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[alsa-devel] Mic/Line-in record is not working with wm9713 AC97
by Sreekanth W 14 Oct '09

14 Oct '09

Hi, I am trying to record a wav file using mic/line-in with wm9713 driver. The Wav file is created, only header is created but almost there is no data. This is seen using CoolEdit/Vim editor. If I try to play the recorded file, there is no sound. But, playing works for other existing files, able to using aplay. I am issuing the following command to record # ./arecord –t wav –c 1 –r 8000 –d 10 –f S16_LE test.wav The below is the dump of /sys/devices/platform/soc-audio/dapm_widget, while "arecord" is running (using mic): # cat /sys/devices/platform/soc-audio/dapm_widget Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right Voice ADC: Off Left Voice ADC: Off Right HiFi ADC: On Left HiFi ADC: On Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off Line Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: On I'm using DMA for data transfer, and getting the callback function after DMA completed. Also I'm printing the number of Bytes transfered. Any idea why record is not working? Anything wrong with these settings? Regards, Sreekanth

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[alsa-devel] Channel conversion in plughw not working for my driver... confused about snd_pcm_plug_hw_params()
by David Huggins-Daines 14 Oct '09

14 Oct '09

Hi, I'm trying to make mono output work on the audio driver for the Cirrus EDB9302 evaluation board. For some reason, plughw is refusing to do any channel conversion. When I try to play a mono file, I simply get: aplay: set_params:1022: Unable to install hw params: The driver's struct snd_pcm_hardware contains: .channels_min = 2, .channels_max = 2, which I believe is the correct way to tell userspace "this driver can only do stereo output" - at least, this is how intel8x0.c does it, and that driver really ought to work by now! I've traced this to the exact point of failure, and what I see is that in snd_pcm_plug_hw_params(), despite figuring out exactly what the set of hardware parameters will work for the slave device, it goes and calls _snd_pcm_hw_params() with the actual parameters requested by the client, which obviously fails, and leads to the error shown above. I added some calls to snd_pcm_hw_params_dump() which give this output. The first is sparams, which is the refined set of parameters which I assume are supposed to be set in the slave device. The second is params, which is the set of parameters requested by the client (aplay in this case). sparams are: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 16 CHANNELS: 2 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 2756 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 11026 TICK_TIME: 0 params are: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 1378 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 5513 TICK_TIME: 0 inserting plugins whatever that means in snd_pcm_plug_change_channels! So, yeah, that's great, it figured it all out, and realized that it had to do channel conversion, which presumably is done in snd_pcm_plug_change_channels(). But now, we see that it goes ahead and tries to set these parameters in the slave device: calling _snd_pcm_hw_params with params: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 1378 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 5513 TICK_TIME: 0 But that's wrong! The slave device can't do single-channel input! So what gives? This can't possibly be broken, or nobody's Intel on-board audio would ever work at all... so what is my driver doing wrong?

1 0

[alsa-devel] SPDIF with X-Fi
by Daniel L. Miller 14 Oct '09

14 Oct '09

Perhaps I'm going about this the wrong way (if so, please tell me), but... My X-Fi card is now working with the new driver. I am connected to my amplifier via a fiber optic cable. Sending standard two-channel audio to the "front" alsa plug gets directed to the spdif output and sounds great. But I want to have my full surround active - particularly for movies already encoded with dts and/or ac-3. Do I need to specially configure ALSA is any fashion to make this work? Do I need to use the iecset tool? I'm using Kubuntu Karmic, running a 2.6.31 kernel, if that makes any difference. -- Daniel

2 1

[alsa-devel] [PATCH 1/1] Echoaudio: fix broken Mia driver
by Jordan 14 Oct '09

14 Oct '09

2 1

[alsa-devel] [PATCH] aaci: ARM1176 aaci-pl041 AC97 register read timeout
by Philby John 13 Oct '09

13 Oct '09

From: Philby John <pjohn(a)in.mvista.com> Date: Tue, 13 Oct 2009 16:30:22 +0530 Subject: [PATCH] aaci: ARM1176 aaci-pl041 AC97 register read timeout After a reboot on an ARM1176 which amounts to a softreset, it has been noted that the ALSA driver does not get registered and the probe fails with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process of reading from a register the SL1TxBusy bit is set indicating that the transceiver is busy and remains so until the default timeout occurs. Set the Power down register 0x26 to an arbitrary value as specified in the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take their default state. Signed-off-by: Philby John <pjohn(a)in.mvista.com> --- sound/arm/aaci.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 89096e8..98b6ffe 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; + writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ -- 1.6.3.3.MVISTA

2 1

[alsa-devel] S3C: Remove <plat/audio.h>
by Ben Dooks 13 Oct '09

13 Oct '09

Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h as it provides nothing to the current kernel and is not in any future plans for the system. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> Signed-off-by: Simtec Linux Team <linux(a)simtec.co.uk> --- arch/arm/plat-s3c/include/plat/audio.h | 45 --------------------------------- sound/soc/s3c24xx/neo1973_wm8753.c | 1 sound/soc/s3c24xx/s3c-i2s-v2.c | 1 sound/soc/s3c24xx/s3c2412-i2s.c | 1 sound/soc/s3c24xx/s3c2443-ac97.c | 1 sound/soc/s3c24xx/s3c24xx-i2s.c | 2 - sound/soc/s3c24xx/s3c24xx-pcm.c | 1 sound/soc/s3c24xx/s3c64xx-i2s.c | 1 8 files changed, 1 insertion(+), 52 deletions(-) Index: b/arch/arm/plat-s3c/include/plat/audio.h =================================================================== --- a/arch/arm/plat-s3c/include/plat/audio.h 2009-10-12 21:11:08.000000000 +0100 +++ /dev/null 1970-01-01 00:00:00.000000000 +0000 @@ -1,45 +0,0 @@ -/* arch/arm/mach-s3c2410/include/mach/audio.h - * - * Copyright (c) 2004-2005 Simtec Electronics - * http://www.simtec.co.uk/products/SWLINUX/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * S3C24XX - Audio platfrom_device info - * - * 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 __ASM_ARCH_AUDIO_H -#define __ASM_ARCH_AUDIO_H __FILE__ - -/* struct s3c24xx_iis_ops - * - * called from the s3c24xx audio core to deal with the architecture - * or the codec's setup and control. - * - * the pointer to itself is passed through in case the caller wants to - * embed this in an larger structure for easy reference to it's context. -*/ - -struct s3c24xx_iis_ops { - struct module *owner; - - int (*startup)(struct s3c24xx_iis_ops *me); - void (*shutdown)(struct s3c24xx_iis_ops *me); - int (*suspend)(struct s3c24xx_iis_ops *me); - int (*resume)(struct s3c24xx_iis_ops *me); - - int (*open)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); - int (*close)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); - int (*prepare)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm, struct snd_pcm_runtime *rt); -}; - -struct s3c24xx_platdata_iis { - const char *codec_clk; - struct s3c24xx_iis_ops *ops; - int (*match_dev)(struct device *dev); -}; - -#endif /* __ASM_ARCH_AUDIO_H */ Index: b/sound/soc/s3c24xx/s3c-i2s-v2.c =================================================================== --- a/sound/soc/s3c24xx/s3c-i2s-v2.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c 2009-10-12 21:11:39.000000000 +0100 @@ -32,7 +32,6 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/dma.h> #include "s3c-i2s-v2.h" Index: b/sound/soc/s3c24xx/s3c2412-i2s.c =================================================================== --- a/sound/soc/s3c24xx/s3c2412-i2s.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c2412-i2s.c 2009-10-12 21:11:39.000000000 +0100 @@ -34,7 +34,6 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/regs-gpio.h> #include <mach/dma.h> Index: b/sound/soc/s3c24xx/s3c2443-ac97.c =================================================================== --- a/sound/soc/s3c24xx/s3c2443-ac97.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c2443-ac97.c 2009-10-12 21:11:39.000000000 +0100 @@ -32,7 +32,6 @@ #include <plat/regs-ac97.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> #include <asm/dma.h> #include <mach/dma.h> Index: b/sound/soc/s3c24xx/s3c24xx-i2s.c =================================================================== --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-10-12 21:11:39.000000000 +0100 @@ -32,7 +32,7 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> + #include <asm/dma.h> #include <mach/dma.h> Index: b/sound/soc/s3c24xx/s3c24xx-pcm.c =================================================================== --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-10-12 21:11:39.000000000 +0100 @@ -29,7 +29,6 @@ #include <asm/dma.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <plat/audio.h> #include "s3c24xx-pcm.h" Index: b/sound/soc/s3c24xx/neo1973_wm8753.c =================================================================== --- a/sound/soc/s3c24xx/neo1973_wm8753.c 2009-10-12 21:13:18.000000000 +0100 +++ b/sound/soc/s3c24xx/neo1973_wm8753.c 2009-10-12 21:13:29.000000000 +0100 @@ -29,7 +29,6 @@ #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h> -#include <plat/audio.h> #include <linux/io.h> #include <mach/spi-gpio.h> Index: b/sound/soc/s3c24xx/s3c64xx-i2s.c =================================================================== --- a/sound/soc/s3c24xx/s3c64xx-i2s.c 2009-10-12 21:13:35.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c 2009-10-12 21:13:40.000000000 +0100 @@ -31,7 +31,6 @@ #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> -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] [PATCH 0/1] ASoC: Serialize access to dapm_power_widgets()
by ext-eero.nurkkala＠nokia.com 13 Oct '09

