (1): ASoC: Add SND_SOC_DAPM_PIN_SWITCH controls for exposing DAPM pins include/sound/soc-dapm.h | 12 +++++++ sound/soc/codecs/cs4270.c | 22 ++++++++++++- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- sound/soc/soc-dapm.c | 70 +++++++++++++++++++++++++++++++++++++++ 5 files changed, 105 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH 1/4] ASoC: Fix copyright statements on Simtec files
by Mark Brown 02 Mar '09

02 Mar '09

From: Ben Dooks <ben(a)simtec.co.uk> Fix the copyright statements in two of the S3C24XX ASoC files that have (c) when we require the full word. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6f4d439..1c2b054 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -4,7 +4,7 @@ * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com * - * (c) 2004-2005 Simtec Electronics + * Copyright 2004-2005 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks <ben(a)simtec.co.uk> * diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7c64d31..ba1ae09 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -4,7 +4,7 @@ * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com * - * (c) 2004-2005 Simtec Electronics + * Copyright 2004-2005 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks <ben(a)simtec.co.uk> * -- 1.6.1.3

1 3

[alsa-devel] Analog Loopback controls disabled for IDT/STAC codecs
by Takashi Iwai 02 Mar '09

02 Mar '09

Hi, I disabled now "Analog Loopback" controls for IDT/STAC codecs on sound git tree as default since these are rather confusing for users, more harmful than useful. But the feature is still available. It can be enabled via sysfs "hints". For example, # echo "loopback = yes" > /sys/class/sound/hwC0D0/hints # echo 1 > /sys/class/sound/hwC0D0/reconfig will take the mixer elements back. Also, hp_detect and gpio-bits can be changed similarly on the fly. If you find any problems with this change, please let me know. thanks, Takashi

1 0

[alsa-devel] [PATCH] cs4270: add support for slave mode configurations
by Daniel Mack 02 Mar '09

02 Mar '09

Added support for scenarios where the Cirrus CS4270 audio codec is slave to the bitclk and lrclk. Mixed setups are unsupported. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/codecs/cs4270.c | 24 +++++++++++++++++++++++- 1 files changed, 23 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index cd4a9ee..339e0f6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -109,6 +109,7 @@ struct cs4270_private { u8 reg_cache[CS4270_NUMREGS]; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ + unsigned int slave_mode; }; /** @@ -247,6 +248,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, struct cs4270_private *cs4270 = codec->private_data; int ret = 0; + /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -257,6 +259,21 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, ret = -EINVAL; } + /* set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4270->slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4270->slave_mode = 0; + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* unsupported - cs4270 can eigther be slave or master to + * both the bitclk and the lrclk. */ + default: + ret = -EINVAL; + } + return ret; } @@ -399,7 +416,12 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); - reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + reg |= cs4270_mode_ratios[i].mclk; + + if (cs4270->slave_mode) + reg |= CS4270_MODE_SLAVE; + else + reg |= cs4270_mode_ratios[i].speed_mode; ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { -- 1.6.1.3

4 11

[alsa-devel] Support for fujitsu T421x
by John Kha 02 Mar '09

02 Mar '09

I have a Fujitsu Lifebook T4210 (very similar to the T4215) and wish to work on adding support for these models in patch_sigmatel.c. I have ran the alsa-info script, once after a fresh boot, and once after changing the model a few times: Virgin: http://www.alsa-project.org/db/?f=e76430dbf61b0e2c1ff199c9d1c85051a0341508 With Model Changed: http://www.alsa-project.org/db/?f=cdd45e343c143052e18f9706e65eef8865696606 After trying various models, I have been unable to get the audio to output to the correct pin groups. I have managed to use hda_analyzer.py to change some of my hardware settings, and have an idea of what verbs i need to send. What I need is some information on the architecture of the driver, or an idea of where to edit things. I would just say the config here and ask someone else to edit patch_sigmatel.c, but this is only the first of the issues I want to fix, so some knowledge of how to edit myself would be useful. -- John C. Kha United States Navy

2 1

[alsa-devel] arecord channel limitation
by Tim Blechmann 02 Mar '09

02 Mar '09

hi all, is there any reason, why arecord limits the channel count to 32? best, tim -- tim(a)klingt.org http://tim.klingt.org Desperation is the raw material of drastic change. Only those who can leave behind everything they have ever believed in can hope to escape. William S. Burroughs

2 1

[alsa-devel] Status of the alsa sbxfi driver
by Brandon Ehle 02 Mar '09

02 Mar '09

I merged the topic/sbxfi branch from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git into my latest 2.6.29-rc6 build. I've done a few quick tests and I have not run into any issues so far (except the driver has an excessive amount of kprintfs). Is this the latest version of the sbxfi driver? _From following the alsa-devel list and the git tree above, there doesn't seem to be any changes since Oct 30 or so. What work needs to be done to get this driver into a state where it could be pushed to the main alsa git and eventually to Linus' tree?

2 1

[alsa-devel] [PATCH] Arcam AV Control Plugin Update
by Peter Stokes 02 Mar '09

02 Mar '09

arcam_av: Refactor event callback to facilitate addition of "numid" optimisation Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk>

2 1

[alsa-devel] [PATCH] opl3sa2 fix irq releasing and short name of card
by Krzysztof Helt 02 Mar '09

02 Mar '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Two simple fixes: 1. Use the same pointer for the free_irq() and the request_irq() calls. 2. A short name of card is appended with '2' or '3' character depending on a detected chip. Remove the '2' character from the short name. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The short name without the patch is "OPL3-SA23" for opl3-sa3. sound/isa/opl3sa2.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 58c972b..9a53ab4 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -613,7 +613,7 @@ static void snd_opl3sa2_free(struct snd_card *card) { struct snd_opl3sa2 *chip = card->private_data; if (chip->irq >= 0) - free_irq(chip->irq, (void *)chip); + free_irq(chip->irq, card); release_and_free_resource(chip->res_port); } @@ -626,7 +626,7 @@ static struct snd_card *snd_opl3sa2_card_new(int dev) if (card == NULL) return NULL; strcpy(card->driver, "OPL3SA2"); - strcpy(card->shortname, "Yamaha OPL3-SA2"); + strcpy(card->shortname, "Yamaha OPL3-SA"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; -- 1.5.2.2 ---------------------------------------------------------------------- 169 stacji TV! Naziemne i satelitarne! Sprawdz program TV w kilka sekund >> http://link.interia.pl/f2050

2 1

[alsa-devel] pointer callback question
by Tim Blechmann 28 Feb '09

28 Feb '09

hi all, one question about the pointer callback. the docs say: > This callback is called when the PCM middle layer inquires the current > hardware position on the buffer. The position must be returned in frames, > ranging from 0 to buffer_size - 1. i am using a ringbuffer to communicate with my audio device. but when the position is reset to 0, alsa requests to stop the stream (SNDRV_PCM_TRIGGER_STOP). the brief log is (4 periods of 64 frames): stream_pointer at 64 stream_pointer at 128 stream_pointer at 192 stream_pointer at 256 stream_pointer at 0 after returning 0, a stop is requested. am i missing something? thanks, tim -- tim(a)klingt.org http://tim.klingt.org Contrary to general belief, an artist is never ahead of his time but most people are far behind theirs. Edgar Varèse

2 7

[alsa-devel] Xonar U1 volume controls
by Elie Morisse 28 Feb '09

28 Feb '09

Hi! I'm answering to cladisch call for the "lsusb -v" ouput for this device on Asus forums: http://vip.asus.com/forum/view.aspx?id=20090105132008643&board_id=21&model=… us The analog ouput and microphone input work flawlessly with alsa-driver-1.0.19 but the volume controls don't, although they are listed in evtest pushing or rotating the knob(X not running) does nothing : $ sudo evtest /dev/input/event2 Input driver version is 1.0.0 Input device ID: bus 0x3 vendor 0xb05 product 0x1743 version 0x100 Input device name: "C-Media Electronics Inc. USB Advanced Audio Device" Supported events: Event type 0 (Reset) Event code 0 (Reset) Event code 1 (Key) Event code 4 (?) Event type 1 (Key) Event code 113 (Mute) Event code 114 (VolumeDown) Event code 115 (VolumeUp) Event type 4 (?) Event code 4 (?) Testing ... (interrupt to exit) Besides it also has a LED that can "pulse" and can be blue, purple or red, and there doesn't seem to be any way to control this either. On Vista the Asus driver defaults it to pulsing blue when ouputting sound while on Linux it blinks alternatively purple and blue, how ugly and less relaxing!.. Hope the lsusb output will be useful and thanks a lot for all your hard work, Elie

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[alsa-devel] E-MU 0404 Support
by Kristian Lumme 27 Feb '09

27 Feb '09

Hi! I reported a bug (or what I think is a bug) with the E-MU 0404 USB driver (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4410) (this probably concerns other E-MU cards as well). Now, the supported soundcards list states that E-MU (or Creative) have supplied ALSA developers with datasheets to help the development of the driver. Well... in my bug report I was referred to this discussion on the ALSA devel list: http://thread.gmane.org/gmane.linux.alsa.devel/56288/focus=56355 If I understand this correctly, people that are offering to help develop the driver can't get in touch with the person who has the datasheets (and there hasn't been any update from him in a long time... or so it seems--sorry if I'm mistaken). Surely, there's some way to make these specifications available to interested developers so the support for the E-MU cards can be improved?

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Re: [alsa-devel] [PATCH] hda_intel: Add the Device IDs for MCP89 and remove IDs of MCP7B to/from hda_intel.c
by Takashi Iwai 27 Feb '09

27 Feb '09

At Fri, 27 Feb 2009 17:03:19 +0800, peerchen wrote: > > Added the Device IDs for MCP89 HD audio controller. > Removed the IDs of MCP7B cause this chipset had been cancelled. > > Signed-off-by: Peer Chen <peerchen(a)gmail.com> Applied now. Thanks! Takashi > --- > --- linux-2.6.29-rc6/sound/pci/hda/hda_intel.c.orig 2009-02-26 08:51:02.000000000 -0500 > +++ linux-2.6.29-rc6/sound/pci/hda/hda_intel.c 2009-02-26 08:51:23.000000000 -0500 > @@ -2442,10 +2442,10 @@ static struct pci_device_id azx_ids[] = > { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, > { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, > { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, > - { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, > - { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, > - { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, > - { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, > + { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, > + { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, > + { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, > + { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, > /* Teradici */ > { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, > /* AMD Generic, PCI class code and Vendor ID for HD Audio */ > - >

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 26 Feb '09

26 Feb '09

The following changes since commit ce3bdaa8710c10eec5a6dae67aaf73088d0ced4f: Mark Brown (1): ASoC: Disable WM8731 line bypass by default are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (7): ASoC: Report I/O errors from WM8753 reset ASoC: Only unregister drivers we registered for WM8753 ASoC: Improve WM9713 voice DAC shutdown procedure ASoC: Fix Zylonite voice interface stereo configurations ASoC: Shuffle WM8753 device registration code ASoC: Convert WM8753 to register via normal device probe ASoC: Only write back non-default registers when resuming WM8753 sound/soc/codecs/wm8753.c | 421 +++++++++++++++++------------------- sound/soc/codecs/wm8753.h | 6 - sound/soc/codecs/wm9713.c | 7 +- sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/neo1973_wm8753.c | 6 - 5 files changed, 210 insertions(+), 235 deletions(-)

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[alsa-devel] [PATCH 3/3] ASoC: Add headset jack detection for SDP3430 machine driver
by Lopez Cruz, Misael 26 Feb '09

26 Feb '09

Add headset jack detection for SDP3430 boards using SoC jack reporting interface. Headset detection on SDP3430 board is achieved through TWL4030 GPIO_2 pin. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 31 ++++++++++++++++++++++++++++++- 1 files changed, 30 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 7a996d9..7695675 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -141,6 +142,18 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +/* Jack detection */ +struct snd_soc_jack *sdp3430_jack; + +struct snd_soc_jack_pin sdp3430_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + .invert = 0, + .gpio = (OMAP_MAX_GPIO_LINES + 2), + }, +}; + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", @@ -191,7 +204,23 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - return 0; + /* Jack detection */ + sdp3430_jack = kzalloc(sizeof(struct snd_soc_jack), GFP_KERNEL); + if (!sdp3430_jack) { + printk(KERN_ERR "SDP3430 SoC Jack allocation failed\n"); + return -ENOMEM; + } + + snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 SoC Jack", + SND_JACK_HEADSET, sdp3430_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(sdp3430_jack, + ARRAY_SIZE(sdp3430_jack_pins), + sdp3430_jack_pins); + + return ret; err1: printk(KERN_ERR "Unable to add platform device\n"); -- 1.5.4.3

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[alsa-devel] [PATCH 2/3] ASoC: Add DAPM machine widgets to SDP3430 driver
by Lopez Cruz, Misael 26 Feb '09

26 Feb '09

Add DAPM machine domain widgets to SDP3430 machine driver. Current interconection: * Ext Mic: MAINMIC, SUBMIC * Ext Spk: HFL, HFR * Headset Jack: HSMIC, HSOL, HSOR Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 61 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 61 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e226fa7..7a996d9 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -81,12 +81,73 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +/* SDP3430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC */ + {"MAINMIC", NULL, "Ext Mic"}, + {"SUBMIC", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset: HSMIC, HSOL, HSOR */ + {"Headset Jack", NULL, "HSOL"}, + {"Headset Jack", NULL, "HSOR"}, + {"HSMIC", NULL, "Headset Jack"}, +}; + +static int sdp3430_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* SDP3430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AUXL"); + snd_soc_dapm_nc_pin(codec, "AUXR"); + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + /* Add SDP3430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets, + ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP3430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai, + .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- 1.5.4.3

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[alsa-devel] [PATCH 0/3] GPIO for jack reporting interface
by Lopez Cruz, Misael 26 Feb '09

26 Feb '09

The following patch set implements GPIO support for SoC jack reporting interface and uses it in SDP3430 machine driver. The jack detection interface is used in SDP3430 driver to handle headset events. That requires DAPM machine widgets to be added to the machine driver before the actual use of the interface. ASoC: Add GPIO support for jack reporting interface ASoC: Add DAPM machine widgets to SDP3430 driver ASoC: Add headset jack detection for SDP3430 machine driver

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Re: [alsa-devel] [PATCH] ASoC: Add support for OMAP3 EVM
by Peter Ujfalusi 26 Feb '09

26 Feb '09

On Wednesday 25 February 2009 22:25:59 ext George G. Davis wrote: > Hi, > > On Tue, Nov 25, 2008 at 02:52:58PM +0000, Mark Brown wrote: > > On Tue, Nov 25, 2008 at 01:21:20PM +0200, Jarkko Nikula wrote: > > > Or if those EVM's & SDP's can route TWL4030 audio connections more > > > flexible than Beagle but somewhat similar manner, then probably have > > > one single machine driver for all EVM's? > > > > > > Otherwise it doesn't make very much sense to have n similar machine > > > drivers where only functions and variable names differ. > > > > Yes, it looks like a lot of these drivers could be redone along the > > lines of s3c24xx_uda134x.c with platform data specifying the differences > > between the boards. > > Is someone working on updating ASoC drivers for OMAP TWL4030 based > boards as recommended above? Just curious since I already made the > mistake of creating my own OMAP3 EVM ASoC driver w/o checking the > list first. : / Well, I have checked the s3c24xx_uda134x.c and the corresponding uda134x.c codec driver. In case of the uda134x codec - since it does not have DAPM implementation, the board file does not need to deal with the output/input pins. In case of boards using the TWL4030 codec it is not that simple IMHO. You see TWL4030 has nine output 'pins' and six (plus 2x stereo digital mic) input 'pins'. So, yes it is doable, but will it make things cleaner and more manageable? Few things came to my mind, which needs to be addressed: 1. Who is the master (OMAP McBSP or TWL4030)? 2. What is the reference clock in the given board 3. Connected pins (or not connected pins) 4. I2S or TDM mode (this is not implemented yet, but it is coming at some point) It easy to provide 1, 2 and 4. But when it comes to 3... One needs to build up the DAPM routes in the common board file for the supported boards, which again doable based on - for example - some model/board ID, but I think it will look ugly. Or I might over complicate things ;) > -- > Regards, > George -- Péter

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Re: [alsa-devel] [PATCH] ASoC: Add support for OMAP3 EVM
by Mark Brown 26 Feb '09

26 Feb '09

On Wed, Feb 25, 2009 at 03:25:59PM -0500, George G. Davis wrote: > Is someone working on updating ASoC drivers for OMAP TWL4030 based > boards as recommended above? Just curious since I already made the > mistake of creating my own OMAP3 EVM ASoC driver w/o checking the > list first. : / Not to my knowledge.