13 Oct '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> It was detected that in some rare instances the dapm powering cycles got out of order. That lead to investigation of the use of all the mutexes used in the ASoC core. The patch is tested with 2.6.28 (with lockdep also) , but applies to v2.6.32-rc3. Please review.

2 3

[alsa-devel] Long-standing SDL ALSA bug
by Sam Lantinga 13 Oct '09

13 Oct '09

Hey guys, we're in the process of getting ready to release SDL 1.2.14, and we're trying to fix a long standing issue with crackling and pops in ALSA with some configurations: http://bugzilla.libsdl.org/show_bug.cgi?id=650 What's really puzzling is that we're not able to reproduce it ourselves at all. I'm attaching the latest versions of the SDL audio files, and I'd really appreciate it if you take a look and sanity check our code. You can get the full pre-release here: http://www.libsdl.org/tmp/SDL-1.2.zip Thanks! -- -Sam Lantinga, Founder and President, Galaxy Gameworks LLC

4 4

[alsa-devel] [PATCH] ASoC: Codec driver for Texas Instruments tlv320dac33 codec
by Peter Ujfalusi 13 Oct '09

13 Oct '09

Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- include/sound/tlv320dac33-plat.h | 20 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320dac33.c | 1178 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320dac33.h | 267 +++++++++ 5 files changed, 1471 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320dac33-plat.h create mode 100644 sound/soc/codecs/tlv320dac33.c create mode 100644 sound/soc/codecs/tlv320dac33.h diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h new file mode 100644 index 0000000..5858d06 --- /dev/null +++ b/include/sound/tlv320dac33-plat.h @@ -0,0 +1,20 @@ +/* + * Platform header for Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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 __TLV320DAC33_PLAT_H +#define __TLV320DAC33_PLAT_H + +struct tlv320dac33_platform_data { + int power_gpio; +}; + +#endif /* __TLV320DAC33_PLAT_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fab01c9..49bab93 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X @@ -142,6 +143,9 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate +config SND_SOC_TLV320DAC33 + tristate + config SND_SOC_TWL4030 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2f14391..8f519ee 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -70,6 +71,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(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-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(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/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c new file mode 100644 index 0000000..6170b5d --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.c @@ -0,0 +1,1178 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <sound/tlv320dac33-plat.h> +#include "tlv320dac33.h" + +#define DAC33_BUFFER_SIZE_BYTES 24576 /* bytes, 12288 16 bit words, + * 6144 stereo */ +#define DAC33_BUFFER_SIZE_SAMPLES 6144 + +#define NSAMPLE_MAX 5700 + +#define LATENCY_TIME_MS 20 + +static struct i2c_client *tlv320dac33_client; +static struct snd_soc_codec *tlv320dac33_codec; + +enum dac33_state { + DAC33_IDLE = 0, + DAC33_PREFILL, + DAC33_PLAYBACK, + DAC33_FLUSH, +}; + +struct tlv320dac33_priv { + struct mutex mutex; + struct work_struct work; + struct snd_soc_codec codec; + int power_gpio; + int power_state; + int irq; + unsigned int refclk; + + unsigned int alarm_threshold; /* set to be half of LATENCY_TIME_MS */ + unsigned int nsample_min; /* nsample should not be lower than + * this */ + unsigned int nsample_max; /* nsample should not be higher than + * this */ + unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + unsigned int nsmaple_cmode; /* Currently used nsample mode */ + unsigned int nsample; /* burst read amount from host */ + + enum dac33_state state; +}; + +static struct workqueue_struct *dac33_wq; + +static const u8 dac33_reg[DAC33_CACHEREGNUM] = { +0x00, 0x00, 0x00, 0x00, /* 0x00 - 0x03 */ +0x00, 0x00, 0x00, 0x00, /* 0x04 - 0x07 */ +0x00, 0x00, 0x00, 0x00, /* 0x08 - 0x0b */ +0x00, 0x00, 0x00, 0x00, /* 0x0c - 0x0f */ +0x00, 0x00, 0x00, 0x00, /* 0x10 - 0x13 */ +0x00, 0x00, 0x00, 0x00, /* 0x14 - 0x17 */ +0x00, 0x00, 0x00, 0x00, /* 0x18 - 0x1b */ +0x00, 0x00, 0x00, 0x00, /* 0x1c - 0x1f */ +0x00, 0x00, 0x00, 0x00, /* 0x20 - 0x23 */ +0x00, 0x00, 0x00, 0x00, /* 0x24 - 0x27 */ +0x00, 0x00, 0x00, 0x00, /* 0x28 - 0x2b */ +0x00, 0x00, 0x00, 0x80, /* 0x2c - 0x2f */ +0x80, 0x00, 0x00, 0x00, /* 0x30 - 0x33 */ +0x00, 0x00, 0x00, 0x00, /* 0x34 - 0x37 */ +0x00, 0x00, /* 0x38 - 0x39 */ +/* Registers 0x3a - 0x3f are reserved */ + 0x00, 0x00, /* 0x3a - 0x3b */ +0x00, 0x00, 0x00, 0x00, /* 0x3c - 0x3f */ + +0x00, 0x00, 0x00, 0x00, /* 0x40 - 0x43 */ +0x00, 0x80, /* 0x44 - 0x45 */ +/* Registers 0x46 - 0x47 are reserved */ + 0x80, 0x80, /* 0x46 - 0x47 */ + +0x80, 0x00, 0x00, /* 0x48 - 0x4a */ +/* Registers 0x4b - 0x7c are reserved */ + 0x00, /* 0x4b */ +0x00, 0x00, 0x00, 0x00, /* 0x4c - 0x4f */ +0x00, 0x00, 0x00, 0x00, /* 0x50 - 0x53 */ +0x00, 0x00, 0x00, 0x00, /* 0x54 - 0x57 */ +0x00, 0x00, 0x00, 0x00, /* 0x58 - 0x5b */ +0x00, 0x00, 0x00, 0x00, /* 0x5c - 0x5f */ +0x00, 0x00, 0x00, 0x00, /* 0x60 - 0x63 */ +0x00, 0x00, 0x00, 0x00, /* 0x64 - 0x67 */ +0x00, 0x00, 0x00, 0x00, /* 0x68 - 0x6b */ +0x00, 0x00, 0x00, 0x00, /* 0x6c - 0x6f */ +0x00, 0x00, 0x00, 0x00, /* 0x70 - 0x73 */ +0x00, 0x00, 0x00, 0x00, /* 0x74 - 0x77 */ +0x00, 0x00, 0x00, 0x00, /* 0x78 - 0x7b */ +0x00, /* 0x7c */ + + 0xda, 0x33, 0x03, /* 0x7d - 0x7f */ +}; + +/* Register read and write */ +static inline unsigned int dac33_read_reg_cache(struct snd_soc_codec *codec, + unsigned reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return 0; + + return cache[reg]; +} + +static inline void dac33_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return; + + cache[reg] = value; +} + +static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int val; + + *value = reg & 0xff; + + /* If powered off, return the cached value */ + mutex_lock(&dac33->mutex); + if (dac33->power_state) { + val = i2c_smbus_read_byte_data(codec->control_data, value[0]); + if (val < 0) { + dev_err(codec->dev, "Read failed\n"); + value[0] = dac33_read_reg_cache(codec, reg); + } else { + value[0] = val; + dac33_write_reg_cache(codec, reg, val); + } + + } else { + value[0] = dac33_read_reg_cache(codec, reg); + } + mutex_unlock(&dac33->mutex); + + return 0; +} + +static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[2]; + int ret = 0; + + /* + * data is + * D15..D8 dac33 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + mutex_lock(&dac33->mutex); + if (dac33->power_state) { + if (codec->hw_write(codec->control_data, data, 2) != 2) { + dev_err(codec->dev, "Write failed\n"); + ret = -EIO; + } + } + mutex_unlock(&dac33->mutex); + + return ret; +} + +#define DAC33_I2C_ADDR_AUTOINC 0x80 +static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[3]; + int ret = 0; + + /* + * data is + * D23..D16 dac33 register offset + * D15..D8 register data MSB + * D7...D0 register data LSB + */ + data[0] = reg & 0xff; + data[1] = (value >> 8) & 0xff; + data[2] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + dac33_write_reg_cache(codec, data[0] + 1, data[2]); + + mutex_lock(&dac33->mutex); + if (dac33->power_state) { + /* We need to set autoincrement mode for 16 bit writes */ + data[0] |= DAC33_I2C_ADDR_AUTOINC; + if (codec->hw_write(codec->control_data, data, 3) != 3) + ret = -EIO; + } + mutex_unlock(&dac33->mutex); + + return ret; +} + +static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) +{ + u8 reg; + + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + if (power) + reg |= PDNALLB; + else + reg &= ~PDNALLB; + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + +static void dac33_set_power(struct snd_soc_codec *codec, int power) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + + if (power) { + if (dac33->power_gpio >= 0) { + mutex_lock(&dac33->mutex); + gpio_set_value(dac33->power_gpio, 1); + dac33->power_state = 1; + mutex_unlock(&dac33->mutex); + } + dac33_soft_power(codec, 1); + } else { + dac33_soft_power(codec, 0); + if (dac33->power_gpio >= 0) { + mutex_lock(&dac33->mutex); + gpio_set_value(dac33->power_gpio, 0); + dac33->power_state = 0; + mutex_unlock(&dac33->mutex); + } + } + +} + +static int dac33_get_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample; + + return 0; +} + +static int dac33_set_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = -EINVAL; + + if (dac33->nsample == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + else if (ucontrol->value.integer.value[0] > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; + else { + dac33->nsample = ucontrol->value.integer.value[0]; + ret = 0; + } + + return ret; +} + +static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample_switch; + + return 0; +} + +static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = -EINVAL; + + if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] < 0) + dac33->nsample_switch = 0; + else if (ucontrol->value.integer.value[0] > 1) + dac33->nsample_switch = 1; + else { + dac33->nsample_switch = ucontrol->value.integer.value[0]; + ret = 0; + } + + return ret; +} + +/* + * DACL/R digital volume control: + * from 0 dB to -63.5 in 0.5 dB steps + * Need to be inverted later on: + * 0x00 == 0 dB + * 0x7f == -63.5 dB + */ +static DECLARE_TLV_DB_SCALE(dac_digivol_tlv, -6350, 50, 0); + +static const struct snd_kcontrol_new dac33_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC Digital Playback Volume", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, + 0, 0x7f, 1, dac_digivol_tlv), + SOC_DOUBLE_R("DAC Digital Playback Switch", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1), + SOC_DOUBLE_R("Line to Line Out Volume", + DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, + dac33_get_nsample_switch, dac33_set_nsample_switch), +}; + +/* Analog bypass */ +static const struct snd_kcontrol_new dac33_dapm_abypassl_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); + +static const struct snd_kcontrol_new dac33_dapm_abypassr_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1); + +static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("LEFT_LO"), + SND_SOC_DAPM_OUTPUT("RIGHT_LO"), + + SND_SOC_DAPM_INPUT("LINEL"), + SND_SOC_DAPM_INPUT("LINER"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", DAC33_LDAC_PWR_CTRL, 2, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", DAC33_RDAC_PWR_CTRL, 2, 0), + + /* Analog bypass */ + SND_SOC_DAPM_SWITCH("Analog Left Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassl_control), + SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassr_control), + + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Analog bypass */ + {"Analog Left Bypass", "Switch", "LINEL"}, + {"Analog Right Bypass", "Switch", "LINER"}, + + {"Output Left Amp Power", NULL, "DACL"}, + {"Output Right Amp Power", NULL, "DACR"}, + + {"Output Left Amp Power", NULL, "Analog Left Bypass"}, + {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + + /* output */ + {"LEFT_LO", NULL, "Output Left Amp Power"}, + {"RIGHT_LO", NULL, "Output Right Amp Power"}, +}; + +static int dac33_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int dac33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dac33_soft_power(codec, 1); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + dac33_soft_power(codec, 0); + break; + case SND_SOC_BIAS_OFF: + dac33_set_power(codec, 0); + break; + } + codec->bias_level = level; + + return 0; +} + +static void dac33_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + u8 reg; + + dac33 = container_of(work, struct tlv320dac33_priv, work); + codec = &dac33->codec; + + switch (dac33->state) { + case DAC33_PREFILL: + dac33->state = DAC33_PLAYBACK; + dac33_write16(codec, DAC33_NSAMPLE_MSB, + THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + THRREG(dac33->alarm_threshold)); + break; + case DAC33_PLAYBACK: + dac33_write16(codec, DAC33_NSAMPLE_MSB, THRREG(dac33->nsample)); + break; + case DAC33_IDLE: + break; + case DAC33_FLUSH: + /* flush fifo */ + dac33->state = DAC33_IDLE; + reg = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + reg |= FIFOFLUSH; + dac33_write(codec, DAC33_FIFO_CTRL_A, reg); + break; + } +} + +static irqreturn_t dac33_interrupt_handler(int irq, void *dev) +{ + struct snd_soc_codec *codec = dev; + struct tlv320dac33_priv *dac33 = codec->private_data; + + if (dac33->state == DAC33_PLAYBACK) + queue_work(dac33_wq, &dac33->work); + + return IRQ_HANDLED; +} + +static int dac33_startup(struct snd_pcm_substream *substream, + 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; + struct tlv320dac33_priv *dac33 = codec->private_data; + int err; + + /* Save the nsample mode to be used until the stream terminates */ + dac33->nsmaple_cmode = dac33->nsample_switch; + + /* We only need the interrupt in nSample mode */ + if (dac33->nsmaple_cmode) { + err = request_irq(dac33->irq, dac33_interrupt_handler, + IRQF_TRIGGER_RISING | IRQF_DISABLED, + codec->name, codec); + if (err < 0) { + dev_err(codec->dev, "Could not request IRQ\n"); + return err; + } + } + + return 0; +} + +static void dac33_shutdown(struct snd_pcm_substream *substream, + 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; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int pwr_ctrl; + + /* Only in nSample mode */ + if (dac33->nsmaple_cmode) + free_irq(dac33->irq, codec); + + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl &= ~(OSCPDNB | DACRPDNB | DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); +} + +static void dac33_oscwait(struct snd_soc_codec *codec) +{ + int timeout = 20; + u8 reg; + + do { + msleep(1); + dac33_read(codec, DAC33_INT_OSC_STATUS, &reg); + } while (((reg & 