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[alsa-devel] possible recursive locking in snd_pcm_playback_ioctl()
by Chuck Ebbert 26 Feb '09

26 Feb '09

https://bugzilla.redhat.com/show_bug.cgi?id=486729 [ INFO: possible recursive locking detected ] 2.6.29-0.137.rc5.git4.fc11.x86_64 #1 --------------------------------------------- pulseaudio/4381 is trying to acquire lock: (&substream->self_group.lock/1){....}, at: [<ffffffffa0161309>] snd_pcm_action_group+0x53/0x217 [snd_pcm] but task is already holding lock: (&substream->self_group.lock/1){....}, at: [<ffffffffa0161309>] snd_pcm_action_group+0x53/0x217 [snd_pcm] other info that might help us debug this: 4 locks held by pulseaudio/4381: #0: (snd_pcm_link_rwlock){....}, at: [<ffffffffa0160450>] snd_pcm_stream_lock_irq+0x1d/0x2d [snd_pcm] #1: (&substream->self_group.lock){....}, at: [<ffffffffa016045c>] snd_pcm_stream_lock_irq+0x29/0x2d [snd_pcm] #2: (&substream->group->lock){....}, at: [<ffffffffa0161619>] snd_pcm_action+0x33/0x90 [snd_pcm] #3: (&substream->self_group.lock/1){....}, at: [<ffffffffa0161309>] snd_pcm_action_group+0x53/0x217 [snd_pcm] stack backtrace: Pid: 4381, comm: pulseaudio Not tainted 2.6.29-0.137.rc5.git4.fc11.x86_64 #1 Call Trace: [<ffffffff8106faa1>] __lock_acquire+0x83b/0xc0d [<ffffffff8106e500>] ? check_usage_backwards+0x4a/0x53 [<ffffffff8106ff05>] lock_acquire+0x92/0xc0 [<ffffffffa0161309>] ? snd_pcm_action_group+0x53/0x217 [snd_pcm] [<ffffffff81394c10>] _spin_lock_nested+0x34/0x67 [<ffffffffa0161309>] ? snd_pcm_action_group+0x53/0x217 [snd_pcm] [<ffffffffa0161309>] snd_pcm_action_group+0x53/0x217 [snd_pcm] [<ffffffffa016164f>] snd_pcm_action+0x69/0x90 [snd_pcm] [<ffffffffa016045c>] ? snd_pcm_stream_lock_irq+0x29/0x2d [snd_pcm] [<ffffffffa0161690>] snd_pcm_stop+0x1a/0x1c [snd_pcm] [<ffffffffa0161722>] snd_pcm_drop+0x90/0xa6 [snd_pcm] [<ffffffffa01642e9>] snd_pcm_common_ioctl1+0xb76/0xc26 [snd_pcm] [<ffffffff8101781b>] ? native_sched_clock+0x2d/0x5a [<ffffffff810170b5>] ? sched_clock+0x9/0xc [<ffffffff8106d04d>] ? lock_release_holdtime+0x2c/0x11d [<ffffffff81167865>] ? avc_has_perm_noaudit+0x444/0x497 [<ffffffff8116788c>] ? avc_has_perm_noaudit+0x46b/0x497 [<ffffffff81167458>] ? avc_has_perm_noaudit+0x37/0x497 [<ffffffffa01648ed>] snd_pcm_playback_ioctl1+0x282/0x29f [snd_pcm] [<ffffffff8106d04d>] ? lock_release_holdtime+0x2c/0x11d [<ffffffffa016529d>] snd_pcm_playback_ioctl+0x2f/0x33 [snd_pcm] [<ffffffff810ee59b>] vfs_ioctl+0x22/0x87 [<ffffffff810eea6b>] do_vfs_ioctl+0x46b/0x4ac [<ffffffff810eeb02>] sys_ioctl+0x56/0x79 [<ffffffff8101133a>] system_call_fastpath+0x16/0x1b

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[alsa-devel] Rationale behind snd_pcm_htimestamp()
by Lennart Poettering 25 Feb '09

25 Feb '09

Heya! I am interested to query the current delay and the current avail atomically together with the timestamp when they were current. Apparantly there is now a function that can get the former two atomically (snd_pcm_avail_delay()) and another one that gets the latter two atomically (snd_pcm_htimestamp()). _avail_delay() seems to sync the hw index data, _htimestamp() apparently not. Hence I had assume that first calling _avail_delay() and then _htimestamp() one after the other without any further access would return the same avail value. Turns out, my test results on an USB card show that this assumption is not true. An alternative way seemed to be to go via the snd_pcm_status structure. However querying that seems not to sync the hw index data. Hmm... so let's call snd_pcm_hwsync() right before querying the struct -- as it turns out however, that function is now deprecated. So which function should I call? The doxygen comments of _hwsync() kind of suggest in a way to call snd_pcm_avail_update(), whose doxygen comments however declare that it does in fact *not* sync the hw state either, but suggest to use snd_pcm_avail() for that. Ok, so I am now tempted to first call snd_pcm_avail() and then snd_pcm_status() and have all three values atomically. But unfortunately that's not the case. Again the avail value of the two calls differs sometimes. How can I query the three values atomically? Can i do that at all? Also, snd_pcm_hw_params_is_monotonic() always returns 0 on all my devices. I am calling that function after having set both HW and SW params. Kernel is 2.6.27, alsa-libs is 1.0.19. The kernel itself does monotonic/hrtimers just fine. Hmm, last week I started this discussion about "snd_pcm_busy_for()" (aka granularity API). Who can I bribe to get an API like this? Although snd_pcm_busy_for() would suit me much better than just a call to query the granularity, I'd be happy with either... Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Timer instability
by Lennart Poettering 25 Feb '09

25 Feb '09

Heya! As you might now PulseAudio schedules audio by system timers, not by sound card interrupts. Unfortunately there are are couple of issues with the reliability of the timing/buffering information we get from ALSA. For example, on an ens1371 I have tracked down the following issue: I use the full buffer that the sound card provides (64K, i.e. 371ms). I sleep for about 350 ms, then fill up again, and sleep 350ms, and so and so on. Everytime I wake up I check snd_pcm_avail() and write exactly as much as this call tells me to. I also listen to POLLOUT events on the fd, however I call snd_pcm_sw_params_set_period_event(, 0) for it to make sure I actually don't get any real events on that. After a couple of minutes of running this loop I can reliably reproduce that _avail() tells me I should fill the buffer up, I do so, query it again and it shows the buffer is filled up. I then go to sleep but immediately get POLLIN (although I shouldnt, given that I called snd_pcm_sw_params_set_period_event()), wake up again, call snd_pcm_avail() and according to it the buffer ran completely empty. The poll() took less than 1 ms time, and the last thing I did before going to sleep was checking that the buffer was full. So certainly ALSA is lying to me here... The buffer couldn't have run empty in less than 1 ms! The same with real numbers as one example: 1. ... we are woken up 2. _avail() returns 15496 samples (i.e. 350ms to fill up) 3. we gather and write 15496 samples 4. _avail() returns 48 samples, which we consider "filled up enough" to go to sleep 5. we call poll() 6. after < 1ms we are woken up by a POLLIN (which we actually thought we had explicitly disabled) 7. _avail() returns 16448 samples (i.e. 372ms to fill up -- larger than the actual buffer of 16384 samples!) This smells a lot like some kind of overflow to me, given that in every case where I could reproduce this it was possible to 'fix' the issue by substracting the buffer size from the _avail() after the poll(). After that substraction the value started to make sense again. (i.e. in the example above it then becomes 64 samples, which is reasonable after sleep less than 1ms I guess). The stop threshold is set to the boundary here btw. If necessary I could extract a test case for this, but my hope that you guys might have an idea what goes on without a test case, given that this smells so "overflowy" ;-) And of course, how come ALSA triggers POLLIN if I asked it not to? This is alsa-libs 1.0.19 on a 2.6.27 kernel. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] snd-powermac: timeout reaches -1
by Roel Kluin 25 Feb '09

25 Feb '09

If unsuccessful, timeout reaches -1 after the loop. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index f860d39..45a7629 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -35,7 +35,7 @@ snd_pmac_burgundy_busy_wait(struct snd_pmac *chip) int timeout = 50; while ((in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_busy_wait: timeout\n"); }

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Re: [alsa-devel] No sound for HP Pavilion DV4z
by aeleneski＠comcast.net 25 Feb '09

25 Feb '09

Alright. Well, it again seems like a no go. Ran all the commands, still no sound. Upon starting Amarok, I receive the error "xine was unable to initialize any audio drivers." Then also when I do aplay -l I get: **** List of PLAYBACK Hardware Devices **** card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 Attached is the alsa-info text. Thanks again. Andrew ----- Original Message ----- From: "Takashi Iwai" <tiwai(a)suse.de> To: aeleneski(a)comcast.net Cc: alsa-devel(a)alsa-project.org Sent: Wednesday, February 11, 2009 1:34:29 AM GMT -05:00 US/Canada Eastern Subject: Re: [alsa-devel] No sound for HP Pavilion DV4z At Tue, 10 Feb 2009 23:58:19 +0000 (UTC), aeleneski(a)comcast.net wrote: > > Ok, well I tried this: > > The above might be not suffice to give the initialization before the > parser. Try to call hda-verb like the following instead: > > hda-verb /dev/snd/hwC0D0 0x0d 0x71c 0x10 > hda-verb /dev/snd/hwC0D0 0x0d 0x71d 0x11 > hda-verb /dev/snd/hwC0D0 0x0d 0x71e 0x17 > hda-verb /dev/snd/hwC0D0 0x0d 0x71f 0x90 > > then > > echo 1 > /sys/class/sound/hwC0D0/reconfig > > I started my system, changed to probe_only=1. Ran the commands above, and > removed probe_only=1. > Upon shutdown during restart, I again heard the speakers crackle. When the > system came back up, no sound. Ah no, you don't have to restart the system at all. Maybe I wasn't clear enough. The sequence above itself will make your sound system already usable. (and you don't have to probe_only=1 option.) Just use your sound apps after the sequence above after checking the mixer status. If it still doesn't work, run alsa-info and attach the outupt again. Also, show the kernel message, too. Takashi > > ----- Original Message ----- > From: "Takashi Iwai" <tiwai(a)suse.de> > To: aeleneski(a)comcast.net > Cc: alsa-devel(a)alsa-project.org > Sent: Tuesday, February 10, 2009 6:34:02 AM GMT -05:00 US/Canada Eastern > Subject: Re: [alsa-devel] No sound for HP Pavilion DV4z > > At Mon, 09 Feb 2009 18:24:57 +0100, > I wrote: > > > > At Mon, 9 Feb 2009 17:18:59 +0000 (UTC), > > aeleneski(a)comcast.net wrote: > > > > > > Thanks for your time and all these suggestions. Still no luck though. > > > > > > Seems like you might not be able to have two files ending in alsa-base in > > > the /etc/modprobe.d/ folder. That was the reason for the orange speaker > > > indicator. I had > > > alsa-base and backup-alsa-base, for some reason it was grabbing the > settings > > > from the backup. > > > > Ah, yes, module-init-tools doesn't check every backup file suffix. > > Better to remove completely old files. > > > > > Anyway, with probe only, orange sound indicator, when doing aplay -l I get > > > > > > **** List of PLAYBACK Hardware Devices **** > > > card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] > > > Subdevices: 1/1 > > > Subdevice #0: subdevice #0 > > > > With probe_only option, it doesn't make sense to run aplay. This > > option basically means "probe codecs but don't set up for use yet". > > This is useful to check the raw setup by BIOS before overriding or > > modifying by the driver. > > > > One thing we can try is to add the pin config dynamically then > > reconfigure. For example, load the driver with probe_only=1, then > > do the following as root: > > > > echo 0x0d 0x71c 0x10 > /sys/class/sound/hwC0D0/init_verbs > > echo 0x0d 0x71d 0x11 > /sys/class/sound/hwC0D0/init_verbs > > echo 0x0d 0x71e 0x17 > /sys/class/sound/hwC0D0/init_verbs > > echo 0x0d 0x71f 0x90 > /sys/class/sound/hwC0D0/init_verbs > > The above might be not suffice to give the initialization before the > parser. Try to call hda-verb like the following instead: > > hda-verb /dev/snd/hwC0D0 0x0d 0x71c 0x10 > hda-verb /dev/snd/hwC0D0 0x0d 0x71d 0x11 > hda-verb /dev/snd/hwC0D0 0x0d 0x71e 0x17 > hda-verb /dev/snd/hwC0D0 0x0d 0x71f 0x90 > > then > > echo 1 > /sys/class/sound/hwC0D0/reconfig > > Takashi > >

3 9

[alsa-devel] Docking stations
by Lennart Poettering 25 Feb '09

25 Feb '09

Heya, there's now some support to be notified about docking station events in the kernel. It probably would make a lot of sense if PulseAudio would listen for these and reprogram the alsa control stuff to switch over to the audio jacks of the docking station and back. I am wondering if there is any defined scheme for doing this switching? i.e. is there an established name for a mixer element that needs to be switched for this? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

Re: [alsa-devel] Configuring a TWL GPIO pin as an interrupt
by Lopez Cruz, Misael 24 Feb '09

24 Feb '09

> On Sunday 22 February 2009, Lopez Cruz, Misael wrote: > > In the particular case of ALSA SoC, could the machine/board > > driver be a better place to handle all GPIO/IRQ configuration? > > That driver also contains only board specific code. > > It'd be best of the ASoC stuff could sit with all the other > board-specfic init code, in arch/*/mach-*/board-*.c files, > but I understand those interfaces are not yet stable enough > to support that ... that's why they're in sound/soc/*/*.c > files instead. > > In any case ... everything I said still stands. If you're > doing this for ASoC, you'll need some way to pass data to the > ASoC board-specific code from normal board-specific code, > since some of the relevant config data is not static. I think that if I move the platform_device registration from machine driver to board file I can append jack detection information (gpio pin, irq) through "platform_data" of "dev" field in platform_device structure. And then in the "probe" part in ASoC machine driver I can receive it. Could that be correct? Any other better/standard option? > The current ASoC model seems to be biased towards static > configurations. Notice how it's got to create its own > platform_device nodes ... it can't easily use the standard > mechanisms for associating platform_data or archdata with > those nodes, ditto clocks.

2 3

Re: [alsa-devel] [Fwd: HP DV4 1155 SE - No Sound through speakers...]
by Jean-Pierre André 24 Feb '09

24 Feb '09

Hi, > Hi all, I have an HP DV4 1155-SE, I've been unable to get sound working > through the speakers, although it does work through the headhpones. I ran > alsa-.upgrade which downloaded compiled, and installed 1.0.19 - still no > luck. > I have the same IDT codec 92HD71B7X on another HP DV4 model, and I have tracked the problem to be the kernel configuring IRQ 22 through the APIC, but interrupts triggerring IRQ 20 (on kernel 2.6.27 and alsa 1.0.19). It would be interesting if you could confirm this by analyzing /proc/interrupts. The workarounds I know of are : - adding irqpoll to the boot options, - or adding noapic to the boot options, - using mplayer to play music. There are with option "snd-hda-intel model=hp-m4" (this option is for HP DV5/DV7, but HP DV4 is probably different) I have not enough knowledge to tell where the problem lies : kernel, apic, alsa. Regards Jean-Pierre > I ran alsa-info, output here: > http://www.alsa-project.org/db/?f=844b7ee6468067460e17b7fc694e49052072a26f > > I filed a bug, here: > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4374 > > Any ideas? >

4 32

[alsa-devel] Patch for possible race with /proc/asound symlinks in core/info.c
by Brad Bosch 24 Feb '09

24 Feb '09

Jaroslav, I discovered and fixed an ALSA oops in a very old kernel (2.6.5) used in an embedded application and looking at the latest kernels, it's not clear to me that this was ever completely fixed in newer kernels. I'm sending this email so someone who understands the interactions between the USB and ALSA subsystems better than I do can determine if a similar fix might be required in the newer kernels. In the old kernel, the oops often happens when a USB device is unplugged and then a device with the same ID (or the same device) is immediately plugged back in before the old device is completely cleaned up. The included patch resolves the problem in that kernel. The problem was that two cards were creating symbolic links in /proc/asound with the same name (id) but pointing to differen't cards. The proc subsystem apparently allowed this. Later, when one of the entries is removed (using it's name), the wrong entry may be removed, freeing the wrong struct proc_dir_entry and leaving a pointer to the freed memory in the other card's proc_root_link member. Later when that USB device is unplugged and snd_info_card_disconnect() is called, snd_remove_proc_entry() will reference the name member of struct proc_dir_entry which by now may be changed, or even null. It turns out to be trivial to avoid this problem by simply not storing the pointer to the proc_dir_entry as we don't need it anyway to determine the name of the link to remove. Instead, for the case of the symbolic link, we call remove_proc_entry directly using the card id for the link name (just as we did when it was created). I don't know if two symbolic links are ever created in /proc/asound with the same name in the newer kernels or newer ALSA code and in fact, I haven't been able to reproduce the problem there. However, since the patch reduces both the data and code size, and eliminates a potential crash, you may want to apply it anyway. This patch won't apply directly to newer kernels because of minor changes, but it's trivial to apply by hand. Let me know if you need me to re-base the patch. --Brad Index: sound/core/info.c =================================================================== RCS file: /cvs/kernels/kernel_2_6_5/sound/core/info.c,v retrieving revision 1.1.4.1.10.1 diff -u -r1.1.4.1.10.1 info.c --- sound/core/info.c 22 Feb 2006 14:06:25 -0000 1.1.4.1.10.1 +++ sound/core/info.c 20 Feb 2009 02:05:01 -0000 @@ -618,7 +618,6 @@ p = proc_symlink(card->id, snd_proc_root, card->proc_root->name); if (p == NULL) return -ENOMEM; - card->proc_root_link = p; return 0; } @@ -629,10 +628,7 @@ int snd_info_card_free(snd_card_t * card) { snd_assert(card != NULL, return -ENXIO); - if (card->proc_root_link) { - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); - card->proc_root_link = NULL; - } + remove_proc_entry(card->id, snd_proc_root); if (card->proc_root) { snd_info_unregister(card->proc_root); card->proc_root = NULL; Index: include/sound/core.h =================================================================== RCS file: /cvs/kernels/kernel_2_6_5/include/sound/core.h,v retrieving revision 1.1.4.1 diff -u -r1.1.4.1 core.h --- include/sound/core.h 15 Apr 2005 21:45:13 -0000 1.1.4.1 +++ include/sound/core.h 20 Feb 2009 02:05:02 -0000 @@ -152,7 +152,6 @@ snd_info_entry_t *proc_root; /* root for soundcard specific files */ snd_info_entry_t *proc_id; /* the card id */ - struct proc_dir_entry *proc_root_link; /* number link to real id */ struct snd_monitor_file *files; /* all files associated to this card */ struct snd_shutdown_f_ops *s_f_ops; /* file operations in the shutdown state */

2 3

[alsa-devel] Standard mixer control names
by Lennart Poettering 24 Feb '09

24 Feb '09

Heya! I'd like to add functionality to PulseAudio to allow a simplified selection of capture input (i.e. "Mic", "Line-In", ...) and playback outputs ("Headphones", "Line-Out", ...) for ALSA cards. By "simplified" I mean that only one input and output shall be active at a time. Also, some more nonsense items shall be filtered out (i.e. recording from "Phone"). Hence, basically I want to compile a list of available and whitelisted inputs and outputs and allow the user to select exactly one from each. ALSA is making that very hard to implement something like this because every driver seems to wrap input/output selection differently. On one card I have only has a couple of cswitches (snd-es1371). The same one has an enum "Mic Select". Another card has an enum "Input Source", but no cswitches (a HDA chip). The "ControlNames.txt" file in the kernel seems to suggest that there is an element "Capture Source". For playback it seems that some cards have a a headphone switch, and others a headphone slider (which i guess makes sense). Now, the question, how should I implement this? For playback the handling is easy as long as there is only one element to deal with, but what about capture? One option would be to simply go by cswitch and nothing else. Or go by "Input Source" and nothing else. Or combine some form. Now I'd of course prefer if the drivers get fixed to use a single element naming scheme only. Is there any chance to get that? And which one would that be? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

5 7

Re: [alsa-devel] PATCH: Maxdata Favorit 100XS support
by Takashi Iwai 24 Feb '09

24 Feb '09

At Sun, 22 Feb 2009 18:58:45 +0100, Michael Schwingen wrote: > > On Thu, Feb 05, 2009 at 08:28:50AM +0100, Takashi Iwai wrote: > > The patch looks almost good but there are a few issues to fix. > > > > First off, please use a unified diff (diff -u) for a patch. > > Then, rebase your patch to the latest sound git tree[*] or 2.6.29-rc3 > > at least. > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > Some stuff in patch_realtek.c in the latest tree was changed to be > > incompatible from 2.6.28. For example, > > > > > + [ALC260_FAVORIT100] = { > > > + .mixers = { alc260_favorit100_mixer, > > > + alc260_capture_mixer }, > > > > The capture mixer is now created dynamically. Usually you don't need > > pass capture_mixer array here. > > > > > + .init_verbs = { alc260_acer_init_verbs }, > > > > Don't you use alc260_favorit100_init_verbs? > > > > > > Could you fix these and repost? > > Here is the updated patch, against the git version - is the diff format OK > now? Yes, it was merged to sound tree. Sorry, forgot to notify you. Thanks! Takashi

1 0

[alsa-devel] [PATCH 0/2] Arcam AV Control Plugin Updates
by Peter Stokes 24 Feb '09

24 Feb '09

Hi, The following patches provide minor updates to the Arcam AV control plug-in. * [PATCH 1/2] Remove incorrect usage of the static SHM ID variable * [PATCH 2/2] Refactor control specifications to facilitate addition of "numid" optimisation Takashi, I attempted to follow the pattern of your recent commits regarding the "numid" optimisations for external control plug-ins. Hopefully I've provided the correct behavior. Something that am not sure about is whether I should be calling the "snd_ctl_elem_id_set_numid()" function from "ctl_arcam_av.c" "arcam_av_read_event()" either in addition or instead of the calls to the "snd_ctl_elem_id_set_name()" function? Thanks Peter

3 11

[alsa-devel] idt blue jack patch
by Paulo Cavalcanti 24 Feb '09

24 Feb '09

Hi, I received a patch form Fengguang that made my Intel DG45ID (snd-hda-intel, codec IDT 92HD73E1X5) behave the way it should, in my opinion. It just makes the blue jack behave as an input source by default (in alsa 1.0.19 it is an output jack). It is widespread that the green jack is output, the blue input and the pink is mic. Some cards allow reconfiguring the jacks in alsamixer, but the DG45ID already has an orange and black jacks for rear and center speakers, and a digital output. Other people have also raised this problem some time ago: https://fcp.surfsite.org/modules/newbb/viewtopic.php?viewmode=threaded&orde… Fengguang suggested me to direct my request to the development list, so the other developers could decide what to do. I would like to add that I can use the input source and the mic just fine now (the mic could be a bit louder, but it works fine with skype). The patch is very simple and is attached. This would conclude the fix for this card, Takashi started with me some time ago (I just made the tests). Thanks. -- Paulo Roma Cavalcanti LCG - UFRJ

5 31

[alsa-devel] pxassp dma pointer not moving
by Daniel Mack 23 Feb '09

23 Feb '09

Hi, I'm currently implementing support for a board based on a PXA300 with an I2S codec (CS4270) next to it. Things do work in general, including the setup of the SSP engine and clocks, and when using the OSS compat layer, I can see data on the bus. With the native ALSA API, however, the DMA pointer does not move forward but is stuck at position 8. ALSA's core calls pxa2xx_pcm_pointer() a couple of times and eventually gives up on it. Unfortunately, I don't have a Zylonite board for cross-check; could anyone verify that things are not currently broken in a general way? My code base is up-to-date to sound-2.6.git/for-2.6.30. Thanks and best regards, Daniel

3 3

[alsa-devel] PCM delay compensation
by Takashi Iwai 23 Feb '09

23 Feb '09

Hi, the patch below (to the latest sound git tree) adds the extra delay count for USB-audio driver. This change will appear as the return value of snd_pcm_delay(). Could you check whether it's appropriate behavior you've wanted? thanks, Takashi diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 40c5a6f..dcbdc60 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -269,6 +269,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ + snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e61e125..df7c3fa 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -597,14 +597,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { status->avail = snd_pcm_playback_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || - runtime->status->state == SNDRV_PCM_STATE_DRAINING) + runtime->status->state == SNDRV_PCM_STATE_DRAINING) { status->delay = runtime->buffer_size - status->avail; - else + status->delay += runtime->delay; + } else status->delay = 0; } else { status->avail = snd_pcm_capture_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - status->delay = status->avail; + status->delay = status->avail + runtime->delay; else status->delay = 0; } @@ -2423,6 +2424,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, n = snd_pcm_playback_hw_avail(runtime); else n = snd_pcm_capture_avail(runtime); + n += runtime->delay; break; case SNDRV_PCM_STATE_XRUN: err = -EPIPE; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6e70ba4..7d5a103 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -629,6 +629,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += offs; if (subs->hwptr_done >= runtime->buffer_size) subs->hwptr_done -= runtime->buffer_size; + runtime->delay += offs; spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = offs * stride; if (period_elapsed) @@ -638,12 +639,20 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, /* * process after playback data complete - * - nothing to do + * - decrease the delay count again */ static int retire_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { + unsigned long flags; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); return 0; } @@ -1542,6 +1551,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->phase = 0; + runtime->delay = 0; /* clear urbs (to be sure) */ deactivate_urbs(subs, 0, 1);

2 6

[alsa-devel] snd_pcm_hw_params_get_period_size points to __old_ symbol
by Daniel Mack 23 Feb '09

23 Feb '09

Hi, With alsa-lib and alsa-utils cross-compiled for ARM by buildroot (currently version 1.0.19, but earlier versions seem to be equally affected), I encounter the effect that snd_pcm_hw_params_get_period_size() does not write the expected value to the given snd_pcm_uframes_t pointer. In fact, this variable is not written at all. This makes aplay calculate 0 for chunk_bytes in set_params() and then exit with the bogus error message "Not enough memory". I did some tracing and found out that the function called for snd_pcm_hw_params_get_period_size() is in fact __old_snd_pcm_hw_params_get_period_size() which has a different footprint and hence the pointer given to it is leaved untouched. As I don't fully understand all the system behind the symbol names remapping, I'm stuck here. Can anybody reproduce this bug? Daniel

4 9

Re: [alsa-devel] [PATCH] ALS4000, slight mixer improvements
by Takashi Iwai 23 Feb '09

23 Feb '09

At Sun, 22 Feb 2009 20:33:41 +0100, Andreas Mohr wrote: > > Hi all, > > finally got around to submitting this "slightly unimportant" patch. > > ChangeLog: > - add 8kHz / 20 kHz low-pass filter switch control > - add ALS4000 Mono capture route control > - add annotations to specs pages > - improve ALS4000 PM saved regs selection (remove SB dummy register, > add missing ones) > - add some missing ALS4000 register defines > - constify two variables > > Built on 2.6.29-rc5 and playing. Applied now. Thanks. > > Will now try to get HDA settings on Aspire One A110L investigated > (using Takashi's elaborate tools) and corrected > (the non-working microphone thingy, plus some weird random switching/activation > of noise in sound sources at times and stuff). Let me know if it's finished! > Speaking of sound problems, any thoughts about how to end up with > _automatic_, generic ALSA soft volume for those cheap non-hardvol USB sticks? > Short experimentation with that stick was - well - rather short since > hardvol-only is problematic (the ear-piercing kind of "problematic" ;) > and seems to need manual .asoundrc softvol addition. > (don't ask me why I was buying this stick for my A110L in the first place ;) The only problem right now is that the driver might give a real volume control but in a different name. OTOH, softvol itself isn't too bad as default PCM definition. Just editing USB-Audio.conf should suffice... thanks, Takashi

2 3

[alsa-devel] [PATCH] Quirk for Acer Aspire 6530G
by Juan Jesús García de Soria Lucena 23 Feb '09

23 Feb '09

Hi everyone. The Acer Aspire 6530G needs the 4930G "model" for the front mic to work properly. Please apply the attached patch in order to properly support it out of the box. Best regards, Juan Jesus Garcia de Soria -- Dream small if success is enough for you; dream big if you need to change the world.

2 1

[alsa-devel] [PATCH] Fix typos in ALSA Lib documentation
by Patricio Paez 23 Feb '09

23 Feb '09

Signed-off-by: Patricio Paez <pp(a)pp.com.mx> diff --git a/src/seq/seq.c b/src/seq/seq.c index feb9733..7b777a1 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -30,7 +30,7 @@ /*! \page seq Sequencer interface -\section seq_general Genral +\section seq_general General The ALSA sequencer interface is designed to deliver the MIDI-like events between clients/ports. @@ -76,10 +76,10 @@ A client can have one or more ports to communicate between other clients. A port is corresponding to the MIDI port in the case of MIDI device, but in general it is nothing but the access point between other clients. Each port may have capability flags, which specify the read/write -accessbility and subscription permissions of the port. +accessibility and subscription permissions of the port. For creation of a port, call #snd_seq_create_port() -with the appropirate port attribute specified in #snd_seq_port_info_t -reocrd. +with the appropriate port attribute specified in #snd_seq_port_info_t +record. For creating a port for the normal use, there is a helper function #snd_seq_create_simple_port(). An example with this function is like below. @@ -102,7 +102,7 @@ Here, input and output mean input (read) from other clients and output (write) to others, respectively. Since memory pool of each client is independent from others, it avoids such a situation that a client eats the whole events pool -and interfere other clients' responce. +and interfere other clients' response. The all scheduled output events or input events from dispatcher are stored on these pools until delivered to other clients or extracted to user space. @@ -171,7 +171,7 @@ the MIDI events like program, velocity or chorus effects. This application can accept arbitrary MIDI input and send to arbitrary port, just like a Unix pipe application using stdin and stdout files. -We can even connect several filter applictions which work individually +We can even connect several filter applications which work individually in order to process the MIDI events. Subscription can be used for this purpose. The connection between ports can be done also by the "third" client. @@ -199,7 +199,7 @@ All the sequencer events are stored in a sequencer event record, #snd_seq_event_t type. Application can send and receive these event records to/from other clients via sequencer. -An event has several stroage types according to its usage. +An event has several storage types according to its usage. For example, a SYSEX message is stored on the variable length event, and a large synth sample data is delivered using a user-space data pointer. @@ -227,7 +227,7 @@ The type field contains the type of the event (1 byte). The flags field consists of bit flags which describe several conditions of the event (1 byte). -It includes the time-stamp mode, data storage type, and scheduling prority. +It includes the time-stamp mode, data storage type, and scheduling priority. The tag field is an arbitrary tag. This tag can used for removing a distinct event from the event queue via #snd_seq_remove_events(). @@ -240,7 +240,7 @@ The data field is a union of event data. An event can be delivered either on scheduled or direct dispatch mode. On the scheduling mode, an event is once stored on the priority queue and delivered later (or even immediately) to the destination, -whereas on the direct disatch mode, an event is passed to the destination +whereas on the direct dispatch mode, an event is passed to the destination without any queue. For a scheduled delivery, a queue to process the event must exist. @@ -284,7 +284,7 @@ The time stored in an event record is a union of these two different time values. Note that the time format used for real time events is very similar to -timeval struct used for unix system time. +timeval struct used for Unix system time. The absurd resolution of the timestamps allows us to perform very accurate conversions between songposition and real time. Round-off errors can be neglected. @@ -299,7 +299,7 @@ counted from the moment when the queue started. An client that relies on these relative timestamps is the MIDI input port. As each sequencer queue has it's own clock the only way to deliver events at the right time is by using the relative timestamp format. When the event -arrives at the queue it is normalised to absolute format. +arrives at the queue it is normalized to absolute format. The timestamp format is specified in the flag bitfield masked by #SND_SEQ_TIME_STAMP_MASK. @@ -320,7 +320,7 @@ fill the port id of source.port and both client and port of dest field. If an existing address is set to the destination, -the event is simplly delivered to it. +the event is simply delivered to it. When #SND_SEQ_ADDRESS_SUBSCRIBERS is set to the destination client id, the event is delivered to all the clients connected to the source port. @@ -346,7 +346,7 @@ an announcement is sent to subscribers from this port. Some events like SYSEX message, however, need larger data space than the standard data. -For such events, ALSA sequencer provides seveal different data storage types. +For such events, ALSA sequencer provides several different data storage types. The data type is specified in the flag bits masked by #SND_SEQ_EVENT_LENGTH_MASK. The following data types are available: @@ -359,7 +359,7 @@ A macro #snd_seq_ev_set_fixed() is provided to set this type. \par Variable length data SYSEX or a returned error use this type. The actual data is stored on an extra allocated space. -On sequecer kernel, the whole extra-data is duplicated, so that the event +On sequencer kernel, the whole extra-data is duplicated, so that the event can be scheduled on queue. The data contains only the length and the pointer of extra-data. @@ -430,7 +430,7 @@ Note that PPQ cannot be changed while the queue is running. It must be set before the queue is started. On the other hand, in the case of realtime queue, the -time resolution is fixed to nanosecononds. There is, however, +time resolution is fixed to nanoseconds. There is, however, a parameter to change the speed of this queue, called skew. You can make the queue faster or slower by setting the skew value bigger or smaller. In the API, the skew is defined by two values, @@ -488,7 +488,7 @@ special settings. In the above example, the tempo is changed immediately after the buffer is flushed by #snd_seq_drain_output() call. You can schedule the event in a certain queue so that the tempo -change happes at the scheduled time, too. +change happens at the scheduled time, too. \subsection seq_ev_start Starting and stopping a queue @@ -515,7 +515,7 @@ Each ALSA port can have capability flags. The most basic capability flags are #SND_SEQ_PORT_CAP_READ and #SND_SEQ_PORT_CAP_WRITE. The former means that the port allows to send events to other ports, -whereas the latter capability menas +whereas the latter capability means that the port allows to receive events from other ports. You may have noticed that meanings of \c READ and \c WRITE are permissions of the port from the viewpoint of other ports. @@ -536,7 +536,7 @@ Obviously, these flags have no influence if \c READ or \c WRITE> capability is not set. Note that these flags are not necessary if the client subscribes itself -to the spcified port. +to the specified port. For example, when a port makes READ subscription to MIDI input port, this port must have #SND_SEQ_PORT_CAP_WRITE capability, but no #SND_SEQ_PORT_CAP_SUBS_WRITE capability is required. @@ -610,7 +610,7 @@ if #SND_SEQ_PORT_CAP_NO_EXPORT capability is set in either sender or receiver po Assume MIDI input port = 64:0, application port = 128:0, and queue for timestamp = 1 with real-time stamp. -The application port must have capabilty #SND_SEQ_PORT_CAP_WRITE. +The application port must have capability #SND_SEQ_PORT_CAP_WRITE. \code void capture_keyboard(snd_seq_t *seq) { @@ -633,7 +633,7 @@ void capture_keyboard(snd_seq_t *seq) \subsection seq_subs_ex_out Output to MIDI device Assume MIDI output port = 65:1 and application port = 128:0. -The application port must have capabilty #SND_SEQ_PORT_CAP_READ. +The application port must have capability #SND_SEQ_PORT_CAP_READ. \code void subscribe_output(snd_seq_t *seq) { @@ -790,7 +790,7 @@ void event_filter(snd_seq_t *seq, snd_seq_event_t *ev) /** * \brief get identifier of sequencer handle * \param seq sequencer handle - * \return ascii identifier of sequencer handle + * \return ASCII identifier of sequencer handle * * Returns the ASCII identifier of the given sequencer handle. It's the same * identifier specified in snd_seq_open(). @@ -1610,7 +1610,7 @@ void snd_seq_client_info_event_filter_del(snd_seq_client_info_t *info, int event * \param event_type event type to be checked * \return 1 if the event type is present, 0 otherwise * - * Test if the event type is in the filter bitamp of this client_info container. + * Test if the event type is in the filter bitmap of this client_info container. * * \sa snd_seq_get_client_info(), * snd_seq_set_client_info(), diff --git a/src/seq/seqmid.c b/src/seq/seqmid.c index 3b0960d..86a4970 100644 --- a/src/seq/seqmid.c +++ b/src/seq/seqmid.c @@ -317,7 +317,7 @@ int snd_seq_set_client_pool_input(snd_seq_t *seq, size_t size) * \param seq sequencer handle * \return 0 on success or negative error code * - * So far, this works ideically like #snd_seq_drop_output(). + * So far, this works identically like #snd_seq_drop_output(). */ int snd_seq_reset_pool_output(snd_seq_t *seq) { @@ -329,7 +329,7 @@ int snd_seq_reset_pool_output(snd_seq_t *seq) * \param seq sequencer handle * \return 0 on success or negative error code * - * So far, this works ideically like #snd_seq_drop_input(). + * So far, this works identically like #snd_seq_drop_input(). */ int snd_seq_reset_pool_input(snd_seq_t *seq) { -- 1.5.6.3

2 1

[alsa-devel] [PATCH] ALSA: hda - add another MacBook Pro 3,1 SSID
by Luke Yelavich 23 Feb '09

23 Feb '09

Reference: Ubuntu bug #33245 https://bugs.launchpad.net/bugs/332456 Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 169b383..02ef827 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7094,6 +7094,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b3e00: /* iMac 24 Aluminium */ board_config = ALC885_IMAC24; break; + case 0x106b00a0: /* MacBookPro3,1 - Another revision */ case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ -- 1.6.0.4

2 1

[alsa-devel] [PATCH] ALSA: hdsp - poll for iobox
by Tim Blechmann 23 Feb '09

23 Feb '09

sleeping for 2 seconds before checking for the iobox is not enough on some systems. this patch increases the timeout, but polls the card during that time. it thus speeds up the module loading when the card has already been initialized, while being more robust on systems, which require a higher timeout than the predefined 2 seconds. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 26 ++++++++++++++++++++++---- 1 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 44d0c15..a586a80 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -653,7 +653,6 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { - if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); @@ -661,7 +660,27 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp) return -EIO; } return 0; +} +static int hdsp_wait_for_iobox (struct hdsp *hdsp, unsigned int loops, unsigned int delay) +{ + unsigned int i; + + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) + return 0; + + for (i = 0; i != loops; ++i) { + if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) + msleep(delay); + else { + snd_printd ("Hammerfall-DSP: iobox found after %ums!\n", i * delay); + return 0; + } + } + + snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { @@ -5055,10 +5074,9 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; if (!is_9652 && !is_9632) { - /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ - ssleep(2); + /* we wait a maximum of 10 seconds to let freshly inserted cardbus cards do their hardware init */ + err = hdsp_wait_for_iobox(hdsp, 1000, 10); - err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; -- 1.6.1.2

2 1

Re: [alsa-devel] ALSA: fix excessive background noise introduced by OSS emulation rate shrink
by Takashi Iwai 23 Feb '09

23 Feb '09

At Sat, 21 Feb 2009 08:05:04 -0600, Steve Chen wrote: > > > Incorrect variable was used to get the next sample which caused S2 > to be stuck with the same value resulting in loud background noise. > > Signed-off-by: Steve Chen <schen at mvista.com> Oh what a long-living bug. Applied now. Thanks! Takashi > > diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c > index a466443..2fa9299 100644 > --- a/sound/core/oss/rate.c > +++ b/sound/core/oss/rate.c > @@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, > while (dst_frames1 > 0) { > S1 = S2; > if (src_frames1-- > 0) { > - S1 = *src; > + S2 = *src; > src += src_step; > } > if (pos & ~R_MASK) { > >

1 0

Re: [alsa-devel] [patch] aw2: do not grab every saa7146 based device
by Takashi Iwai 23 Feb '09

23 Feb '09

At Sun, 22 Feb 2009 14:42:54 +0200, Anssi Hannula wrote: > > Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not > check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver > grabs them as well. > > According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the > subsystem ids set to 0, the saa7146 default. > > Fix conflicts with DVB devices by checking for subsystem ids = 0 > specifically. > > Signed-off-by: Anssi Hannula <anssi.hannula(a)gmail.com> Thanks! Applied it as is, since this isn't worse than now, at least (although this is no real solution yet...) Takashi > > --- linux-2629-pidff/sound/pci/aw2/aw2-alsa.c.old 2009-02-13 03:47:15.000000000 +0200 > +++ linux-2629-pidff/sound/pci/aw2/aw2-alsa.c 2009-02-22 14:16:20.000000000 +0200 > @@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0 > MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); > > static struct pci_device_id snd_aw2_ids[] = { > - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, > + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, > 0, 0, 0}, > {0} > }; > > > > -- > Anssi Hannula >