0x03) != OSCSTATUS_NORMAL) && timeout--); + if ((reg & 0x03) != OSCSTATUS_NORMAL) + dev_err(codec->dev, + "internal oscillator calibration failed\n"); +} + +static int dac33_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; + + /* Check parameters for validity */ + switch (params_rate(params)) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + params_rate(params)); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + params_format(params)); + return -EINVAL; + } + + return 0; +} + +#define CALC_OSCSET(rate, refclk) ( \ + ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) +#define CALC_RATIOSET(rate, refclk) ( \ + ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) + +/* + * tlv320dac33 is strict on the sequence of the register writes, if the register + * writes happens in different order, than dac33 might end up in unknown state. + * Use the known, working sequence of register writes to initialize the dac33. + */ +static int dac33_prepare_chip(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; + u8 aictrl_a, fifoctrl_a; + + switch (substream->runtime->rate) { + case 44100: + case 48000: + oscset = CALC_OSCSET(substream->runtime->rate, dac33->refclk); + ratioset = CALC_RATIOSET(substream->runtime->rate, + dac33->refclk); + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + substream->runtime->rate); + return -EINVAL; + } + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_a &= ~(NCYCL_MASK | WLEN_MASK); + fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~WIDTH; + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + aictrl_a |= (NCYCL_16 | WLEN_16); + fifoctrl_a |= WIDTH; + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + substream->runtime->format); + return -EINVAL; + } + dac33_soft_power(codec, 1); + + reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + dac33_write(codec, DAC33_INT_OSC_CTRL, reg_tmp); + + /* Write registers 0x08 and 0x09 (MSB, LSB) */ + dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset); + + /* calib time: 128 is a nice number ;) */ + dac33_write(codec, DAC33_CALIB_TIME, 128); + + /* adjustment treshold & step */ + dac33_write(codec, DAC33_INT_OSC_CTRL_B, ADJTHRSHLD(2) | ADJSTEP(1)); + + /* div=4 / gain=1 / div */ + dac33_write(codec, DAC33_INT_OSC_CTRL_C, REFDIV(4)); + + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl |= OSCPDNB | DACRPDNB | DACLPDNB; + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + + dac33_oscwait(codec); + + if (dac33->nsmaple_cmode) { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ + + /* Write registers 0x34 and 0x35 (MSB, LSB) */ + dac33_write16(codec, DAC33_SRC_REF_CLK_RATIO_A, ratioset); + + /* Set interrupts to high active */ + dac33_write(codec, DAC33_INTP_CTRL_A, INTPM_AHIGH); + + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + ATM(FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, MAT); + } else { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, SRCBYP); + dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ + } + + if (dac33->nsmaple_cmode) + fifoctrl_a &= ~FBYPAS; + else + fifoctrl_a |= FBYPAS; + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + if (dac33->nsmaple_cmode) + reg_tmp &= ~BCLKON; + else + reg_tmp |= BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + + if (dac33->nsmaple_cmode) { + /* 20: BCLK divide ratio */ + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + + dac33_write16(codec, DAC33_ATHR_MSB, + THRREG(dac33->alarm_threshold)); + } else { + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + } + + return 0; +} + +static void dac33_calculate_times(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int nsample_limit; + + /* Number of samples (16bit, stereo) in one period */ + dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; + + /* Number of samples (16bit, stereo) in ALSA buffer */ + dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; + /* Subtract one period from the total */ + dac33->nsample_max -= dac33->nsample_min; + + /* Number of samples for LATENCY_TIME_MS / 2 */ + dac33->alarm_threshold = substream->runtime->rate / + (1000 / (LATENCY_TIME_MS / 2)); + + /* Find and fix up the lowest nsmaple limit */ + nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); + + if (dac33->nsample_min < nsample_limit) + dac33->nsample_min = nsample_limit; + + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + + /* + * Find and fix up the highest nsmaple limit + * In order to not overflow the DAC33 buffer substract the + * alarm_threshold value from the size of the DAC33 buffer + */ + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; + + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; +} + +static int dac33_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + dac33_calculate_times(substream); + dac33_prepare_chip(substream); + + return 0; +} + +static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + 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; + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (dac33->nsmaple_cmode) { + dac33->state = DAC33_PREFILL; + queue_work(dac33_wq, &dac33->work); + } + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (dac33->nsmaple_cmode) { + dac33->state = DAC33_FLUSH; + queue_work(dac33_wq, &dac33->work); + } + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int dac33_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 ioc_reg, asrcb_reg; + + ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + asrcb_reg = dac33_read_reg_cache(codec, DAC33_ASRC_CTRL_B); + switch (clk_id) { + case TLV320DAC33_MCLK: + ioc_reg |= REFSEL; + asrcb_reg |= SRCREFSEL; + break; + case TLV320DAC33_SLEEPCLK: + ioc_reg &= ~REFSEL; + asrcb_reg &= ~SRCREFSEL; + break; + default: + dev_err(codec->dev, "Invalid clock ID (%d)\n", clk_id); + break; + } + dac33->refclk = freq; + + dac33_write_reg_cache(codec, DAC33_INT_OSC_CTRL, ioc_reg); + dac33_write_reg_cache(codec, DAC33_ASRC_CTRL_B, asrcb_reg); + + return 0; +} + +static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 aictrl_a, aictrl_b; + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec Master */ + aictrl_a |= (MSBCLK | MSWCLK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* Codec Slave */ + aictrl_a &= ~(MSBCLK | MSWCLK); + break; + default: + return -EINVAL; + } + + aictrl_a &= ~AFMT_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aictrl_a |= AFMT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + aictrl_a |= AFMT_DSP; + aictrl_b &= ~DATA_DELAY_MASK; + aictrl_b |= DATA_DELAY(1); /* 1 bit delay */ + break; + case SND_SOC_DAIFMT_DSP_B: + aictrl_a |= AFMT_DSP; + aictrl_b &= ~DATA_DELAY_MASK; /* No delay */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + aictrl_a |= AFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + aictrl_a |= AFMT_LEFT_J; + break; + default: + dev_err(codec->dev, "Unsupported format (%u)\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); + + return 0; +} + +static void dac33_init_chip(struct snd_soc_codec *codec) +{ + /* 44-46: DAC Control Registers */ + /* A : DAC sample rate Fsref/1.5 */ + dac33_write(codec, DAC33_DAC_CTRL_A, DACRATE(1)); + /* B : DAC src=normal, not muted */ + dac33_write(codec, DAC33_DAC_CTRL_B, DACSRCR_RIGHT | DACSRCL_LEFT); + /* C : (defaults) */ + dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); + + /* 64-65 : L&R DAC power control + Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ + dac33_write(codec, DAC33_LDAC_PWR_CTRL, LROUT_GAIN(2)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, LROUT_GAIN(2)); + + /* 73 : volume soft stepping control, + clock source = internal osc (?) */ + dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, VOLCLKEN); + + /* 66 : LOP/LOM Modes */ + dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); + + /* 68 : LOM inverted from LOP */ + dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); + + dac33_write(codec, DAC33_PWR_CTRL, PDNALLB); +} + +static int dac33_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + int ret = 0; + + BUG_ON(!tlv320dac33_codec); + + codec = tlv320dac33_codec; + socdev->card->codec = codec; + dac33 = codec->private_data; + + /* Power up the codec */ + dac33_set_power(codec, 1); + /* Set default configuration */ + dac33_init_chip(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"); + goto pcm_err; + } + + snd_soc_add_controls(codec, dac33_snd_controls, + ARRAY_SIZE(dac33_snd_controls)); + dac33_add_widgets(codec); + + /* power on device */ + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto card_err; + } + + return 0; +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + dac33_set_power(codec, 0); + return ret; +} + +static int dac33_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +static int dac33_soc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int dac33_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_power(codec, 1); + dac33_init_chip(codec); + + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dac33_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_tlv320dac33 = { + .probe = dac33_soc_probe, + .remove = dac33_soc_remove, + .suspend = dac33_soc_suspend, + .resume = dac33_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); + +#define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) +#define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops dac33_dai_ops = { + .startup = dac33_startup, + .shutdown = dac33_shutdown, + .hw_params = dac33_hw_params, + .prepare = dac33_pcm_prepare, + .trigger = dac33_pcm_trigger, + .set_sysclk = dac33_set_dai_sysclk, + .set_fmt = dac33_set_dai_fmt, +}; + +struct snd_soc_dai dac33_dai = { + .name = "tlv320dac33", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DAC33_RATES, + .formats = DAC33_FORMATS,}, + .ops = &dac33_dai_ops, +}; +EXPORT_SYMBOL_GPL(dac33_dai); + +static int dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_priv *dac33; + struct snd_soc_codec *codec; + int ret = 0; + + if (client->dev.platform_data == NULL) { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } + pdata = (struct tlv320dac33_platform_data *)client->dev.platform_data; + + dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL); + if (dac33 == NULL) + return -ENOMEM; + + codec = &dac33->codec; + codec->private_data = dac33; + codec->control_data = client; + + mutex_init(&codec->mutex); + mutex_init(&dac33->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "tlv320dac33"; + codec->owner = THIS_MODULE; + codec->read = dac33_read_reg_cache; + codec->write = dac33_write; + codec->hw_write = (hw_write_t) i2c_master_send; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = dac33_set_bias_level; + codec->dai = &dac33_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(dac33_reg); + codec->reg_cache = kmemdup(dac33_reg, ARRAY_SIZE(dac33_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_reg; + } + + tlv320dac33_client = client; + + i2c_set_clientdata(tlv320dac33_client, dac33); + + dac33->power_gpio = pdata->power_gpio; + dac33->irq = client->irq; + dac33->nsample = NSAMPLE_MAX; + /* Disable FIFO use by default */ + dac33->nsample_switch = 0; + + tlv320dac33_codec = codec; + + /* Setup work queue */ + dac33_wq = create_rt_workqueue("tlv320dac33"); + if (dac33_wq == NULL) { + ret = -ENOMEM; + goto error_wq; + } + + INIT_WORK(&dac33->work, dac33_work); + + codec->dev = &client->dev; + dac33_dai.dev = codec->dev; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dai(&dac33_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } + + if (dac33->power_gpio >= 0) { + ret = gpio_request(dac33->power_gpio, "tlv320dac33 reset"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to request reset GPIO (%d)\n", + dac33->power_gpio); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(codec); + goto error_codec; + } + gpio_direction_output(dac33->power_gpio, 0); + } + + dac33->power_state = 1; + + /* Shut down the codec for now */ + dac33_soft_power(codec, 0); + + return ret; + +error_codec: + destroy_workqueue(dac33_wq); +error_wq: + kfree(codec->reg_cache); +error_reg: + tlv320dac33_codec = NULL; + kfree(dac33); + + return ret; +} + +static int dac33_i2c_remove(struct i2c_client *client) +{ + struct tlv320dac33_priv *dac33; + + dac33 = i2c_get_clientdata(client); + dac33_set_power(&dac33->codec, 0); + + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); + + destroy_workqueue(dac33_wq); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(&dac33->codec); + kfree(dac33->codec.reg_cache); + kfree(dac33); + tlv320dac33_codec = NULL; + tlv320dac33_client = NULL; + + return 0; +} + +static const struct i2c_device_id tlv320dac33_i2c_id[] = { + { + .name = "tlv320dac33", + .driver_data = 0, + }, + { }, +}; + +static struct i2c_driver tlv320dac33_i2c_driver = { + .driver = { + .name = "tlv320dac33", + .owner = THIS_MODULE, + }, + .probe = dac33_i2c_probe, + .remove = __devexit_p(dac33_i2c_remove), + .id_table = tlv320dac33_i2c_id, +}; + +static int __init dac33_module_init(void) +{ + int r; + r = i2c_add_driver(&tlv320dac33_i2c_driver); + if (r < 0) { + printk(KERN_ERR "DAC33: driver registration failed\n"); + return r; + } + return 0; +} +module_init(dac33_module_init); + +static void __exit dac33_module_exit(void) +{ + i2c_del_driver(&tlv320dac33_i2c_driver); +} +module_exit(dac33_module_exit); + + +MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)nokia.