1 0

Re: [alsa-devel] pxassp dma pointer not moving
by Liam Girdwood 23 Feb '09

23 Feb '09

On Sun, 2009-02-22 at 23:16 +0100, Daniel Mack wrote: > Hi Liam, > > On Sun, Feb 22, 2009 at 10:07:57PM +0000, Liam Girdwood wrote: > > > I'm currently implementing support for a board based on a PXA300 with an > > > I2S codec (CS4270) next to it. Things do work in general, including the > > > setup of the SSP engine and clocks, and when using the OSS compat layer, > > > I can see data on the bus. With the native ALSA API, however, the DMA > > > pointer does not move forward but is stuck at position 8. ALSA's core > > > calls pxa2xx_pcm_pointer() a couple of times and eventually gives up on > > > it. Unfortunately, I don't have a Zylonite board for cross-check; could > > > anyone verify that things are not currently broken in a general way? > > > > > > My code base is up-to-date to sound-2.6.git/for-2.6.30. > > > > Can't test on my zylonite atm, but static DMA pointers usually mean data > > is not being clocked out of the SSP FIFO. Can you check you are > > supplying a BCLK and LRC to the SSP port (if codec is BCLK/LRC master) > > or have enabled the PXA SSP master mode (if codec slave). > > All the clocks are there, they're only driven by the Codec, and they look > fine. And thus, the CPU DAI is set to SND_SOC_DAIFMT_CBM_CFM. > > What puzzles me is that the whole system works when I cat /dev/urandom > to /dev/dsp (the DMA pointer is moving forward just as expected), only > when using aplay it doesn't do anything. Ah, Please see later email re buildroot. Interestingly someone had exactly the same issue (on OMAP iirc) with buildroot alsa-lib on #alsa-soc last week. Liam

1 0

[alsa-devel] Problem recording from an RME Hammerfall DSP 9632
by Lars Magne Ingebrigtsen 22 Feb '09

22 Feb '09

At least that's what I think I have. lspci says: 00:08.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 97) Playback works just fine, and analogue input works as well. With hdspmixer I'm able to route the analogue input to the SPDIF outputs, and I'm able to control the volume of the playback via amixer: [larsi@rocket-sam /tmp]$ amixer cset numid=5 16,8,32768 numid=5,iface=HWDEP,name='Mixer' ; type=INTEGER,access=rw------,values=3,min=0,max=65536,step=1 : values=0,0,0 $ amixer cset numid=5 17,9,32768 (etc) So playback and routing from inputs to outputs works as expected. However, I'm not able to record anything. $ arecord -t raw -r 44100 -c 2 -f S16_LE | od Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo 0000000 000000 000000 000000 000000 000000 000000 000000 000000 * So I'm getting output from the capture device -- but it's all zeroes. I've tried using amixer to route everything to everything else, but that doesn't have any effect for arecord, either. What am I missing? -- (domestic pets only, the antidote for overdose, milk.) larsi(a)gnus.org * Lars Magne Ingebrigtsen

1 0

[alsa-devel] Distortion in SND_PCM_FORMAT_S32 playback with Intel HDA
by Lasse Kärkkäinen 22 Feb '09

22 Feb '09

This problem only occurs when samples very close to min/max limits is played. Sounds like integer overflows to the other side. My sample data is correct (not overflowing itself), but apparently ALSA has stricter limits. I'm using device hw:Intel, so ALSA shouldn't be doing any sample format conversions. Thus I suspect that this might be a fault (a feature) of the hardware itself. Can anyone confirm this? Playing in 16 bit mode there is no problem and even -32768 seems to be playing properly.

1 0

[alsa-devel] [PATCH 2/2] Arcam AV Control Plugin Updates
by Peter Stokes 21 Feb '09

21 Feb '09

arcam_av: Refactor control specifications to facilitate addition of "numid" optimisation Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk>

1 0

[alsa-devel] [PATCH 1/2] Arcam AV Control Plugin Updates
by Peter Stokes 21 Feb '09

21 Feb '09

arcam_av: Remove incorrect usage of the static SHM ID variable Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk>

1 0

[alsa-devel] set_params:1073: not enough memory
by maarten van es 21 Feb '09

21 Feb '09

Hello all, I am trying to get alsa to work using buildroot to crosscompile to i686-linux-uclibc. I am using the latest git version of alsa-lib and a recent git version of alsa-util. I am stuck with the same error as in the message included below when using aplay. So far I have been able to narrow this down to the function set_params in aplay.c, lines 1070, 1071: chunk_bytes = chunk_size * bits_per_frame / 8; audiobuf = realloc(audiobuf, chunk_bytes); chunk_size is 0 here and obviously the realloc fails. Any ideas on what's going on here? For the record, "aplay -L" lists the correct devices, "aplay -D null test.wav" works ok and the card works correctly using alsa version 1.0.16 on standard suse 11.0. Best regards Maarten van Es Hello all, I'm trying to get alsa 1.0.18 working on a ARM PXA processor and I'm stuck with the following error message: aplay /Sinus.raw -fcd Playing raw data '/Sinus.raw' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1073: not enough memory alsa kernel was compiled with alsa 1.0.16, alsa.conf, located in /usr/share/alsa is the basic: pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } also aplay -l and alsaconf store show the message: ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 However the card shows up /proc/asound/cards and alsamixer is able to access it. Any ideas? Best regards Michael

2 2

[alsa-devel] [PATCH] ALSA: hdsp - whitespace cleanup
by Tim Blechmann 20 Feb '09

20 Feb '09

Impact: remove trailing spaces Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 474 +++++++++++++++++++++++----------------------- 1 files changed, 237 insertions(+), 237 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index a586a80..796b989 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ @@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); 26*3 values are read in ss mode 14*3 in ds mode, with no gap between values */ -#define HDSP_9652_peakBase 7164 +#define HDSP_9652_peakBase 7164 #define HDSP_9652_rmsBase 4096 /* c.f. the hdsp_9632_meters_t struct */ @@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */ #define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */ #define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */ -#define HDSP_SyncRef2 (1<<13) -#define HDSP_SPDIFInputSelect0 (1<<14) -#define HDSP_SPDIFInputSelect1 (1<<15) -#define HDSP_SyncRef0 (1<<16) +#define HDSP_SyncRef2 (1<<13) +#define HDSP_SPDIFInputSelect0 (1<<14) +#define HDSP_SPDIFInputSelect1 (1<<15) +#define HDSP_SyncRef0 (1<<16) #define HDSP_SyncRef1 (1<<17) -#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ +#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ #define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */ #define HDSP_Midi0InterruptEnable (1<<22) #define HDSP_Midi1InterruptEnable (1<<23) @@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_TimecodeSync (1<<27) #define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ #define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */ -#define HDSP_midi0IRQPending (1<<30) +#define HDSP_midi0IRQPending (1<<30) #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) @@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the Multiface still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -460,7 +460,7 @@ struct hdsp { unsigned char qs_in_channels; /* quad speed mode for H9632 */ unsigned char ds_in_channels; unsigned char ss_in_channels; /* different for multiface/digiface */ - unsigned char qs_out_channels; + unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; @@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = { static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ /* Analog */ - 0, 1, 2, 3, 4, 5, 6, 7, + 0, 1, 2, 3, 4, 5, 6, 7, /* ADAT 2 */ - 16, 17, 18, 19, 20, 21, 22, 23, + 16, 17, 18, 19, 20, 21, 22, 23, /* SPDIF */ 24, 25, -1, -1, -1, -1, -1, -1, -1, -1 @@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ - -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; @@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ @@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d static struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, - .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, + .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, .subvendor = PCI_ANY_ID, .subdevice = PCI_ANY_ID, }, /* RME Hammerfall-DSP */ @@ -689,19 +689,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { unsigned long flags; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + snd_printk ("Hammerfall-DSP: loading firmware\n"); hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); return -EIO; } - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - + for (i = 0; i < 24413; ++i) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { @@ -711,7 +711,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { } ssleep(3); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); return -EIO; @@ -724,15 +724,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { #endif hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); - + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); spin_lock_irqsave(&hdsp->lock, flags); snd_hdsp_set_defaults(hdsp); - spin_unlock_irqrestore(&hdsp->lock, flags); + spin_unlock_irqrestore(&hdsp->lock, flags); } - + hdsp->state |= HDSP_FirmwareLoaded; return 0; @@ -741,7 +741,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) @@ -757,7 +757,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); } else { hdsp->io_type = Digiface; - } + } } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) @@ -805,13 +805,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout) -{ +{ int i; /* the fifoStatus registers reports on how many words are available in the command FIFO. */ - + for (i = 0; i < timeout; i++) { if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count) @@ -843,11 +843,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short if (addr >= HDSP_MATRIX_MIXER_SIZE) return -1; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) { /* from martin bjornsen: - + "You can only write dwords to the mixer memory which contain two mixer values in the low and high @@ -866,7 +866,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short hdsp->mixer_matrix[addr] = data; - + /* `addr' addresses a 16-bit wide address, but the address space accessed via hdsp_write uses byte offsets. put another way, addr @@ -875,17 +875,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short to access 0 to 2703 ... */ ad = addr/2; - - hdsp_write (hdsp, 4096 + (ad*4), - (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + + + hdsp_write (hdsp, 4096 + (ad*4), + (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + hdsp->mixer_matrix[addr&0x7fe]); - + return 0; } else { ad = (addr << 16) + data; - + if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT)) return -1; @@ -921,7 +921,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) if (status & HDSP_SPDIFErrorFlag) return 0; - + switch (rate_bits) { case HDSP_spdifFrequency32KHz: return 32000; case HDSP_spdifFrequency44_1KHz: return 44100; @@ -929,13 +929,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) case HDSP_spdifFrequency64KHz: return 64000; case HDSP_spdifFrequency88_2KHz: return 88200; case HDSP_spdifFrequency96KHz: return 96000; - case HDSP_spdifFrequency128KHz: + case HDSP_spdifFrequency128KHz: if (hdsp->io_type == H9632) return 128000; break; - case HDSP_spdifFrequency176_4KHz: + case HDSP_spdifFrequency176_4KHz: if (hdsp->io_type == H9632) return 176400; break; - case HDSP_spdifFrequency192KHz: + case HDSP_spdifFrequency192KHz: if (hdsp->io_type == H9632) return 192000; break; default: @@ -1046,7 +1046,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -1072,35 +1072,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) there is no need for it (e.g. during module initialization). */ - - if (!(hdsp->control_register & HDSP_ClockModeMaster)) { + + if (!(hdsp->control_register & HDSP_ClockModeMaster)) { if (called_internally) { /* request from ctl or card initialization */ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n"); return -1; - } else { + } else { /* hw_param request while in AutoSync mode */ int external_freq = hdsp_external_sample_rate(hdsp); int spdif_freq = hdsp_spdif_sample_rate(hdsp); - + if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n"); else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) - snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); else if (rate != external_freq) { snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n"); return -1; - } - } + } + } } current_rate = hdsp->system_sample_rate; /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -1108,7 +1108,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) if (rate > 96000 && hdsp->io_type != H9632) return -EINVAL; - + switch (rate) { case 32000: if (current_rate > 48000) @@ -1198,7 +1198,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) break; } } - + hdsp->system_sample_rate = rate; return 0; @@ -1264,16 +1264,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output) { if (!snd_rawmidi_transmit_empty (hmidi->output)) { if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]); } } @@ -1344,14 +1344,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data) { struct hdsp_midi *hmidi = (struct hdsp_midi *) data; unsigned long flags; - + snd_hdsp_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1520,7 +1520,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif); return 0; } @@ -1530,7 +1530,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif; @@ -1549,7 +1549,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream); return 0; } @@ -1559,7 +1559,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif_stream; @@ -1621,7 +1621,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp); return 0; } @@ -1631,7 +1631,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3); @@ -1668,7 +1668,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp); return 0; } @@ -1678,7 +1678,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1712,7 +1712,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp); return 0; } @@ -1722,7 +1722,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1756,7 +1756,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp); return 0; } @@ -1766,7 +1766,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1800,7 +1800,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp); return 0; } @@ -1810,7 +1810,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1847,7 +1847,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_spdif_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1877,7 +1877,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct ucontrol->value.enumerated.item[0] = 9; break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1901,7 +1901,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate; return 0; } @@ -1918,7 +1918,7 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; @@ -1931,7 +1931,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_external_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1959,9 +1959,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru break; case 192000: ucontrol->value.enumerated.item[0] = 9; - break; + break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1987,7 +1987,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp) static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Master", "Slave" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 2; @@ -2000,7 +2000,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp); return 0; } @@ -2037,7 +2037,7 @@ static int hdsp_clock_source(struct hdsp *hdsp) case 192000: return 9; default: - return 3; + return 3; } } else { return 0; @@ -2051,7 +2051,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) case HDSP_CLOCK_SOURCE_AUTOSYNC: if (hdsp_external_sample_rate(hdsp) != 0) { if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) { - hdsp->control_register &= ~HDSP_ClockModeMaster; + hdsp->control_register &= ~HDSP_ClockModeMaster; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); return 0; } @@ -2062,7 +2062,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) break; case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ: rate = 44100; - break; + break; case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ: rate = 48000; break; @@ -2097,13 +2097,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ { static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; if (hdsp->io_type == H9632) uinfo->value.enumerated.items = 10; else - uinfo->value.enumerated.items = 7; + uinfo->value.enumerated.items = 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2113,7 +2113,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp); return 0; } @@ -2123,7 +2123,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2149,7 +2149,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; return 0; } @@ -2184,7 +2184,7 @@ static int hdsp_da_gain(struct hdsp *hdsp) case HDSP_DAGainMinus10dBV: return 2; default: - return 1; + return 1; } } @@ -2199,8 +2199,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_DAGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_DAGainMinus10dBV; - break; + hdsp->control_register |= HDSP_DAGainMinus10dBV; + break; default: return -1; @@ -2212,7 +2212,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2225,7 +2225,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp); return 0; } @@ -2235,7 +2235,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2269,7 +2269,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp) case HDSP_ADGainLowGain: return 2; default: - return 1; + return 1; } } @@ -2281,11 +2281,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_ADGainMinus10dBV; break; case 1: - hdsp->control_register |= HDSP_ADGainPlus4dBu; + hdsp->control_register |= HDSP_ADGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_ADGainLowGain; - break; + hdsp->control_register |= HDSP_ADGainLowGain; + break; default: return -1; @@ -2297,7 +2297,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2310,7 +2310,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp); return 0; } @@ -2320,7 +2320,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2354,7 +2354,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp) case HDSP_PhoneGainMinus12dB: return 2; default: - return 0; + return 0; } } @@ -2366,11 +2366,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_PhoneGain0dB; break; case 1: - hdsp->control_register |= HDSP_PhoneGainMinus6dB; + hdsp->control_register |= HDSP_PhoneGainMinus6dB; break; case 2: - hdsp->control_register |= HDSP_PhoneGainMinus12dB; - break; + hdsp->control_register |= HDSP_PhoneGainMinus12dB; + break; default: return -1; @@ -2382,7 +2382,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2395,7 +2395,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp); return 0; } @@ -2405,7 +2405,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2451,7 +2451,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode) static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp); return 0; } @@ -2461,7 +2461,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2507,7 +2507,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode) static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp); return 0; } @@ -2517,7 +2517,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2595,7 +2595,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd { static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2614,7 +2614,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd uinfo->value.enumerated.items = 0; break; } - + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2624,7 +2624,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp); return 0; } @@ -2634,7 +2634,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; @@ -2683,7 +2683,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) case HDSP_SelSyncRef_SPDIF: return HDSP_AUTOSYNC_FROM_SPDIF; case HDSP_SelSyncRefMask: - return HDSP_AUTOSYNC_FROM_NONE; + return HDSP_AUTOSYNC_FROM_NONE; case HDSP_SelSyncRef_ADAT1: return HDSP_AUTOSYNC_FROM_ADAT1; case HDSP_SelSyncRef_ADAT2: @@ -2699,7 +2699,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 7; @@ -2712,7 +2712,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp); return 0; } @@ -2746,7 +2746,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp_line_out(hdsp); spin_unlock_irq(&hdsp->lock); @@ -2758,7 +2758,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2792,7 +2792,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise) static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->precise_ptr; spin_unlock_irq(&hdsp->lock); @@ -2804,7 +2804,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2838,7 +2838,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet) static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; spin_unlock_irq(&hdsp->lock); @@ -2850,7 +2850,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2892,12 +2892,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem source = ucontrol->value.integer.value[0]; destination = ucontrol->value.integer.value[1]; - + if (source >= hdsp->max_channels) addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination); else addr = hdsp_input_to_output_key(hdsp,source, destination); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr); spin_unlock_irq(&hdsp->lock); @@ -2945,7 +2945,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; + static char *texts[] = {"No Lock", "Lock", "Sync" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2990,7 +2990,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp) int status = hdsp_read(hdsp, HDSP_statusRegister); if (status & HDSP_SPDIFErrorFlag) return 0; - else { + else { if (status & HDSP_SPDIFSync) return 2; else @@ -3026,7 +3026,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp) return 1; } else return 0; -} +} static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3044,17 +3044,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc } static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx) -{ +{ int status = hdsp_read(hdsp, HDSP_statusRegister); - + if (status & (HDSP_Lock0>>idx)) { if (status & (HDSP_Sync0>>idx)) return 2; else - return 1; + return 1; } else return 0; -} +} static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3072,7 +3072,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn break; case Multiface: case H9632: - if (offset >= 1) + if (offset >= 1) return -EINVAL; break; default: @@ -3134,7 +3134,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); return 0; } @@ -3144,7 +3144,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -3189,7 +3189,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = { .get = snd_hdsp_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -3207,7 +3207,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0), HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0), HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), -/* 'Sample Clock Source' complies with the alsa control naming scheme */ +/* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3259,7 +3259,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) return err; } } - + /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */ if (hdsp->io_type == H9632) { for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) { @@ -3278,7 +3278,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3317,7 +3317,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } } } - + status = hdsp_read(hdsp, HDSP_statusRegister); status2 = hdsp_read(hdsp, HDSP_status2Register); @@ -3381,17 +3381,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ: clock_source = "Internal 192 kHz"; - break; + break; default: - clock_source = "Error"; + clock_source = "Error"; } snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source); - + if (hdsp_system_clock_mode(hdsp)) system_clock_mode = "Slave"; else system_clock_mode = "Master"; - + switch (hdsp_pref_sync_ref (hdsp)) { case HDSP_SYNC_FROM_WORD: pref_sync_ref = "Word Clock"; @@ -3416,7 +3416,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref); - + switch (hdsp_autosync_ref (hdsp)) { case HDSP_AUTOSYNC_FROM_WORD: autosync_ref = "Word Clock"; @@ -3429,7 +3429,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_AUTOSYNC_FROM_NONE: autosync_ref = "None"; - break; + break; case HDSP_AUTOSYNC_FROM_ADAT1: autosync_ref = "ADAT1"; break; @@ -3444,14 +3444,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref); - + snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp)); - + snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); - + snd_iprintf(buffer, "\n"); switch (hdsp_spdif_in(hdsp)) { @@ -3471,7 +3471,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "IEC958 input: ???\n"); break; } - + if (hdsp->control_register & HDSP_SPDIFOpticalOut) snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); else @@ -3529,13 +3529,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf (buffer, "SPDIF: No Lock\n"); else snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock"); - + x = status2 & HDSP_wc_sync; if (status2 & HDSP_wc_lock) snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock"); else snd_iprintf (buffer, "Word Clock: No Lock\n"); - + x = status & HDSP_TimecodeSync; if (status & HDSP_TimecodeLock) snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock"); @@ -3543,11 +3543,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "ADAT Sync: No Lock\n"); snd_iprintf(buffer, "\n"); - + /* Informations about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; - + switch (hdsp_ad_gain(hdsp)) { case 0: tmp = "-10 dBV"; @@ -3573,7 +3573,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf(buffer, "DA Gain : %s\n", tmp); - + switch (hdsp_phone_gain(hdsp)) { case 0: tmp = "0 dB"; @@ -3587,8 +3587,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } snd_iprintf(buffer, "Phones Gain : %s\n", tmp); - snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); - + snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); + if (hdsp->control_register & HDSP_AnalogExtensionBoard) snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n"); else @@ -3651,18 +3651,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax Master clock mode maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - Enable line out. + Enable line out. */ - hdsp->control_register = HDSP_ClockModeMaster | - HDSP_SPDIFInputCoaxial | - hdsp_encode_latency(7) | + hdsp->control_register = HDSP_ClockModeMaster | + HDSP_SPDIFInputCoaxial | + hdsp_encode_latency(7) | HDSP_LineOut; - + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3680,7 +3680,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) hdsp_compute_period_size(hdsp); /* silence everything */ - + for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i) hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN; @@ -3688,7 +3688,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN)) return -EIO; } - + /* H9632 specific defaults */ if (hdsp->io_type == H9632) { hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB); @@ -3706,12 +3706,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) static void hdsp_midi_tasklet(unsigned long arg) { struct hdsp *hdsp = (struct hdsp *)arg; - + if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); if (hdsp->midi[1].pending) snd_hdsp_midi_input_read (&hdsp->midi[1]); -} +} static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) { @@ -3723,7 +3723,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) unsigned int midi0status; unsigned int midi1status; int schedule = 0; - + status = hdsp_read(hdsp, HDSP_statusRegister); audio = status & HDSP_audioIRQPending; @@ -3737,15 +3737,15 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; - + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - + if (hdsp->playback_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); } - + if (midi0 && midi0status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ @@ -3788,10 +3788,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp, if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels)) return NULL; - + if ((mapped_channel = hdsp->channel_map[channel]) < 0) return NULL; - + if (stream == SNDRV_PCM_STREAM_CAPTURE) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else @@ -3984,7 +3984,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; int running; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; @@ -4078,10 +4078,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4107,10 +4107,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4189,7 +4189,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_in_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_in_channels, @@ -4220,7 +4220,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_out_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_out_channels, @@ -4337,8 +4337,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; - } - + } + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -4555,7 +4555,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_playbackRmsLevel + i * 8)) return -EFAULT; - if (copy_u64_le(&peak_rms->input_rms[i], + if (copy_u64_le(&peak_rms->input_rms[i], hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_inputRmsLevel + i * 8)) return -EFAULT; @@ -4565,7 +4565,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; int err; @@ -4599,7 +4599,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_config_info info; unsigned long flags; int i; - + err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; @@ -4633,7 +4633,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp); info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); - + } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4644,7 +4644,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne } case SNDRV_HDSP_IOCTL_GET_9632_AEB: { struct hdsp_9632_aeb h9632_aeb; - + if (hdsp->io_type != H9632) return -EINVAL; h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS; h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS; @@ -4655,7 +4655,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne case SNDRV_HDSP_IOCTL_GET_VERSION: { struct hdsp_version hdsp_version; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; if (hdsp->io_type == Undefined) { if ((err = hdsp_get_iobox_version(hdsp)) < 0) @@ -4671,7 +4671,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_firmware __user *firmware; u32 __user *firmware_data; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */ if (hdsp->io_type == Undefined) return -EINVAL; @@ -4684,25 +4684,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (get_user(firmware_data, &firmware->firmware_data)) return -EFAULT; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0) return -EFAULT; - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; - - snd_hdsp_initialize_channels(hdsp); + + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - + if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n"); return err; @@ -4749,10 +4749,10 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) { struct snd_hwdep *hw; int err; - + if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0) return err; - + hdsp->hwdep = hw; hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); @@ -4760,7 +4760,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) hw->ops.open = snd_hdsp_hwdep_dummy_op; hw->ops.ioctl = snd_hdsp_hwdep_ioctl; hw->ops.release = snd_hdsp_hwdep_dummy_op; - + return 0; } @@ -4793,24 +4793,24 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp) static int snd_hdsp_enable_io (struct hdsp *hdsp) { int i; - + if (hdsp_fifo_wait (hdsp, 0, 100)) { snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n"); return -EIO; } - + for (i = 0; i < hdsp->max_channels; ++i) { hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1); hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1); } - + return 0; } static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { int status, aebi_channels, aebo_channels; - + switch (hdsp->io_type) { case Digiface: hdsp->card_name = "RME Hammerfall DSP + Digiface"; @@ -4823,7 +4823,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS; break; - + case H9632: status = hdsp_read(hdsp, HDSP_statusRegister); /* HDSP_AEBx bits are low when AEB are connected */ @@ -4843,7 +4843,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; - + default: /* should never get here */ break; @@ -4859,12 +4859,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp) static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp) { int err; - + if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n"); return err; } - + if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n"); @@ -4895,19 +4895,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n"); return err; } - + if (!(hdsp->state & HDSP_InitializationComplete)) { strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - + if ((err = snd_card_register(card)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n"); return err; } hdsp->state |= HDSP_InitializationComplete; } - + return 0; } @@ -4918,7 +4918,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) const char *fwfile; const struct firmware *fw; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp->io_type == Undefined) { @@ -4927,7 +4927,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; } - + /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { case Multiface: @@ -4961,12 +4961,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache)); release_firmware(fw); - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; @@ -5013,14 +5013,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card, hdsp->max_channels = 26; hdsp->card = card; - + spin_lock_init(&hdsp->lock); tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); - + pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; - + /* From Martin Bjoernsen : "It is important that the card's latency timer register in the PCI configuration space is set to a value much larger @@ -5029,7 +5029,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, to its maximum 255 to avoid problems with some computers." */ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF); - + strcpy(card->driver, "H-DSP"); strcpy(card->mixername, "Xilinx FPGA"); @@ -5043,7 +5043,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, } else { hdsp->card_name = "RME HDSP 9632"; hdsp->max_channels = 16; - is_9632 = 1; + is_9632 = 1; } if ((err = pci_enable_device(pci)) < 0) @@ -5072,7 +5072,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) return err; - + if (!is_9652 && !is_9632) { /* we wait a maximum of 10 seconds to let freshly inserted cardbus cards do their hardware init */ err = hdsp_wait_for_iobox(hdsp, 1000, 10); @@ -5098,35 +5098,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; return 0; } else { - snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; - else + else hdsp->io_type = Digiface; } } - + if ((err = snd_hdsp_enable_io(hdsp)) != 0) return err; - + if (is_9652) hdsp->io_type = H9652; - + if (is_9632) hdsp->io_type = H9632; if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; - + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - hdsp->state |= HDSP_FirmwareLoaded; + hdsp->state |= HDSP_FirmwareLoaded; if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0) return err; - return 0; + return 0; } static int snd_hdsp_free(struct hdsp *hdsp) @@ -5142,13 +5142,13 @@ static int snd_hdsp_free(struct hdsp *hdsp) free_irq(hdsp->irq, (void *)hdsp); snd_hdsp_free_buffers(hdsp); - + if (hdsp->iobase) iounmap(hdsp->iobase); if (hdsp->port) pci_release_regions(hdsp->pci); - + pci_disable_device(hdsp->pci); return 0; } @@ -5191,7 +5191,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, } strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); if ((err = snd_card_register(card)) < 0) { -- 1.6.1.2

1 0

Re: [alsa-devel] [PATCH 0/4] Intel HDMI audio fixes
by Takashi Iwai 20 Feb '09

20 Feb '09

At Wed, 11 Feb 2009 14:32:41 -0800, Shane W wrote: > > On Wed, Feb 11, 2009 at 03:22:27PM +0800, Wu Fengguang wrote: > > Shane: the first two patches should address the two bugs you reported, i.e. > > - multi-channel HDMI audio playback problem > > - first half-second audio samples missing > > These two bugs was reproduced and fixed in the combination of > > G35 SDVO HDMI + Yamaha RX-V1800 AV receiver. The first problem cannot be > > verified in G45 system because we still cannot make G45 work with > > Yamaha. The second problem is interesting in that it can be fixed by the > > second patch in G35, but still remains in G45. > > Hi, > > Ok mixed results here. I'm using this for my test signal: > http://www.csy.ca/~shane/51test.wav > > when using: > aplay -D hw:0,3 51test.wav > > Front and rear left and right are correct but centre is > missing. The order in the wav is: > front left > centre > front right > rear left > rear right > sub buz The order assumed here doesn't match with ALSA mapping. In ALSA, front left front right rear left rear right center LFE Better to test with speaker-test program whether the mapping is correct. Takashi

2 5

[PATCH] Preliminary soc-scenario API.
by Robert Jarzmik 20 Feb '09

20 Feb '09

Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- include/sound/soc-scenario.h | 119 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 119 insertions(+), 0 deletions(-) create mode 100644 include/sound/soc-scenario.h diff --git a/include/sound/soc-scenario.h b/include/sound/soc-scenario.h new file mode 100644 index 0000000..bf02d1e --- /dev/null +++ b/include/sound/soc-scenario.h @@ -0,0 +1,119 @@ +/* + * Handles the machine boards scenarios + * + * Copyright (C) 2008 Robert Jarzmik + * + * 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 + * + * + * Some boards need to enforce use cases of audio paths to protect the hardware + * from overheating if uncorrectly used. + * + * This infrastructure is _not_ to be used as a scenario API. This should be + * left to userland applications. This API unique purpose is to protect devices + * from hardware destruction through pre-defined use cases. + * + * ALWAYS CONSIDER A USERLAND SOLUTION before using soc-scenario !!! + */ + +/* + * Usage example: + * static struct scen_gsm = { + * .pin_activate = { "Front Speaker", "Front Mic", "GSM Line Out", + * "GSM Line In", NULL }, + * .pin_deactivate = { "Rear Speaker", NULL }, + * .mixes_num = 4, + * .mixes = { + * SOC_SCN_MIX("Left HP Mixer PC Beep Playback Switch", 1), + * SOC_SCN_MIX("Left Headphone Out Mux", 2), + * SOC_SCN_MIX("Right HP Mixer MonoIn Playback Switch" , 1), + * SOC_SNC_MIX("Right Headphone Out Mux", 2), + * }, + * }; + */ + +#define SOC_SNC_MIX(_name, _val) { .mixname = _name, .val = _val } +struct soc_scen_setup_mixmux { + char *mixname; /* Codec Mixer or Mux name */ + snd_ctl_elem_value val; /* Value to enforce */ +}; + +/** + * struct soc_scenario - describes one sound usecase + * @name: Scenario name, value as will be seen in alsa "SoC Mode" alsa control + * @pin_activate: Pins to activate on scenario activation + * @pin_deactivate: Pins to deactivate on scenario activation + * Table of all pins, as they should be configured. Each elements is a + * pin_change value, describing how to handle a specific pin. + * @mixes_num: Size of array mixes[] + * @mixes: Mixers/muxes to set up in phase (b) + * Table of all mixers and muxes to set up upon entering this scenario + * @lvol_master: Left volume aliased to "SoC Volume" + * @rvol_master: Right volume aliased to "SoC Volume" + * @lvol_mute: Left volume mute aliased to "SoC Volume" mute control + * @rvol_mute: Right volume mute aliased to "SoC Volume" mute control + * + * This structure describes what a scenario change implies. The behaviour is to : + * a) enable several pins, disable others, leave others in their state + * (understand here snd_soc_dapm_enable_pin) + * Note: tables are NULL terminated + * b) set up some mixers and muxes to the final state This should be normally + * used to activate the mixers/muxes into their final state. + * Note: table is NULL, or always terminated by NULL pointer + */ +struct soc_scenario { + const char *name; /* scenario name */ + const char *pin_activate[]; /* pins to activate */ + const char *pin_deactivate[]; /* pins to deactivate */ + const int mixes_num; /* size of mixes[] */ + const struct soc_scen_setup_mixmux mixes[]; /* mixers/muxes setup */ + const char *lvol_master; /* left volume master */ + const char *rvol_master; /* right volume master */ + const char *lvol_mute; /* left volume mute */ + const char *rvol_mute; /* right volume mute */ +}; + +/** + * snd_soc_scenario_init - initialize soc scenarios + * @card: card associated to the pins/mixers/muxes/volumes/mutes + * @scenario: table of scenarios + * @num_scenarios: number of scenarios + * + * Remember, only use if hardware damage is to be prevented. See file header. + * + * Initialise the soc scenarios engine. The first scenario (0) will be used. By + * default, the user could leave this scenario as non intrusive (not a single + * pin changed, no mixers/muxes changes, and volume master inactive). + */ +int snd_soc_scenario_init(struct snd_soc_card *card, + struct soc_scenario *scenario, int num_scenarios); + +/** + * snd_soc_scenario_release - release soc scenarios + * @card: card used for the init + */ +void snd_soc_scenario_release(struct snd_soc_card *card); + +/** + * snd_soc_scenario_activate - activate a scenario + * @card: card associated to the pins/mixers/muxes/volumes/mutes + * @name: scenario name + * + * Activates a scenario. This is the same behaviour as through alsa control, if + * "SoC Mode" control was assigned a "name" value. + * + * Returns -EINVAL if name not known, or 0 upon success. + */ +int snd_soc_scenario_activate(struct snd_soc_card *card, char *name); -- 1.5.6.5

1 0

[PATCH] Preliminary soc-scenario API.
by Robert Jarzmik 20 Feb '09

20 Feb '09

Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- include/sound/soc-scenario.h | 119 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 119 insertions(+), 0 deletions(-) create mode 100644 include/sound/soc-scenario.h diff --git a/include/sound/soc-scenario.h b/include/sound/soc-scenario.h new file mode 100644 index 0000000..bf02d1e --- /dev/null +++ b/include/sound/soc-scenario.h @@ -0,0 +1,119 @@ +/* + * Handles the machine boards scenarios + * + * Copyright (C) 2008 Robert Jarzmik + * + * 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 + * + * + * Some boards need to enforce use cases of audio paths to protect the hardware + * from overheating if uncorrectly used. + * + * This infrastructure is _not_ to be used as a scenario API. This should be + * left to userland applications. This API unique purpose is to protect devices + * from hardware destruction through pre-defined use cases. + * + * ALWAYS CONSIDER A USERLAND SOLUTION before using soc-scenario !!! + */ + +/* + * Usage example: + * static struct scen_gsm = { + * .pin_activate = { "Front Speaker", "Front Mic", "GSM Line Out", + * "GSM Line In", NULL }, + * .pin_deactivate = { "Rear Speaker", NULL }, + * .mixes_num = 4, + * .mixes = { + * SOC_SCN_MIX("Left HP Mixer PC Beep Playback Switch", 1), + * SOC_SCN_MIX("Left Headphone Out Mux", 2), + * SOC_SCN_MIX("Right HP Mixer MonoIn Playback Switch" , 1), + * SOC_SNC_MIX("Right Headphone Out Mux", 2), + * }, + * }; + */ + +#define SOC_SNC_MIX(_name, _val) { .mixname = _name, .val = _val } +struct soc_scen_setup_mixmux { + char *mixname; /* Codec Mixer or Mux name */ + snd_ctl_elem_value val; /* Value to enforce */ +}; + +/** + * struct soc_scenario - describes one sound usecase + * @name: Scenario name, value as will be seen in alsa "SoC Mode" alsa control + * @pin_activate: Pins to activate on scenario activation + * @pin_deactivate: Pins to deactivate on scenario activation + * Table of all pins, as they should be configured. Each elements is a + * pin_change value, describing how to handle a specific pin. + * @mixes_num: Size of array mixes[] + * @mixes: Mixers/muxes to set up in phase (b) + * Table of all mixers and muxes to set up upon entering this scenario + * @lvol_master: Left volume aliased to "SoC Volume" + * @rvol_master: Right volume aliased to "SoC Volume" + * @lvol_mute: Left volume mute aliased to "SoC Volume" mute control + * @rvol_mute: Right volume mute aliased to "SoC Volume" mute control + * + * This structure describes what a scenario change implies. The behaviour is to : + * a) enable several pins, disable others, leave others in their state + * (understand here snd_soc_dapm_enable_pin) + * Note: tables are NULL terminated + * b) set up some mixers and muxes to the final state This should be normally + * used to activate the mixers/muxes into their final state. + * Note: table is NULL, or always terminated by NULL pointer + */ +struct soc_scenario { + const char *name; /* scenario name */ + const char *pin_activate[]; /* pins to activate */ + const char *pin_deactivate[]; /* pins to deactivate */ + const int mixes_num; /* size of mixes[] */ + const struct soc_scen_setup_mixmux mixes[]; /* mixers/muxes setup */ + const char *lvol_master; /* left volume master */ + const char *rvol_master; /* right volume master */ + const char *lvol_mute; /* left volume mute */ + const char *rvol_mute; /* right volume mute */ +}; + +/** + * snd_soc_scenario_init - initialize soc scenarios + * @card: card associated to the pins/mixers/muxes/volumes/mutes + * @scenario: table of scenarios + * @num_scenarios: number of scenarios + * + * Remember, only use if hardware damage is to be prevented. See file header. + * + * Initialise the soc scenarios engine. The first scenario (0) will be used. By + * default, the user could leave this scenario as non intrusive (not a single + * pin changed, no mixers/muxes changes, and volume master inactive). + */ +int snd_soc_scenario_init(struct snd_soc_card *card, + struct soc_scenario *scenario, int num_scenarios); + +/** + * snd_soc_scenario_release - release soc scenarios + * @card: card used for the init + */ +void snd_soc_scenario_release(struct snd_soc_card *card); + +/** + * snd_soc_scenario_activate - activate a scenario + * @card: card associated to the pins/mixers/muxes/volumes/mutes + * @name: scenario name + * + * Activates a scenario. This is the same behaviour as through alsa control, if + * "SoC Mode" control was assigned a "name" value. + * + * Returns -EINVAL if name not known, or 0 upon success. + */ +int snd_soc_scenario_activate(struct snd_soc_card *card, char *name); -- 1.5.6.5

1 0

[alsa-devel] Hammerfall hdsploader not working in ubuntu 8.10
by Robert Gründler 20 Feb '09

20 Feb '09

Hi, i'm using a rme Hammerfall hdsp multiface card which i've never had problems so far, due to the fact that it's pretty well supported. However, after upgrading to Ubuntu 8.10 the hdsploader fails to upload the firmware to the card with the following error: hdsploader - firmware loader for RME Hammerfall DSP cards Looking for HDSP + Multiface or Digiface cards : Card 0 : RME Hammerfall DSP + Multiface at 0xf87f0000, irq 22 Upload firmware for card hw:0 Hwdep ioctl error on card hw:0 : Device or resource busy. The hdspmixer shows up without problems but there's no sound in my system, which i guess is caused by the hdsploader problem. Here's the system info generated by the alsa information script i got in the irc channel: http://www.alsa-project.org/db/?f=79389b0e9bc3cce3a4d393e02db6ca476f85c4e7 The only difference i noticed between ubuntu 8.04 and 8.10 concerning the firmware was that the alsa-firmware package isn't included in the standard repos anymore. I've installed the package through the Medibuntu repository. Anyone knows what could cause this behaviour ? thanks ! -robert