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h new file mode 100644 index 0000000..c391bae --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.h @@ -0,0 +1,267 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TLV320DAC33_H +#define __TLV320DAC33_H + +#define DAC33_PAGE_SELECT 0x00 +#define DAC33_PWR_CTRL 0x01 +#define DAC33_PLL_CTRL_A 0x02 +#define DAC33_PLL_CTRL_B 0x03 +#define DAC33_PLL_CTRL_C 0x04 +#define DAC33_PLL_CTRL_D 0x05 +#define DAC33_PLL_CTRL_E 0x06 +#define DAC33_INT_OSC_CTRL 0x07 +#define DAC33_INT_OSC_FREQ_RAT_A 0x08 +#define DAC33_INT_OSC_FREQ_RAT_B 0x09 +#define DAC33_INT_OSC_DAC_RATIO_SET 0x0A +#define DAC33_CALIB_TIME 0x0B +#define DAC33_INT_OSC_CTRL_B 0x0C +#define DAC33_INT_OSC_CTRL_C 0x0D +#define DAC33_INT_OSC_STATUS 0x0E +#define DAC33_INT_OSC_DAC_RATIO_READ 0x0F +#define DAC33_INT_OSC_FREQ_RAT_READ_A 0x10 +#define DAC33_INT_OSC_FREQ_RAT_READ_B 0x11 +#define DAC33_SER_AUDIOIF_CTRL_A 0x12 +#define DAC33_SER_AUDIOIF_CTRL_B 0x13 +#define DAC33_SER_AUDIOIF_CTRL_C 0x14 +#define DAC33_FIFO_CTRL_A 0x15 +#define DAC33_UTHR_MSB 0x16 +#define DAC33_UTHR_LSB 0x17 +#define DAC33_ATHR_MSB 0x18 +#define DAC33_ATHR_LSB 0x19 +#define DAC33_LTHR_MSB 0x1A +#define DAC33_LTHR_LSB 0x1B +#define DAC33_PREFILL_MSB 0x1C +#define DAC33_PREFILL_LSB 0x1D +#define DAC33_NSAMPLE_MSB 0x1E +#define DAC33_NSAMPLE_LSB 0x1F +#define DAC33_FIFO_WPTR_MSB 0x20 +#define DAC33_FIFO_WPTR_LSB 0x21 +#define DAC33_FIFO_RPTR_MSB 0x22 +#define DAC33_FIFO_RPTR_LSB 0x23 +#define DAC33_FIFO_DEPTH_MSB 0x24 +#define DAC33_FIFO_DEPTH_LSB 0x25 +#define DAC33_SAMPLES_REMAINING_MSB 0x26 +#define DAC33_SAMPLES_REMAINING_LSB 0x27 +#define DAC33_FIFO_IRQ_FLAG 0x28 +#define DAC33_FIFO_IRQ_MASK 0x29 +#define DAC33_FIFO_IRQ_MODE_A 0x2A +#define DAC33_FIFO_IRQ_MODE_B 0x2B +#define DAC33_DAC_CTRL_A 0x2C +#define DAC33_DAC_CTRL_B 0x2D +#define DAC33_DAC_CTRL_C 0x2E +#define DAC33_LDAC_DIG_VOL_CTRL 0x2F +#define DAC33_RDAC_DIG_VOL_CTRL 0x30 +#define DAC33_DAC_STATUS_FLAGS 0x31 +#define DAC33_ASRC_CTRL_A 0x32 +#define DAC33_ASRC_CTRL_B 0x33 +#define DAC33_SRC_REF_CLK_RATIO_A 0x34 +#define DAC33_SRC_REF_CLK_RATIO_B 0x35 +#define DAC33_SRC_EST_REF_CLK_RATIO_A 0x36 +#define DAC33_SRC_EST_REF_CLK_RATIO_B 0x37 +#define DAC33_INTP_CTRL_A 0x38 +#define DAC33_INTP_CTRL_B 0x39 +/* Registers 0x3A - 0x3F Reserved */ +#define DAC33_LDAC_PWR_CTRL 0x40 +#define DAC33_RDAC_PWR_CTRL 0x41 +#define DAC33_OUT_AMP_CM_CTRL 0x42 +#define DAC33_OUT_AMP_PWR_CTRL 0x43 +#define DAC33_OUT_AMP_CTRL 0x44 +#define DAC33_LINEL_TO_LLO_VOL 0x45 +/* Registers 0x45 - 0x47 Reserved */ +#define DAC33_LINER_TO_RLO_VOL 0x48 +#define DAC33_ANA_VOL_SOFT_STEP_CTRL 0x49 +#define DAC33_OSC_TRIM 0x4A +/* Registers 0x4B - 0x7C Reserved */ +#define DAC33_DEVICE_ID_MSB 0x7D +#define DAC33_DEVICE_ID_LSB 0x7E +#define DAC33_DEVICE_REV_ID 0x7F + +#define DAC33_CACHEREGNUM 128 + +/* Bit definitions */ + +/* DAC33_PWR_CTRL (0x01) */ +#define DACRPDNB (0x01 << 0) +#define DACLPDNB (0x01 << 1) +#define OSCPDNB (0x01 << 2) +#define PLLPDNB (0x01 << 3) +#define PDNALLB (0x01 << 4) +#define SOFT_RESET (0x01 << 7) + +/* DAC33_INT_OSC_CTRL (0x07) */ +#define REFSEL (0x01 << 1) + +/* DAC33_INT_OSC_CTRL_B (0x0C) */ +#define ADJSTEP(x) (x << 0) +#define ADJTHRSHLD(x) (x << 4) + +/* DAC33_INT_OSC_CTRL_C (0x0D) */ +#define REFDIV(x) (x << 4) + +/* DAC33_INT_OSC_STATUS (0x0E) */ +#define OSCSTATUS_IDLE_CALIB (0x00) +#define OSCSTATUS_NORMAL (0x01) +#define OSCSTATUS_ADJUSTMENT (0x03) +#define OSCSTATUS_NOT_USED (0x02) + +/* DAC33_SER_AUDIOIF_CTRL_A (0x12) */ +#define MSWCLK (0x01 << 0) +#define MSBCLK (0x01 << 1) +#define AFMT_MASK (0x03 << 2) +#define AFMT_I2S (0x00 << 2) +#define AFMT_DSP (0x01 << 2) +#define AFMT_RIGHT_J (0x02 << 2) +#define AFMT_LEFT_J (0x03 << 2) +#define WLEN_MASK (0x03 << 4) +#define WLEN_16 (0x00 << 4) +#define WLEN_20 (0x01 << 4) +#define WLEN_24 (0x02 << 4) +#define WLEN_32 (0x03 << 4) +#define NCYCL_MASK (0x03 << 6) +#define NCYCL_16 (0x00 << 6) +#define NCYCL_20 (0x01 << 6) +#define NCYCL_24 (0x02 << 6) +#define NCYCL_32 (0x03 << 6) + +/* DAC33_SER_AUDIOIF_CTRL_B (0x13) */ +#define DATA_DELAY_MASK (0x03 << 2) +#define DATA_DELAY(x) (x << 2) +#define BCLKON (0x01 << 5) + +/* DAC33_FIFO_CTRL_A (0x15) */ +#define WIDTH (0x01 << 0) +#define FBYPAS (0x01 << 1) +#define FAUTO (0x01 << 2) +#define FIFOFLUSH (0x01 << 3) + +/* + * UTHR, ATHR, LTHR, PREFILL, NSAMPLE (0x16 - 0x1F) + * 13-bit values +*/ +#define THRREG(x) (((x) & 0x1FFF) << 3) + +/* DAC33_FIFO_IRQ_MASK (0x29) */ +#define MNS (0x01 << 0) +#define MPS (0x01 << 1) +#define MAT (0x01 << 2) +#define MLT (0x01 << 3) +#define MUT (0x01 << 4) +#define MUF (0x01 << 5) +#define MOF (0x01 << 6) + +#define FIFO_IRQ_MODE_MASK (0x03) +#define FIFO_IRQ_MODE_RISING (0x00) +#define FIFO_IRQ_MODE_FALLING (0x01) +#define FIFO_IRQ_MODE_LEVEL (0x02) +#define FIFO_IRQ_MODE_EDGE (0x03) + +/* DAC33_FIFO_IRQ_MODE_A (0x2A) */ +#define UTM(x) (x << 0) +#define UFM(x) (x << 2) +#define OFM(x) (x << 4) + +/* DAC33_FIFO_IRQ_MODE_B (0x2B) */ +#define NSM(x) (x << 0) +#define PSM(x) (x << 2) +#define ATM(x) (x << 4) +#define LTM(x) (x << 4) + +/* DAC33_DAC_CTRL_A (0x2C) */ +#define DACRATE(x) (x << 0) +#define DACDUAL (0x01 << 4) +#define DACLKSEL_MASK (0x03 << 5) +#define DACLKSEL_INTSOC (0x00 << 5) +#define DACLKSEL_PLL (0x01 << 5) +#define DACLKSEL_MCLK (0x02 << 5) +#define DACLKSEL_BCLK (0x03 << 5) + +/* DAC33_DAC_CTRL_B (0x2D) */ +#define DACSRCR_MASK (0x03 << 0) +#define DACSRCR_MUTE (0x00 << 0) +#define DACSRCR_RIGHT (0x01 << 0) +#define DACSRCR_LEFT (0x02 << 0) +#define DACSRCR_MONOMIX (0x03 << 0) +#define DACSRCL_MASK (0x03 << 2) +#define DACSRCL_MUTE (0x00 << 2) +#define DACSRCL_LEFT (0x01 << 2) +#define DACSRCL_RIGHT (0x02 << 2) +#define DACSRCL_MONOMIX (0x03 << 2) +#define DVOLSTEP_MASK (0x03 << 4) +#define DVOLSTEP_SS_PERFS (0x00 << 4) +#define DVOLSTEP_SS_PER2FS (0x01 << 4) +#define DVOLSTEP_SS_DISABLED (0x02 << 4) +#define DVOLCTRL_MASK (0x03 << 6) +#define DVOLCTRL_LR_INDEPENDENT1 (0x00 << 6) +#define DVOLCTRL_LR_RIGHT_CONTROL (0x01 << 6) +#define DVOLCTRL_LR_LEFT_CONTROL (0x02 << 6) +#define DVOLCTRL_LR_INDEPENDENT2 (0x03 << 6) + +/* DAC33_DAC_CTRL_C (0x2E) */ +#define DEEMENR (0x01 << 0) +#define EFFENR (0x01 << 1) +#define DEEMENL (0x01 << 2) +#define EFFENL (0x01 << 3) +#define EN3D (0x01 << 4) +#define RESYNMUTE (0x01 << 5) +#define RESYNEN (0x01 << 6) + +/* DAC33_ASRC_CTRL_A (0x32) */ +#define SRCBYP (0x01 << 0) +#define SRCLKSEL_MASK (0x03 << 1) +#define SRCLKSEL_INTSOC (0x00 << 1) +#define SRCLKSEL_PLL (0x01 << 1) +#define SRCLKSEL_MCLK (0x02 << 1) +#define SRCLKSEL_BCLK (0x03 << 1) +#define SRCLKDIV(x) (x << 3) + +/* DAC33_ASRC_CTRL_B (0x33) */ +#define SRCSETUP(x) (x << 0) +#define SRCREFSEL (0x01 << 4) +#define SRCREFDIV(x) (x << 5) + +/* DAC33_INTP_CTRL_A (0x38) */ +#define INTPSEL (0x01 << 0) +#define INTPM_MASK (0x03 << 1) +#define INTPM_ALOW_OPENDRAIN (0x00 << 1) +#define INTPM_ALOW (0x01 << 1) +#define INTPM_AHIGH (0x02 << 1) + +/* DAC33_LDAC_PWR_CTRL (0x40) */ +/* DAC33_RDAC_PWR_CTRL (0x41) */ +#define DACLRNUM (0x01 << 2) +#define LROUT_GAIN(x) (x << 0) + +/* DAC33_ANA_VOL_SOFT_STEP_CTRL (0x49) */ +#define VOLCLKSEL (0x01 << 0) +#define VOLCLKEN (0x01 << 1) +#define VOLBYPASS (0x01 << 2) + +#define TLV320DAC33_MCLK 0 +#define TLV320DAC33_SLEEPCLK 1 + +extern struct snd_soc_dai dac33_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320dac33; + +#endif /* __TLV320DAC33_H */ -- 1.6.5.rc2