3 5

[alsa-devel] zero appl_ptr reported
by Wu Fengguang 20 Feb '09

20 Feb '09

Hi Takashi, I noticed that 'appl_ptr' is always 0 during playback, whether it be mplayer or aplay, T61 or DG45ID. Is this a bug? % grep -r . /proc/asound/card0/pcm0p /proc/asound/card0/pcm0p/sub0/prealloc_max:32768 /proc/asound/card0/pcm0p/sub0/prealloc:64 /proc/asound/card0/pcm0p/sub0/status:state: RUNNING /proc/asound/card0/pcm0p/sub0/status:trigger_time: 260.550520867 /proc/asound/card0/pcm0p/sub0/status:tstamp : 264.361429403 /proc/asound/card0/pcm0p/sub0/status:delay : -182952 /proc/asound/card0/pcm0p/sub0/status:avail : 191144 /proc/asound/card0/pcm0p/sub0/status:avail_max : 191144 /proc/asound/card0/pcm0p/sub0/status:----- /proc/asound/card0/pcm0p/sub0/status:hw_ptr : 182952 /proc/asound/card0/pcm0p/sub0/status:appl_ptr : 0 /proc/asound/card0/pcm0p/sub0/sw_params:tstamp_mode: ENABLE /proc/asound/card0/pcm0p/sub0/sw_params:period_step: 1 /proc/asound/card0/pcm0p/sub0/sw_params:avail_min: 1024 /proc/asound/card0/pcm0p/sub0/sw_params:start_threshold: 1 /proc/asound/card0/pcm0p/sub0/sw_params:stop_threshold: 4611686018427387904 /proc/asound/card0/pcm0p/sub0/sw_params:silence_threshold: 0 /proc/asound/card0/pcm0p/sub0/sw_params:silence_size: 4611686018427387904 /proc/asound/card0/pcm0p/sub0/sw_params:boundary: 4611686018427387904 /proc/asound/card0/pcm0p/sub0/hw_params:access: MMAP_INTERLEAVED /proc/asound/card0/pcm0p/sub0/hw_params:format: S32_LE /proc/asound/card0/pcm0p/sub0/hw_params:subformat: STD /proc/asound/card0/pcm0p/sub0/hw_params:channels: 2 /proc/asound/card0/pcm0p/sub0/hw_params:rate: 48000 (48000/1) /proc/asound/card0/pcm0p/sub0/hw_params:period_size: 1024 /proc/asound/card0/pcm0p/sub0/hw_params:buffer_size: 8192 /proc/asound/card0/pcm0p/sub0/info:card: 0 /proc/asound/card0/pcm0p/sub0/info:device: 0 /proc/asound/card0/pcm0p/sub0/info:subdevice: 0 /proc/asound/card0/pcm0p/sub0/info:stream: PLAYBACK /proc/asound/card0/pcm0p/sub0/info:id: STAC92xx Analog /proc/asound/card0/pcm0p/sub0/info:name: STAC92xx Analog /proc/asound/card0/pcm0p/sub0/info:subname: subdevice #0 /proc/asound/card0/pcm0p/sub0/info:class: 0 /proc/asound/card0/pcm0p/sub0/info:subclass: 0 /proc/asound/card0/pcm0p/sub0/info:subdevices_count: 1 /proc/asound/card0/pcm0p/sub0/info:subdevices_avail: 0 /proc/asound/card0/pcm0p/xrun_debug:0 /proc/asound/card0/pcm0p/info:card: 0 /proc/asound/card0/pcm0p/info:device: 0 /proc/asound/card0/pcm0p/info:subdevice: 0 /proc/asound/card0/pcm0p/info:stream: PLAYBACK /proc/asound/card0/pcm0p/info:id: STAC92xx Analog /proc/asound/card0/pcm0p/info:name: STAC92xx Analog /proc/asound/card0/pcm0p/info:subname: subdevice #0 /proc/asound/card0/pcm0p/info:class: 0 /proc/asound/card0/pcm0p/info:subclass: 0 /proc/asound/card0/pcm0p/info:subdevices_count: 1 /proc/asound/card0/pcm0p/info:subdevices_avail: 0 Interestingly DG45ID HDMI audio playback does not have this problem: /proc/asound/card0/pcm3p/sub0/prealloc_max:32768 /proc/asound/card0/pcm3p/sub0/prealloc:64 /proc/asound/card0/pcm3p/sub0/status:state: RUNNING /proc/asound/card0/pcm3p/sub0/status:trigger_time: 441.375970806 /proc/asound/card0/pcm3p/sub0/status:tstamp : 447.039903859 /proc/asound/card0/pcm3p/sub0/status:delay : 16096 /proc/asound/card0/pcm3p/sub0/status:avail : 288 /proc/asound/card0/pcm3p/sub0/status:avail_max : 288 /proc/asound/card0/pcm3p/sub0/status:----- /proc/asound/card0/pcm3p/sub0/status:hw_ptr : 249824 /proc/asound/card0/pcm3p/sub0/status:appl_ptr : 265920 /proc/asound/card0/pcm3p/sub0/sw_params:tstamp_mode: NONE /proc/asound/card0/pcm3p/sub0/sw_params:period_step: 1 /proc/asound/card0/pcm3p/sub0/sw_params:avail_min: 1024 /proc/asound/card0/pcm3p/sub0/sw_params:start_threshold: 1024 /proc/asound/card0/pcm3p/sub0/sw_params:stop_threshold: 4611686018427387904 /proc/asound/card0/pcm3p/sub0/sw_params:silence_threshold: 0 /proc/asound/card0/pcm3p/sub0/sw_params:silence_size: 4611686018427387904 /proc/asound/card0/pcm3p/sub0/sw_params:boundary: 4611686018427387904 /proc/asound/card0/pcm3p/sub0/hw_params:access: RW_INTERLEAVED /proc/asound/card0/pcm3p/sub0/hw_params:format: S16_LE /proc/asound/card0/pcm3p/sub0/hw_params:subformat: STD /proc/asound/card0/pcm3p/sub0/hw_params:channels: 2 /proc/asound/card0/pcm3p/sub0/hw_params:rate: 44100 (44100/1) /proc/asound/card0/pcm3p/sub0/hw_params:period_size: 1024 /proc/asound/card0/pcm3p/sub0/hw_params:buffer_size: 16384 /proc/asound/card0/pcm3p/sub0/info:card: 0 /proc/asound/card0/pcm3p/sub0/info:device: 3 /proc/asound/card0/pcm3p/sub0/info:subdevice: 0 /proc/asound/card0/pcm3p/sub0/info:stream: PLAYBACK /proc/asound/card0/pcm3p/sub0/info:id: INTEL HDMI /proc/asound/card0/pcm3p/sub0/info:name: INTEL HDMI /proc/asound/card0/pcm3p/sub0/info:subname: subdevice #0 /proc/asound/card0/pcm3p/sub0/info:class: 0 /proc/asound/card0/pcm3p/sub0/info:subclass: 0 /proc/asound/card0/pcm3p/sub0/info:subdevices_count: 1 /proc/asound/card0/pcm3p/sub0/info:subdevices_avail: 0 /proc/asound/card0/pcm3p/xrun_debug:0 /proc/asound/card0/pcm3p/info:card: 0 /proc/asound/card0/pcm3p/info:device: 3 /proc/asound/card0/pcm3p/info:subdevice: 0 /proc/asound/card0/pcm3p/info:stream: PLAYBACK /proc/asound/card0/pcm3p/info:id: INTEL HDMI /proc/asound/card0/pcm3p/info:name: INTEL HDMI /proc/asound/card0/pcm3p/info:subname: subdevice #0 /proc/asound/card0/pcm3p/info:class: 0 /proc/asound/card0/pcm3p/info:subclass: 0 /proc/asound/card0/pcm3p/info:subdevices_count: 1 /proc/asound/card0/pcm3p/info:subdevices_avail: 0 Thanks, Fengguang

2 2

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 20 Feb '09

20 Feb '09

The following changes since commit 93b760b7072ca6972c15c798e97af3f830d8bbba: Mark Brown (1): ASoC: Implement SPI device unregistration for WM8731 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (3): ASoC: Check for errors when writing WM8731 reset register ASoC: Add device init/exit annotations to new-style Wolfson CODEC drivers ASoC: Disable WM8731 line bypass by default Peter Ujfalusi (1): ASoC: TWL4030: Add digital loopback support sound/soc/codecs/twl4030.c | 56 +++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8731.c | 19 +++++++++++---- sound/soc/codecs/wm8900.c | 8 +++--- sound/soc/codecs/wm8903.c | 8 +++--- 5 files changed, 73 insertions(+), 20 deletions(-)

2 5

[alsa-devel] HP pavilion dv5 1110ew - no hdmi sound
by Bartłomiej Holdenmayer 20 Feb '09

20 Feb '09

Welcome, I have no sound through HDMI. I have Nvidia Geforce 9200M GS with 180.11 driver. Alsa 1.0.19. And newest bios F13A. I was googling for a week and only solution i found was to upgrade Alsa and Nvidia to the newest version. I did, but i still do not have sound over HDMI. In Win$ it is working ok. http://www.alsa-project.org/db/?f=01be9175263a5021fcfef3a47123afc04d62a863 - Pozdrawiam / Regards bartek GG:541634; ICQ:33920811;

4 35

[PATCH v2] Allow choice in ac97 gpio reset line
by Robert Jarzmik 19 Feb '09

19 Feb '09

As the PXA series allow 2 gpios to reset the ac97 bus, allow through platform data configuration the definition of the correct gpio which will reset the AC97 bus. This comes from a silicon defect on the PXA series, where the gpio must be manually controlled in warm reset cases. Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- include/sound/pxa2xx-lib.h | 11 +++++++ sound/arm/pxa2xx-ac97-lib.c | 71 ++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 77 insertions(+), 5 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 2fd3d25..01c2e4a 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -42,4 +42,15 @@ extern int pxa2xx_ac97_hw_resume(void); extern int pxa2xx_ac97_hw_probe(struct platform_device *dev); extern void pxa2xx_ac97_hw_remove(struct platform_device *dev); +/* AC97 platform_data */ +/** + * struct pxa2xx_ac97_platform_data - pxa ac97 platform data + * @reset_gpio_95: AC97 reset gpio is gpio95 + * @reset_gpio_113: AC97 reset gpio is gpio113 + */ +struct pxa2xx_ac97_platform_data { + unsigned reset_gpio_95:1; + unsigned reset_gpio_113:1; +}; + #endif diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c..acf2e5a 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -31,6 +31,8 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int is_ac97_resetgpio_95; +static int is_ac97_resetgpio_113; /* * Beware PXA27x bugs: @@ -42,6 +44,55 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA CPUs suffer from a AC97 bug, a manual control of the reset + * line must be done to insure proper work of AC97 reset line. + * This function computes the correct gpio_mode for further use by reset + * functions, and applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (is_ac97_resetgpio_113) + switch (resetgpio_action) { + case RESETGPIO_FORCE_LOW: + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = 113 | GPIO_OUT | GPIO_DFLT_HIGH; + break; + case RESETGPIO_NORMAL_ALTFUNC: + mode = 113 | GPIO_ALT_FN_2_OUT; + break; + }; + + if (is_ac97_resetgpio_95) + switch (resetgpio_action) { + case RESETGPIO_FORCE_LOW: + mode = 95 | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = 95 | GPIO_OUT | GPIO_DFLT_HIGH; + break; + case RESETGPIO_NORMAL_ALTFUNC: + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -137,10 +188,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -308,8 +359,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -320,6 +371,16 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; + struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + + if (!pdata) { + is_ac97_resetgpio_113 = 1; + } else { + if (pdata->reset_gpio_95) + is_ac97_resetgpio_95 = 1; + if (pdata->reset_gpio_113) + is_ac97_resetgpio_113 = 1; + } if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); @@ -330,7 +391,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- 1.5.6.5

1 0

[PATCH v2] Allow choice in ac97 gpio reset line
by Robert Jarzmik 19 Feb '09

19 Feb '09

As the PXA27x series allow 2 gpios to reset the ac97 bus, allow through platform data configuration the definition of the correct gpio which will reset the AC97 bus. This comes from a silicon defect on the PXA27x series, where the gpio must be manually controlled in warm reset cases. Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- include/sound/pxa2xx-lib.h | 14 +++++++++ sound/arm/pxa2xx-ac97-lib.c | 68 +++++++++++++++++++++++++++++++++++++++--- 2 files changed, 77 insertions(+), 5 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 2fd3d25..9ee0f19 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -42,4 +42,18 @@ extern int pxa2xx_ac97_hw_resume(void); extern int pxa2xx_ac97_hw_probe(struct platform_device *dev); extern void pxa2xx_ac97_hw_remove(struct platform_device *dev); +/* AC97 platform_data */ +/** + * struct pxa2xx_ac97_platform_data - pxa ac97 platform data + * @reset_gpio: AC97 reset gpio (normally gpio113 or gpio95) + * + * Platform data should only be specified for pxa27x CPUs where a silicon bug + * prevents correct operation of the reset line. If not specified, the default + * behaviour is to consider gpio 113 as the AC97 reset line, which is the + * default on most boards. + */ +struct pxa2xx_ac97_platform_data { + int reset_gpio; +}; + #endif diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c..1df4b0b 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -31,6 +31,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int reset_gpio; /* * Beware PXA27x bugs: @@ -42,6 +43,45 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -137,10 +177,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -308,8 +348,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -320,6 +360,24 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; + struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + + if (pdata) { + switch (pdata->reset_gpio) { + case 95: + case 113: + reset_gpio = pdata->reset_gpio; + break; + case 0: + break; + default: + dev_err(dev, "Invalid reset GPIO %d\n", + pdata->reset_gpio); + } + } else { + if (cpu_is_pxa27x()) + reset_gpio = 113; + } if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); @@ -330,7 +388,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- 1.5.6.5

1 0

[PATCH v3] Allow choice in ac97 gpio reset line
by Robert Jarzmik 19 Feb '09

19 Feb '09

As the PXA27x series allow 2 gpios to reset the ac97 bus, allow through platform data configuration the definition of the correct gpio which will reset the AC97 bus. This comes from a silicon defect on the PXA27x series, where the gpio must be manually controlled in warm reset cases. Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- include/sound/pxa2xx-lib.h | 15 +++++++++ sound/arm/pxa2xx-ac97-lib.c | 71 ++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 81 insertions(+), 5 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 2fd3d25..2c894b6 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -42,4 +42,19 @@ extern int pxa2xx_ac97_hw_resume(void); extern int pxa2xx_ac97_hw_probe(struct platform_device *dev); extern void pxa2xx_ac97_hw_remove(struct platform_device *dev); +/* AC97 platform_data */ +/** + * struct pxa2xx_ac97_platform_data - pxa ac97 platform data + * @reset_gpio: AC97 reset gpio (normally gpio113 or gpio95) + * a -1 value means no gpio will be used for reset + * + * Platform data should only be specified for pxa27x CPUs where a silicon bug + * prevents correct operation of the reset line. If not specified, the default + * behaviour is to consider gpio 113 as the AC97 reset line, which is the + * default on most boards. + */ +struct pxa2xx_ac97_platform_data { + int reset_gpio; +}; + #endif diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c..d721ea7 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -31,6 +31,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int reset_gpio; /* * Beware PXA27x bugs: @@ -42,6 +43,45 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -137,10 +177,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -308,8 +348,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -320,6 +360,27 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; + struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + + if (pdata) { + switch (pdata->reset_gpio) { + case 95: + case 113: + reset_gpio = pdata->reset_gpio; + break; + case 0: + reset_gpio = 113; + break; + case -1: + break; + default: + dev_err(dev, "Invalid reset GPIO %d\n", + pdata->reset_gpio); + } + } else { + if (cpu_is_pxa27x()) + reset_gpio = 113; + } if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); @@ -330,7 +391,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- 1.5.6.5

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[PATCH] Allow choice in ac97 gpio reset line
by Robert Jarzmik 19 Feb '09

19 Feb '09

As the PXA series allow 2 gpios to reset the ac97 bus, allow through configuration the definition of the correct gpio which will reset the AC97 bus. This comes from a silicon defect on the PXA series, where the gpio must be manually controlled in warm reset cases. Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- sound/arm/Kconfig | 17 +++++++++++ sound/arm/pxa2xx-ac97-lib.c | 68 +++++++++++++++++++++++++++++++++++++++--- 2 files changed, 80 insertions(+), 5 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de4..2749274 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -39,6 +39,23 @@ config SND_PXA2XX_LIB config SND_PXA2XX_LIB_AC97 bool +choice + prompt "PXA GPIO line used to reset AC97 bus" + depends on SND_PXA2XX_LIB_AC97 + help + Choose which GPIO is assigned the AC97 reset function. + + Usually board designers use GPIO 113, but GPIO 95 can be used + as well. + +config SND_PXA2XX_LIB_AC97_RESET113 + bool "GPIO113" + +config SND_PXA2XX_LIB_AC97_RESET95 + bool "GPIO95" + +endchoice + config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c..aa522d2 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -31,6 +31,8 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int is_ac97_resetgpio_95; +static int is_ac97_resetgpio_113; /* * Beware PXA27x bugs: @@ -42,6 +44,55 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA CPUs suffer from a AC97 bug, a manual control of the reset + * line must be done to insure proper work of AC97 reset line. + * This function computes the correct gpio_mode for further use by reset + * functions, and applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (is_ac97_resetgpio_113) + switch (resetgpio_action) { + case RESETGPIO_FORCE_LOW: + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = 113 | GPIO_OUT | GPIO_DFLT_HIGH; + break; + case RESETGPIO_NORMAL_ALTFUNC: + mode = 113 | GPIO_ALT_FN_2_OUT; + break; + }; + + if (is_ac97_resetgpio_95) + switch (resetgpio_action) { + case RESETGPIO_FORCE_LOW: + mode = 95 | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = 95 | GPIO_OUT | GPIO_DFLT_HIGH; + break; + case RESETGPIO_NORMAL_ALTFUNC: + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -137,10 +188,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -308,8 +359,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -321,6 +372,13 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; +#ifdef CONFIG_SND_PXA2XX_LIB_AC97_RESET113 + is_ac97_resetgpio_113 = 1; +#endif +#ifdef CONFIG_SND_PXA2XX_LIB_AC97_RESET95 + is_ac97_resetgpio_95 = 1; +#endif + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -330,7 +388,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- 1.5.6.5

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Re: [alsa-devel] Regression: snd_hda_intel driver causes quake3 to hang
by Takashi Iwai 19 Feb '09

19 Feb '09

At Thu, 19 Feb 2009 16:55:39 +0000, Charlie Baylis wrote: > > On Tue, Feb 17, 2009 at 01:05:46PM +0100, Takashi Iwai wrote: > > At Tue, 17 Feb 2009 11:49:17 +0000, Charlie Baylis wrote: > > > > > > > > > Hi > > > > > > Some time ago, quake3 stopped running correctly on my PC. This occured when > > > changing kernel from 2.6.25 to 2.6.27. The symptom is that within a > > > fraction of a second of gameplay, the game freezes. Switching to a console > > > allows the game to be killed. I am using standard kernel.org kernels with > > > no additional patches. > > > > Could you check whether it happens with 2.6.29-rc5? On 2.6.29, the delayed > > handling is done via its own workqueue, so this might be no longer a problem. > > > > Also you could try position_fix=1 instead of bdl_pos_adj. > > > > The problem is likely your hardware does unreliable DMA position reporting. > > Detecting such a broken hardware would be possible, but it'll be a bit > > tricky... > > I tried the latest git (shortly after 2.6.29-rc5) and I found that the same > symptoms occur. This happens with the driver defaults, or with bdl_pos_adj=0, > or with position_fix=1. So 2.6.29 is worse, because there seems to be no > workaround. > > Is there a way to find out whether my hardware really is broken? I had no > problems until the bdl_pos_adj workaround/feature was added. On 2.6.29, bdl_pos_adj=0 disables that. So it'll be a workaround for you, at least. Takashi

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[alsa-devel] [PATCH] Remove documentation for OSS CS4232 driver
by Krzysztof Helt 19 Feb '09

19 Feb '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no OSS cs4232 driver in the kernel any more and this documentation does not contain any info useful for ALSA driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Documentation/sound/oss/CS4232 | 23 ----------------------- 1 files changed, 0 insertions(+), 23 deletions(-) delete mode 100644 Documentation/sound/oss/CS4232 diff --git a/Documentation/sound/oss/CS4232 b/Documentation/sound/oss/CS4232 deleted file mode 100644 index 7d6af7a..0000000 --- a/Documentation/sound/oss/CS4232 +++ /dev/null @@ -1,23 +0,0 @@ -To configure the Crystal CS423x sound chip and activate its DSP functions, -modules may be loaded in this order: - - modprobe sound - insmod ad1848 - insmod uart401 - insmod cs4232 io=* irq=* dma=* dma2=* - -This is the meaning of the parameters: - - io--I/O address of the Windows Sound System (normally 0x534) - irq--IRQ of this device - dma and dma2--DMA channels (DMA2 may be 0) - -On some cards, the board attempts to do non-PnP setup, and fails. If you -have problems, use Linux' PnP facilities. - -To get MIDI facilities add - - insmod opl3 io=* - -where "io" is the I/O address of the OPL3 synthesizer. This will be shown -in /proc/sys/pnp and is normally 0x388. -- 1.5.2.2 ---------------------------------------------------------------------- Promocja w Speak Up. Kurs angielskiego za darmo. Liczba miejsc ograniczona. Sprawd�!>> http://link.interia.pl/f205c

2 1

[alsa-devel] [PATCH] alsa-info.sh: Use pgrep -f to match against command line; fix misspellings
by crimsunkg＠yahoo.com 19 Feb '09

19 Feb '09

Hi, The current alsa-info.sh script fails to report running pulseaudio/esd/artsd instances due to missing -f passed to pgrep. The attached patch resolves said issue and corrects trivial misspellings. Thanks, Dan

2 1

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Digital loopback support
by Peter Ujfalusi 18 Feb '09

18 Feb '09

Hello, The following patch adds the digital loopback support for TWL4030 codec driver. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. With the patch applied, the audio coming from digimic0 can be routed back to the playback path without any CPU interaction. Also one can use this loopback instead of the analog bypass for the analog inputs as well, given that the TX1 capture path is configured in such a way. --- Peter Ujfalusi (1): ASoC: TWL4030: Add digital loopback support sound/soc/codecs/twl4030.c | 57 +++++++++++++++++++++++++++++++++++++++---- 1 files changed, 51 insertions(+), 6 deletions(-)

2 2

[alsa-devel] longname in sound/core/jack.c
by Takashi Iwai 18 Feb '09

18 Feb '09

Mark, I'm not sure whether I've already asked it, but I find it ugly to refer to card->longname for the input device name in sound/core/jack.c. Can it be changed to card->shortname, at least? The long name is often really long like "HDA Intel at 0xf8400000 irq 21", thus the resultant name will be "HDA Intel at 0xf8400000 irq 21 Line Out at Ext Left" as the jack name. Takashi --- diff --git a/sound/core/jack.c b/sound/core/jack.c index 43b10d6..c8254c6 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -55,7 +55,7 @@ static int snd_jack_dev_register(struct snd_device *device) int err; snprintf(jack->name, sizeof(jack->name), "%s %s", - card->longname, jack->id); + card->shortname, jack->id); jack->input_dev->name = jack->name; /* Default to the sound card device. */

2 4

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 18 Feb '09

18 Feb '09

The following changes since commit 59544d33ff3118f22a484d8be06cdf5cfc2fdca5: Mark Brown (1): ASoC: Remove version display from the WM8753 driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (2): ASoC: Include I2C headers for corgi and poodle ASoC: Implement SPI device unregistration for WM8731 sound/soc/codecs/wm8731.c | 7 ++++++- sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/poodle.c | 1 +

3 5

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 18 Feb '09

18 Feb '09

The following changes since commit f1464ede550b8fda75086cb9bacf8d60fc9f6780: Takashi Iwai (1): Merge branch 'for-2.6.30' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Kevin Hilman (1): ASoC: Fix DaVinci module unload error Mark Brown (9): ASoC: Remove version display from the WM8731 driver ASoC: Clean up WM8731 bias level configuration ASoC: Improve diagnostics for AT91SAM9G20-EK probe ASoC: Check machine type before loading on AT91SAM9G20-EK ASoC: Actively manage MCLK for AT91SAM9G20-EK ASoC: Rename AT91SAMG20-EK for applications ASoC: Shuffle WM8731 SPI and I2C device registration ASoC: Refactor WM8731 device registration ASoC: Remove version display from the WM8753 driver Paul Fertser (1): ASoC: Fix WM8753 DAIs unregistering sound/soc/atmel/sam9g20_wm8731.c | 124 +++++++++++-- sound/soc/codecs/wm8731.c | 377 ++++++++++++++++-------------------- sound/soc/codecs/wm8731.h | 6 - sound/soc/codecs/wm8753.c | 9 +- sound/soc/davinci/davinci-evm.c | 3 +- sound/soc/davinci/davinci-sffsdr.c | 3 +- sound/soc/pxa/corgi.c | 43 ++++- sound/soc/pxa/poodle.c | 41 ++++- 8 files changed, 361 insertions(+), 245 deletions(-)

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[alsa-devel] [PATCH] OSS: missing parentheses?
by Roel Kluin 18 Feb '09

18 Feb '09

Please review. --------------------------->8-------------8<------------------------------ Add missing parentheses Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c index 25f3a22..7f377ec 100644 --- a/sound/oss/pas2_card.c +++ b/sound/oss/pas2_card.c @@ -156,9 +156,7 @@ static int __init config_pas_hw(struct address_info *hw_config) * 0x80 */ , 0xB88); - pas_write(0x80 - | joystick?0x40:0 - ,0xF388); + pas_write(0x80 | (joystick ? 0x40 : 0), 0xF388); if (pas_irq < 0 || pas_irq > 15) {

2 1

Re: [alsa-devel] HP pavilion dv5 1110ew - no hdmi sound
by Jean-Pierre André 18 Feb '09

18 Feb '09

Hi all, > Dnia 2009-02-14, sob o godzinie 11:30 +0100, Takashi Iwai pisze: > >> At Fri, 13 Feb 2009 22:48:26 +0100, >> Bartłomiej Holdenmayer wrote: >> >> > > I did ehat You wrote. Here is my dmesg: > [ 14.729023] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) > -> IRQ 2 > 2 > [ 14.729028] hda_intel: codec_mask forced to 0xff > [ 14.729054] HDA Intel 0000:00:1b.0: setting latency timer to 64 > [ 15.776021] > ALSA /usr/src/Alsa-1.0.19/alsa-driver-1.0.19/pci/hda/../../alsa-k > ernel/pci/hda/hda_intel.c:634: hda_intel: azx_get_response timeout, > switching to > polling mode: last cmd=0x100f0000 > [ 16.780019] > ALSA /usr/src/Alsa-1.0.19/alsa-driver-1.0.19/pci/hda/../../alsa-k > ernel/pci/hda/hda_intel.c:1271: hda_intel: Codec #1 probe error; > disabling it... > [ 17.816508] > ALSA /usr/src/Alsa-1.0.19/alsa-driver-1.0.19/pci/hda/../../alsa-k > ernel/pci/hda/hda_intel.c:1271: hda_intel: Codec #3 probe error; > disabling it... > [ 17.921600] input: HDA Digital PCBeep > as /devices/pci0000:00/0000:00:1b.0/inp > ut/input11 > [ 18.052194] input: HDA Intel at 0x9f300000 irq 22 Mic at Ext Front > Jack as /d > evices/pci0000:00/0000:00:1b.0/input/input12 > [ 18.052587] input: HDA Intel at 0x9f300000 irq 22 Mic at Sep UNKNOWN > Jack as > /devices/pci0000:00/0000:00:1b.0/input/input13 > [ 18.052894] input: HDA Intel at 0x9f300000 irq 22 HP Out at Ext Front > Jack as > /devices/pci0000:00/0000:00:1b.0/input/input14 This is very interesting for me, as I have another HP dv4 computer with the same sound chip, and I get roughly similar results, but with a significant difference (using Tagashi's snapshot from yesterday) : HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 HDA Intel 0000:00:1b.0: PCI INT B -> GSI 22 (level, low) -> IRQ 22 hda_intel: codec_mask forced to 0xff HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA /home/linux/rpmbuild/BUILD/alsa-driver-1.0.19.217/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:634: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 ALSA /home/linux/rpmbuild/BUILD/alsa-driver-1.0.19.217/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1271: hda_intel: Codec #1 probe error; disabling it... ALSA /home/linux/rpmbuild/BUILD/alsa-driver-1.0.19.217/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1271: hda_intel: Codec #3 probe error; disabling it... input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input12 input: HDA Intel at 0xdf300000 irq 22 Line In at Ext Rear Jack as /devices/pci0000:00/0000:00:1b.0/input/input13 input: HDA Intel at 0xdf300000 irq 22 Mic at Ext Front Jack as /devices/pci0000:00/0000:00:1b.0/input/input14 input: HDA Intel at 0xdf300000 irq 22 HP Out at Ext Front Jack as /devices/pci0000:00/0000:00:1b.0/input/input15 The point is I never get an IRQ 22, I get IRQ 20 instead, so most players do not work unless I boot with irqpoll or noapic options (mplayer is ok though) Now, you get the IRQ 22 routed by APIC from PCI INT A, whereas I have the IRQ 22 routed by APIC from PCI INT B. And I have the PCI INT A routed to IRQ 20 for USB : ehci_hcd 0000:00:1d.7: PCI INT A -> GSI 20 (level, low) -> IRQ 20 ehci_hcd 0000:00:1d.7: setting latency timer to 64 ehci_hcd 0000:00:1d.7: EHCI Host Controller From this I conclude that both our sound chips trigger INT A, but in my case, it is wrongly the INT B that has been connected to IRQ 22. So my question to all the techies on this list : how on earth can I influence on this IRQ routing ? Regards Jean-Pierre

1 0

[alsa-devel] ALI5451: "stimer is not ready"
by hagedorn＠kimae.de 18 Feb '09

18 Feb '09

Dear All, since our hardware supplier changed the processor type from Transmeta T5700 (667MHz) to T5800 (997MHz), we experience massive problems with the sound card access. With the new T5800 processor we first noticed problems with IDE access. But it can be solved by switching "acpi=off" and choosing another processor type in kernel config, e.g. 'Pentium MMX' instead 'Transmeta Crusoe'. Moreover we experienced random appearance of "ali_stimer_read: stimer is not ready" messages from ALSA when accessing the sound card on T5800. I tried also 2.4.37 and 2.6.22 kernels. An update to ALSA 1.0.19 does not help too. The strange thing is that i can boot the same system over and over again... in some cases i get above error message, sometimes i get no ALSA error message at all but the sound card can not be accessed, and sometimes the system works and is recording sound - at least for a few hours. Loading the kernel module trident instead of ALSA makes the system freeze immediately on a T5800. On a T5700 it works. System yields regulary " i8253 count too high ! resetting " but does not seem to have any effect on system with T5700. On T5800 this messages is much more often. On 2.6.22 kernel i saw "TSC unstable" at boot time. I do not have knowledge about these timing things, so i would like to know a) what does above message mean exactly? b) what timing source is alsa using and how can i influence it? c) what timing source is the kernel using and how can i influence it? d) can this be caused by defect hardware or buggy BIOS? e) do you have any other ideas (e.g. kernel options)? Some info about the original system which worked well on T5700: - Linux is self-compiled with 'Linux from Scratch' - System boots from Compact Flashdisk, HDD is used for log files only - Kernel 2.4.22, optimized for Transmeta crusoe - Glibc version 2.3.2, compiled with GNU CC 3.3.3 for i686 on Kernel 2.4.25 - ALSA 1.0.5a, loaded modules are snd, snd-ali5451, soundcore, snd-pcm snd-timer, snd-page-alloc, snd-ac97-codec The same compact flashcard containing this configuration on a T5800 system will yield IDE errors "DRQ after issuing write" and will show the stated ALSA stimer error. Thank you very much in advance. Kind regards, Christoph

2 1

[alsa-devel] Snapshot From ALSA Server
by Jean-Pierre André 18 Feb '09

18 Feb '09

Hi, When I try to compile alsa-driver from the snapshot from alsa server (eg alsa-driver-1.0.19.11.g5fc41.195.g772ce.tar.bz2), the compilation process fails at applying two patches : pci/hda/hda_beep.patch and acore/misc.patch After manually fixing the patches, the compilation goes on and provides apparently valid modules (at least I get sound from them). Why is it so ? In the said patches I see nothing specific to my environment. Could there be a missing patch or overlapping patches applied in a wrong order ? Are these tarballs compiled before being made available ? Note : this problem does not occur in daily snapshots by Tagashi. Regards Jean-Pierre

3 2

[alsa-devel] [PATCH] hda: Add logic to force HP mixer creation
by Matthew Ranostay 18 Feb '09

18 Feb '09

In some instances like a laptop PCI_QUIRK you may want to force a headphone mixer otherwise a "Front" mixer would be in its place, this is to aid user friendlyness. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 25 ++++++++++++++++++------- 1 files changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aeb5d21..e02a187 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -168,6 +168,7 @@ struct sigmatel_spec { unsigned int surr_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; + unsigned int hp_mixer: 1; unsigned int spdif_mute: 1; unsigned int check_volume_offset:1; @@ -857,9 +858,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, @@ -3115,12 +3116,12 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, "Front", "Surround", NULL /*CLFE*/, "Side" }; hda_nid_t nid = 0; - int i, err; + int i, idx = 0, err; unsigned int wid_caps; for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) { nid = spec->multiout.dac_nids[i]; - if (i == 2) { + if (idx == 2) { /* Center/LFE */ err = create_controls(codec, "Center", nid, 1); if (err < 0) @@ -3139,9 +3140,9 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, if (err < 0) return err; } - + idx++; } else { - const char *name = chname[i]; + const char *name = chname[idx]; /* if it's a single DAC, assign a better name */ if (!i && is_unique_dac(spec, nid)) { switch (cfg->line_out_type) { @@ -3153,6 +3154,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, break; } } + if (!i && spec->hp_mixer) + name = "Headphone"; + else + idx++; err = create_controls(codec, name, nid, 3); if (err < 0) return err; @@ -4894,6 +4899,12 @@ again: break; } + switch (spec->board_config) { + case STAC_DELL_S14: + spec->hp_mixer = 1; + break; + } + err = stac92xx_parse_auto_config(codec, 0x1d, 0); if (!err) { if (spec->board_config < 0) { -- 1.5.6.3

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Re: [alsa-devel] [Fwd: HP DV4 1155 SE - No Sound through, speakers...]
by Guy Stalnaker 18 Feb '09

18 Feb '09

Herton and Takashi, I can confirm that compiling and installing the latest snapshot of alsa-drivers, adding "model=hp-dv5" to /etc/modprobe.d/alsa-base, and running hda-verb with the parameters Takashi indicates DOES result in sound from the onboard, internal speakers on my Hewlitt Packard dv4 1225 Pavilion Entertainment PC. The mute on headphone use does not work, but that's okay by me for now. I think you're on the right track in diagnosing the problem and it's solution. Thank you very, very much for helping us. I know it'll likely be a while before the solution actually makes it into distributions, but at the least I can use my sound apps without having to plug my laptop into something to hear what's playing. Kinda defeats the notion of having a laptop if one has to keep it plugged in. Do either of you mind if I collate notes from the posts exchanged on alsa-devel and post them on the Ubuntu forums? There are a lot of other HP users who have posted about the same problem. It's likely they have no idea the resource available to them in the alsa-devel archives. BEST regards, Guy S. > ------------------------------ > > Message: 4 > Date: Fri, 13 Feb 2009 14:16:21 -0500 > From: Tim Barnette <tim.barnette(a)gmail.com> > Subject: Re: [alsa-devel] [Fwd: HP DV4 1155 SE - No Sound through > speakers...] > To: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org, Jean-Pierre Andr? > <jean-pierre.andre(a)wanadoo.fr> > Message-ID: > <b71cdde20902131116l4c7da119s6c51a67e37371ebe(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Takashi, I was able to build hda-verb and when I executed the command below, > it responded with: > nid=0xd, verb=0x707, param=0x40 > value=0x0 > > On Mon, Feb 2, 2009 at 10:25 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> At Mon, 2 Feb 2009 10:17:03 -0500, >> Tim Barnette wrote: >>> Thanks, Takashi. I made this change and rebooted. >>> >>> I noticed right away the mute light is no longer on... but still no sound >>> through speakers, only headphones.. >>> >>> /sys/module/snd_hda_intel/parameters/model >>> contains >>> hp-dv5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> >>> >>> attached is the alsa-info.txt you requested.. >> Did you try the latest alsa-driver snapshot? >> >> If it still doesn't work, I guess this is a BIOS problem. >> The other HP dv laptops show the widget 0x0d as the speaker while >> yours is unused. >> >> Try to run: >> hda-verb /dev/snd/hwC0D0 0x0d SET_PIN_WID 0x40 >> >> >> Takashi >> > > > ------------------------------ > > Message: 5 > Date: Fri, 13 Feb 2009 14:22:00 -0500 > From: Tim Barnette <tim.barnette(a)gmail.com> > Subject: Re: [alsa-devel] [Fwd: HP DV4 1155 SE - No Sound through > speakers...] > To: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org, Jean-Pierre Andr? > <jean-pierre.andre(a)wanadoo.fr> > Message-ID: > <b71cdde20902131122q16eb7e44u92daf2a4c9500bd0(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > ..And it worked! I now have sound! Can you tell me what I actually did > here? -- Guy Stalnaker, I2@DOIT, jstalnak(a)wisc.edu 1210 West Dayton Street, Room 3209 CSS, Madison WI 53706 work 608.263.8035, cell 608.235.4718, fax 608.265.6681

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[alsa-devel] ALSA SoC Architecture Document
by Anuj Aggarwal 17 Feb '09

17 Feb '09

Hi, I was looking for ASoC architecture document and came across a presentation done by Liam Girdwood for ASoC in Jan 2006 (http://www.embedded-kernel-track.org/2006/ASoC.pdf). But I could not find the presentation because the link has expired. Can someone share the presentation or any other ASoC specific architecture document, which can help someone understand the design and data/control paths? Thanks in advance Anuj Aggarwal

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Re: [alsa-devel] Regression: snd_hda_intel driver causes quake3 to hang
by Takashi Iwai 17 Feb '09