3 7

[alsa-devel] [PATCH] snd-usb-us122l: added support for US-144
by Tobias Hansen 13 Oct '09

13 Oct '09

Hi, here is a patch that adds support for the Tascam US-144 to snd-usb-us122l when uhci-hcd is used (ehci-hcd has to be disabled). With ehci-hcd the US-144 behaves completely differently. The difference between US-122L and US-144 on uhci-hcd is, that US-144 uses endpoints on two interfaces, while on US-122L all endpoints are on one interface. US-122L should still work, but I don't have a US-122L to test that. Audio in and out and Midi in were tested with US-144 and work. This is my first contribution to the linux kernel, so please have a critical look on it. ;) Taken against snd-usb-us122l 0.5 / alsa 1.0.20. Regards, Tobias Hansen

2 7

Re: [alsa-devel] [BUG] GPF in snd_hda_intel
by Takashi Iwai 13 Oct '09

13 Oct '09

At Mon, 12 Oct 2009 21:39:21 +0200, Luca Tettamanti wrote: > > On Mon, Oct 12, 2009 at 7:58 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Sat, 10 Oct 2009 15:01:10 +0200, > > Luca Tettamanti wrote: > >> > >> Hello, > >> with current git kernel (bd381934) I see a GPF when the snd_hda_intel driver is > >> loaded; the regression is recent, 0eca52a works fine. > >> The machine is an ASUS laptop (F3Sa), this is the sound card: > > > > Could you load the snd-hda-intel module with probe_only=1 option, > > and give the output of "alsa-info.sh --no-upload" ? > > > > If 0eca52a worked, there are only a few changes regarding hda-intel. > > > > f8f25ba3563dab14b1c3ea4d829642b8a61ca5d7 > > ALSA: hda - Add a workaround for ASUS A7K > > This is probably related; I haven't actually tested yet, but I have > one more information: I had "model=lenovo" in modprobe configuration. Ah, this is it. I can reproduce the problem with my emulator now. The patch below fixes the issue. > This was necessary because otherwise the driver applied the quirk for > Asus G1, resulting in no sound (see bug > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3513). > The driver works fine without "model" parameter (possibly due to the > removal of the quirk entry for the the G1); I get a warning about a > timeout when MSI is not enabled though: > > HDA Intel 0000:00:1b.0: PCI INT A disabled > HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > HDA Intel 0000:00:1b.0: setting latency timer to 64 > hda_codec: ALC660-VD: BIOS auto-probing. > input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input9 > hda-intel: azx_get_response timeout, switching to polling mode: last > cmd=0x018f0900 It's no critical error, but if enable_msi=1 helps, just keep the option. In 2.6.33, MSI will be enabled as default. thanks, Takashi ---

1 0

[PATCH] ALSA: hda - Fix overflow of spec->init_verbs in patch_realtek.c
by Takashi Iwai 13 Oct '09

13 Oct '09

ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to the recent changes. Simply increase the array size to avoid the overflow. Reported-by: Luca Tettamanti <kronos.it(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 470fd74..c08ca66 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -275,7 +275,7 @@ struct alc_spec { struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[5]; /* initialization verbs + const struct hda_verb *init_verbs[10]; /* initialization verbs * don't forget NULL * termination! */ -- 1.6.4.2