17 Feb '09

At Tue, 17 Feb 2009 11:49:17 +0000, Charlie Baylis wrote: > > > Hi > > Some time ago, quake3 stopped running correctly on my PC. This occured when > changing kernel from 2.6.25 to 2.6.27. The symptom is that within a fraction > of a second of gameplay, the game freezes. Switching to a console allows the > game to be killed. I am using standard kernel.org kernels with no additional > patches. Could you check whether it happens with 2.6.29-rc5? On 2.6.29, the delayed handling is done via its own workqueue, so this might be no longer a problem. Also you could try position_fix=1 instead of bdl_pos_adj. The problem is likely your hardware does unreliable DMA position reporting. Detecting such a broken hardware would be possible, but it'll be a bit tricky... Takashi > I have bisected the problem, and found that commit > 675f25d4d3ff6501cbce608bcc2333a56ec4c105 (ALSA: hda - Add bdl_pos_adj option) > introduced the problem. The bdl_pos_adj option described in the commit log can > be adjusted to workaround the problem. > > If I run the following command, the game runs correctly. > # echo 0 > /sys/module/snd_hda_intel/parameters/bdl_pos_adj > > The PC has a Asus N4L VM DH motherboard. lspci describes the hardware as: > $ lspci > 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS, 943/940GML and 945GT Express Memory Controller Hub (rev 03) > 00:02.0 VGA compatible controller: Intel Corporation Mobile 945GM/GMS, 943/940GML Express Integrated Graphics Controller (rev 03) > 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) > 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 02) > 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev 02) > 00:1c.3 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 4 (rev 02) > 00:1d.0 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #1 (rev 02) > 00:1d.1 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #2 (rev 02) > 00:1d.2 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #3 (rev 02) > 00:1d.3 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #4 (rev 02) > 00:1d.7 USB Controller: Intel Corporation 82801G (ICH7 Family) USB2 EHCI Controller (rev 02) > 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev e2) > 00:1f.0 ISA bridge: Intel Corporation 82801GHM (ICH7-M DH) LPC Interface Bridge (rev 02) > 00:1f.2 IDE interface: Intel Corporation 82801GBM/GHM (ICH7 Family) SATA IDE Controller (rev 02) > 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 02) > 01:00.0 Ethernet controller: Intel Corporation 82573L Gigabit Ethernet Controller > 02:00.0 SATA controller: JMicron Technologies, Inc. JMicron 20360/20363 AHCI Controller (rev 02) > 02:00.1 IDE interface: JMicron Technologies, Inc. JMicron 20360/20363 AHCI Controller (rev 02) > 04:03.0 FireWire (IEEE 1394): Texas Instruments TSB43AB22/A IEEE-1394a-2000 Controller (PHY/Link) > > Userspace is debian lenny on i386. > > Let me know if you need more info. > > Regards > Charlie >

1 0

Re: [alsa-devel] [PATCH] cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection (2nd rev)
by Takashi Iwai 17 Feb '09

17 Feb '09

At Mon, 16 Feb 2009 19:16:33 -0500, Stanley Pinchak wrote: > > Krzysztof and Takashi, > I would like to stress test the new driver modifications. I have > run into some troubles with a program called cmus when outputting to > alsa via libao. The problems occurred during seeking, leaving the > driver in a broken state. The bug seemed to disappear when I used > native alsa output. I don't want to be chasing down application bugs > on your time, so I was hoping you could suggest a way to test the > driver while introducing the least vectors for non alsa code base > bugs. Thanks in advance. Thanks. Let me know if you find out a problem. OTOH, the fact that the native ALSA output works implies that it's rather an application-side problem... BTW, I removed snd-cs4236-lib module now since it's a module solely used by snd-cs4236. Now it's merged into snd-cs4236 by the patch below. Takashi === From cda9043d56cee9fea39e4ee33fd605ae477a1950 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Tue, 17 Feb 2009 08:10:54 +0100 Subject: [PATCH] ALSA: cs4236 - Merge snd-cs4236-lib module into snd-cs4236 Since cs4232 and cs4236 drivers are merged, there is no reason to keep snd-cs4236-lib module separately. Let's merge it into the main driver as well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/cs423x/Makefile | 6 +++--- sound/isa/cs423x/cs4236_lib.c | 24 ------------------------ 2 files changed, 3 insertions(+), 27 deletions(-) diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 732f66c..6d397e8 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -3,11 +3,11 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex(a)perex.cz> # -snd-cs4236-lib-objs := cs4236_lib.o snd-cs4231-objs := cs4231.o -snd-cs4236-objs := cs4236.o +snd-cs4236-objs := cs4236.o cs4236_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o -obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o +obj-$(CONFIG_SND_CS4236) += snd-cs4236.o + diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 2406efd..38835f3 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -88,10 +88,6 @@ #include <sound/wss.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); -MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); -MODULE_LICENSE("GPL"); - /* * */ @@ -1022,23 +1018,3 @@ int snd_cs4236_mixer(struct snd_wss *chip) } return 0; } - -EXPORT_SYMBOL(snd_cs4236_create); -EXPORT_SYMBOL(snd_cs4236_pcm); -EXPORT_SYMBOL(snd_cs4236_mixer); - -/* - * INIT part - */ - -static int __init alsa_cs4236_init(void) -{ - return 0; -} - -static void __exit alsa_cs4236_exit(void) -{ -} - -module_init(alsa_cs4236_init) -module_exit(alsa_cs4236_exit) -- 1.6.1.3

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Re: [alsa-devel] No sound for HP Pavilion DV4z
by Takashi Iwai 17 Feb '09

17 Feb '09

At Mon, 16 Feb 2009 23:28:03 +0000 (UTC), aeleneski(a)comcast.net wrote: > > --startcollapse-- > state.SB { > control.1 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > value true Turn off "Analog Loopback", and... > control.16 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Speaker Playback Switch' > value.0 false > value.1 false Unmute "Speaker" output. Takashi

1 0

Re: [alsa-devel] Intel G45 no sound without monitor
by Wu Fengguang 17 Feb '09

17 Feb '09

On Tue, Feb 17, 2009 at 04:20:57AM +0200, Shane W wrote: > On Tue, Feb 17, 2009 at 09:20:38AM +0800, Wu Fengguang wrote: > > On Mon, Feb 16, 2009 at 11:07:02PM +0200, Shane W wrote: > > > Hello, > > > wasn't known. On my G45, I don't get any sound if my > > > television isn't turned on or is disconnected. That is, > > > HDMI from the G45 to the Yamaha amp is a no go without a > > > connected monitor. Turning on the monitor after system > > > > It's related to our intel gfx driver. > > Ok so in this case, X needs to run for sound to work? In > my test environment, X hadn't been started so the intel-gfx > drivers including i915 kernel module weren't loaded, just > ALSA. Yes Xorg is required for G45+Yamaha. However our G35 board can drive Yamaha directly in console. The i915 kernel module is irrelevant. Thanks, Fengguang

1 0

Re: [alsa-devel] Intel G45 no sound without monitor
by Wu Fengguang 17 Feb '09

17 Feb '09

On Mon, Feb 16, 2009 at 11:07:02PM +0200, Shane W wrote: > Hello, > > Not sure if this is a board hardware issue or something > driver specific but thought I'd put it out there in case it > wasn't known. On my G45, I don't get any sound if my > television isn't turned on or is disconnected. That is, > HDMI from the G45 to the Yamaha amp is a no go without a > connected monitor. Turning on the monitor after system > boot has no effect so monitor must be turned on and system > rebooted for sound to work. Hi Shane, It's related to our intel gfx driver. We bought the same Yamaha RX-V1800 and experienced the same problem. I was doing extensive (black box) debugging on it these days, and found that the following process can produce sound on Yamaha: startx xrandr --output HDMI-1 --mode 1920x1200 In fact the default 1920x540 mode is the only mode in which Yamaha cannot produce sound. Our intel gfx maintainer confirmed that it's actually the 1080i mode that is not supported yet. Thanks, Fengguang

2 1

[alsa-devel] dma buffer format
by York Sun 17 Feb '09

17 Feb '09

Takashi, I am now working on a surround sound driver for the first time. I am confused by the packed format of dma buffer. If the channel number is 6, is the buffer filled with data as front left, front right, rear left, rear right, center, LFE or front left, rear left, center, front right, rear left, LFE? Thanks, York

2 3

[alsa-devel] [PATCH] cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection
by Krzysztof Helt 17 Feb '09

17 Feb '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Also, the patch adds recognition if the chip is cs4236b+ or earlier part. This unifies drivers for both cs4232 and cs4236+ chips. It allows to use the PnP BIOS detection for the cs4236+ chips. Previously, only the snd-cs4232 could be detected by the PnP BIOS. The cs4232+ cards reports two separate PnP BIOS ids. The patch adds search for the second id to find out resources assigned to a control port. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The cs4232 and cs4236 are the same now and one can be removed. Stanley tested this patch and it worked for him (cs4232 chip in Tecra 500). I have tested it on Dell Latitude CP (cs4237b chip) which is now correctly recognized. Previously, it was recognized as a basic cs4236 chip. Also, I have tested it with Maxi Sound 32 PnP card (ISA PnP cs4236b) with success. A side effect of this approach is that on ISA PnP cards the driver reports an error of PnP BIOS detection everytime it is loaded. This is due to driver rejecting the PnP BIOS initialization for ISA PnP cards. Probably, the earlier cs4232 driver worked the same way. diff --git a/include/sound/wss.h b/include/sound/wss.h index fd01f22..6d65f32 100644 --- a/include/sound/wss.h +++ b/include/sound/wss.h @@ -154,6 +154,7 @@ int snd_wss_create(struct snd_card *card, unsigned short hardware, unsigned short hwshare, struct snd_wss **rchip); +int snd_wss_free(struct snd_wss *chip); int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer); int snd_wss_mixer(struct snd_wss *chip); diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 5870ca2..dfdd96d 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -10,6 +10,6 @@ snd-cs4236-objs := cs4236.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o -obj-$(CONFIG_SND_CS4232) += snd-cs4232.o +obj-$(CONFIG_SND_CS4232) += snd-cs4232.o snd-cs4236-lib.o obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index f784598..089b7e8 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -33,17 +33,14 @@ MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_LICENSE("GPL"); -#ifdef CS4232 -MODULE_DESCRIPTION("Cirrus Logic CS4232"); +MODULE_DESCRIPTION("Cirrus Logic CS4232-9"); MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000}," "{Turtle Beach,Tropez Plus}," "{SIC CrystalWave 32}," "{Hewlett Packard,Omnibook 5500}," "{TerraTec,Maestro 32/96}," - "{Philips,PCA70PS}}"); -#else -MODULE_DESCRIPTION("Cirrus Logic CS4235-9"); -MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," + "{Philips,PCA70PS}}," + "{{Crystal Semiconductors,CS4235}," "{Crystal Semiconductors,CS4236}," "{Crystal Semiconductors,CS4237}," "{Crystal Semiconductors,CS4238}," @@ -70,15 +67,9 @@ MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," "{Typhoon Soundsystem,CS4236B}," "{Turtle Beach,Malibu}," "{Unknown,Digital PC 5000 Onboard}}"); -#endif -#ifdef CS4232 -#define IDENT "CS4232" -#define DEV_NAME "cs4232" -#else -#define IDENT "CS4236+" -#define DEV_NAME "cs4236" -#endif +#define IDENT "CS4232+" +#define DEV_NAME "cs4232+" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -128,9 +119,7 @@ MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP static int isa_registered; static int pnpc_registered; -#ifdef CS4232 static int pnp_registered; -#endif #endif /* CONFIG_PNP */ struct snd_card_cs4236 { @@ -145,11 +134,10 @@ struct snd_card_cs4236 { #ifdef CONFIG_PNP -#ifdef CS4232 /* * PNP BIOS */ -static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { +static const struct pnp_device_id snd_cs423x_pnpbiosids[] = { { .id = "CSC0100" }, { .id = "CSC0000" }, /* Guillemot Turtlebeach something appears to be cs4232 compatible @@ -157,10 +145,8 @@ static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { { .id = "GIM0100" }, { .id = "" } }; -MODULE_DEVICE_TABLE(pnp, snd_cs4232_pnpbiosids); -#endif /* CS4232 */ +MODULE_DEVICE_TABLE(pnp, snd_cs423x_pnpbiosids); -#ifdef CS4232 #define CS423X_ISAPNP_DRIVER "cs4232_isapnp" static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Philips PCA70PS */ @@ -179,12 +165,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSCf032", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Netfinity 3000 on-board soundcard */ { .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC010f" } } }, - /* --- */ - { .id = "" } /* end */ -}; -#else /* CS4236 */ -#define CS423X_ISAPNP_DRIVER "cs4236_isapnp" -static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Intel Marlin Spike Motherboard - CS4235 */ { .id = "CSC0225", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Intel Marlin Spike Motherboard (#2) - CS4235 */ @@ -266,7 +246,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* --- */ { .id = "" } /* end */ }; -#endif MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); @@ -323,17 +302,19 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) return 0; } -#ifdef CS4232 -static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, - struct pnp_dev *pdev) +static int __devinit snd_card_cs423x_pnp(int dev, struct snd_card_cs4236 *acard, + struct pnp_dev *pdev, + struct pnp_dev *cdev) { acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; - cport[dev] = -1; + if (cdev) + cport[dev] = pnp_port_start(cdev, 0); + else + cport[dev] = -1; return 0; } -#endif static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard, struct pnp_card_link *card, @@ -411,40 +392,39 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } -#ifdef CS4232 err = snd_wss_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_wss_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_wss_mixer(chip); + WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; - -#else /* CS4236 */ - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_cs4236_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_cs4236_mixer(chip); - if (err < 0) - return err; -#endif + if (chip->hardware & WSS_HW_CS4236B_MASK) { + snd_wss_free(chip); + err = snd_cs4236_create(card, + port[dev], cport[dev], + irq[dev], dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) + return err; + acard->chip = chip; + + err = snd_cs4236_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_cs4236_mixer(chip); + if (err < 0) + return err; + } else { + acard->chip = chip; + err = snd_wss_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_wss_mixer(chip); + if (err < 0) + return err; + } strcpy(card->driver, pcm->name); strcpy(card->shortname, pcm->name); sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", @@ -579,13 +559,15 @@ static struct isa_driver cs423x_isa_driver = { #ifdef CONFIG_PNP -#ifdef CS4232 -static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, +static int __devinit snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, const struct pnp_device_id *id) { static int dev; int err; struct snd_card *card; + struct pnp_dev *cdev; + struct list_head *pos; + char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) return -ENOENT; /* we have another procedure - card */ @@ -596,10 +578,20 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; + /* prepare second id */ + strcpy(cid, pdev->id[0].id); + cid[5] = '1'; + cdev = NULL; + list_for_each(pos, &(pdev->protocol->devices)) { + cdev = list_entry(pos, struct pnp_dev, protocol_list); + if (!strcmp(cdev->id[0].id, cid)) + break; + } err = snd_cs423x_card_new(dev, &card); if (err < 0) return err; - if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) { + err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); + if (err < 0) { printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n"); snd_card_free(card); return err; @@ -614,35 +606,34 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, return 0; } -static void __devexit snd_cs4232_pnp_remove(struct pnp_dev * pdev) +static void __devexit snd_cs423x_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM -static int snd_cs4232_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) +static int snd_cs423x_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) { return snd_cs423x_suspend(pnp_get_drvdata(pdev)); } -static int snd_cs4232_pnp_resume(struct pnp_dev *pdev) +static int snd_cs423x_pnp_resume(struct pnp_dev *pdev) { return snd_cs423x_resume(pnp_get_drvdata(pdev)); } #endif -static struct pnp_driver cs4232_pnp_driver = { - .name = "cs4232-pnpbios", - .id_table = snd_cs4232_pnpbiosids, - .probe = snd_cs4232_pnpbios_detect, - .remove = __devexit_p(snd_cs4232_pnp_remove), +static struct pnp_driver cs423x_pnp_driver = { + .name = "cs423x-pnpbios", + .id_table = snd_cs423x_pnpbiosids, + .probe = snd_cs423x_pnpbios_detect, + .remove = __devexit_p(snd_cs423x_pnp_remove), #ifdef CONFIG_PM - .suspend = snd_cs4232_pnp_suspend, - .resume = snd_cs4232_pnp_resume, + .suspend = snd_cs423x_pnp_suspend, + .resume = snd_cs423x_pnp_resume, #endif }; -#endif /* CS4232 */ static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard, const struct pnp_card_device_id *pid) @@ -716,18 +707,14 @@ static int __init alsa_card_cs423x_init(void) #ifdef CONFIG_PNP if (!err) isa_registered = 1; -#ifdef CS4232 - err = pnp_register_driver(&cs4232_pnp_driver); + err = pnp_register_driver(&cs423x_pnp_driver); if (!err) pnp_registered = 1; -#endif err = pnp_register_card_driver(&cs423x_pnpc_driver); if (!err) pnpc_registered = 1; -#ifdef CS4232 if (pnp_registered) err = 0; -#endif if (isa_registered) err = 0; #endif @@ -739,10 +726,8 @@ static void __exit alsa_card_cs423x_exit(void) #ifdef CONFIG_PNP if (pnpc_registered) pnp_unregister_card_driver(&cs423x_pnpc_driver); -#ifdef CS4232 if (pnp_registered) - pnp_unregister_driver(&cs4232_pnp_driver); -#endif + pnp_unregister_driver(&cs423x_pnp_driver); if (isa_registered) #endif isa_unregister_driver(&cs423x_isa_driver); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index aab44e7..5d2ba1b 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1682,7 +1682,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -static int snd_wss_free(struct snd_wss *chip) +int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1705,6 +1705,7 @@ static int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } +EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { ---------------------------------------------------------------------- Zostan mistrzem parkowania w Bombaju! 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[alsa-devel] [PATCH] Workaround for misdetected sample rate with CM6207
by Joris van Rantwijk 17 Feb '09

17 Feb '09

Summary: Workaround for misdetected sample rate with CM6207 The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz in its USB device descriptor. This patch extends an existing workaround in usbaudio.c to also cover the CM6207. This resolves issue 0004249 in the ALSA bug tracker. Signed-off by: Joris van Rantwijk <jorispubl(a)xs4all.nl> diff -ur linux-2.6.28.4/sound/usb/usbaudio.c linux-2.6.28.4_j/sound/usb/usbaudio.c --- linux-2.6.28.4/sound/usb/usbaudio.c 2009-02-13 21:45:17.000000000 +0100 +++ linux-2.6.28.4_j/sound/usb/usbaudio.c 2009-02-13 22:00:44.000000000 +0100 @@ -2533,19 +2533,20 @@ } fp->nr_rates = nr_rates; - fp->rate_min = fp->rate_max = combine_triple(&fmt[8]); + fp->rate_min = fp->rate_max = 0; for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { unsigned int rate = combine_triple(&fmt[idx]); /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && - chip->usb_id == USB_ID(0x0d8c, 0x0201) && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; fp->rate_table[r] = rate; nonzero_rates |= rate; - if (rate < fp->rate_min) + if (r == 0 || rate < fp->rate_min) fp->rate_min = rate; - else if (rate > fp->rate_max) + else if (r == 0 || rate > fp->rate_max) fp->rate_max = rate; fp->rates |= snd_pcm_rate_to_rate_bit(rate); }

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