1 0

[alsa-devel] gta02: fix includes
by Ben Dooks 13 Oct '09

13 Oct '09

The gta02 build is missing <mach/regs-gpio.h> and is also using the incorrect <asm/io.h> header (it should be <linux/io.h>). Fix these two problems. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> Signed-off-by: Simtec Linux Team <linux(a)simtec.co.uk> --- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) Index: b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c =================================================================== --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c 2009-10-10 22:55:53.000000000 +0100 +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c 2009-10-10 22:58:31.000000000 +0100 @@ -19,6 +19,8 @@ #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/gpio.h> +#include <linux/io.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -29,7 +31,7 @@ #include <plat/regs-iis.h> #include <mach/regs-clock.h> -#include <asm/io.h> +#include <mach/regs-gpioj.h> #include <mach/gta02.h> #include "../codecs/wm8753.h" #include "s3c24xx-pcm.h" -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

Re: [alsa-devel] [PATCH] snd-usb-us122l: added support for US-144
by Tobias Hansen 12 Oct '09

12 Oct '09

At Sun, 27 Sep 2009 10:40:24 -0700, Eric Luiggi wrote: > > > Hi Tobias, > > I'd like to get my US-144 interface working on jaunty and am willing to > help out with testing for this US-144 solution. Please send patch > instructions. Hi, here are installation instructions: http://wiki.ubuntuusers.de/Benutzer/BigMc/Tascam_US-144 Regards, Tobias

1 0

[alsa-devel] [PATCH - try2] ASoC: TPA6130A2: Make tpa6130a2_power as static
by Peter Ujfalusi 12 Oct '09

12 Oct '09

The power for the amplifier should be handled internally by the tpa6130a2 driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/tpa6130a2.h | 1 - 2 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0a6e7b4..6b650c1 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -106,7 +106,7 @@ static void tpa6130a2_initialize(void) tpa6130a2_i2c_write(i, data->regs[i]); } -void tpa6130a2_power(int power) +static void tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h index 6a794f1..57e867f 100644 --- a/sound/soc/codecs/tpa6130a2.h +++ b/sound/soc/codecs/tpa6130a2.h @@ -57,6 +57,5 @@ #define TPA6130A2_VERSION_MASK (0x0f) extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); -extern void tpa6130a2_power(int power); #endif /* __TPA6130A2_H__ */ -- 1.6.5.rc2

2 1

[alsa-devel] [PATCH] ASoC: TPA6130A2: Make tpa6130a2_power as static
by Peter Ujfalusi 12 Oct '09

12 Oct '09

The power for the amplifier should be handled internally by the tpa6130a2 driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0a6e7b4..6b650c1 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -106,7 +106,7 @@ static void tpa6130a2_initialize(void) tpa6130a2_i2c_write(i, data->regs[i]); } -void tpa6130a2_power(int power) +static void tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; -- 1.6.5.rc2

1 1

[alsa-devel] [PATCH - try2] ASoC: TPA6130A2 amplifier driver
by Peter Ujfalusi 12 Oct '09

12 Oct '09

Driver for Texas Instruments TPA6130A2 stereo headphone amplifier. The driver provides playback gain control and also pre-defined DAPM_HP widgets and DAPM routings for power management. The DAPM_HP widget names are: "TPA6130A2 Headphone Left" "TPA6130A2 Headphone Right" _From soc machine drivers to use with the tpa6130a2 amplifier, the tpa6130a2_add_controls has to be called, which adds the alsa controls and the DAPM routing needed for the tpa6130a2. After that the machine driver can connect the codec's output with 'TPA6130A2 Left' and 'TPA6130A2 Right': {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"}, {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"}, Internally the left and right channels are powered separately. When none of the channels are needed the amplifier is powered down: hard power: valid GPIO number is passed within platform data soft power: Using the software shutdown of the amplifier Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The previous version of the tpa6130a2 driver was part of Eduardo's series, to add audio support for RX-51. We decided, that I'll send the tpa amplifier driver as separate patch to speed up the process. Comments from Mark has been addressed with this patch: set_power is replaced by GPIO number. DAPM routing added with per channel power management Only warning is printed, when using the driver on untested version of the chip include/sound/tpa6130a2-plat.h | 30 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tpa6130a2.c | 463 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tpa6130a2.h | 62 ++++++ 5 files changed, 561 insertions(+), 0 deletions(-) create mode 100644 include/sound/tpa6130a2-plat.h create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h new file mode 100644 index 0000000..e8c901e --- /dev/null +++ b/include/sound/tpa6130a2-plat.h @@ -0,0 +1,30 @@ +/* + * TPA6130A2 driver platform header + * + * Copyright (C) Nokia Corporation + * + * Written by Peter Ujfalusi <peter.ujfalusi(a)nokia.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#ifndef TPA6130A2_PLAT_H +#define TPA6130A2_PLAT_H + +struct tpa6130a2_platform_data { + int power_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3c46f34..fab01c9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TPA6130A2 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -228,3 +229,6 @@ config SND_SOC_WM9713 # Amp config SND_SOC_MAX9877 tristate + +config SND_SOC_TPA6130A2 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fc1c458..2f14391 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o +snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o @@ -101,3 +102,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o +obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c new file mode 100644 index 0000000..1b77c95 --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.c @@ -0,0 +1,463 @@ +/* + * ALSA SoC Texas Instruments TPA6130A2 headset stereo amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <sound/tpa6130a2-plat.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tpa6130a2.h" + +struct i2c_client *tpa6130a2_client; + +/* This struct is used to save the context */ +struct tpa6130a2_data { + struct mutex mutex; + unsigned char regs[TPA6130A2_CACHEREGNUM]; + int power_gpio; + unsigned char power_state; +}; + +static int tpa6130a2_i2c_read(int reg) +{ + struct tpa6130a2_data *data; + int val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + /* If powered off, return the cached value */ + if (data->power_state) { + val = i2c_smbus_read_byte_data(tpa6130a2_client, reg); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Read failed\n"); + else + data->regs[reg] = val; + } else { + val = data->regs[reg]; + } + + return val; +} + +static int tpa6130a2_i2c_write(int reg, u8 value) +{ + struct tpa6130a2_data *data; + int val = 0; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (data->power_state) { + val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Write failed\n"); + } + + /* Either powered on or off, we save the context */ + data->regs[reg] = value; + + return val; +} + +static u8 tpa6130a2_read(int reg) +{ + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + return data->regs[reg]; +} + +static void tpa6130a2_initialize(void) +{ + struct tpa6130a2_data *data; + int i; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + for (i = 1; i < TPA6130A2_REG_VERSION; i++) + tpa6130a2_i2c_write(i, data->regs[i]); +} + +void tpa6130a2_power(int power) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + if (power) { + /* Power on */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 1); + data->power_state = 1; + tpa6130a2_initialize(); + } + /* Clear SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } else { + /* set SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 0); + data->power_state = 0; + } + } + mutex_unlock(&data->mutex); +} + +static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + + ucontrol->value.integer.value[0] = + (tpa6130a2_read(reg) >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; + + mutex_unlock(&data->mutex); + return 0; +} + +static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val_reg; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (invert) + val = mask - val; + + mutex_lock(&data->mutex); + + val_reg = tpa6130a2_read(reg); + if (((val_reg >> shift) & mask) == val) { + mutex_unlock(&data->mutex); + return 0; + } + + val_reg &= ~(mask << shift); + val_reg |= val << shift; + tpa6130a2_i2c_write(reg, val_reg); + + mutex_unlock(&data->mutex); + + return 1; +} + +/* + * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going + * down in gain. + */ +static const unsigned int tpa6130_tlv[] = { + TLV_DB_RANGE_HEAD(10), + 0, 1, TLV_DB_SCALE_ITEM(-5950, 600, 0), + 2, 3, TLV_DB_SCALE_ITEM(-5000, 250, 0), + 4, 5, TLV_DB_SCALE_ITEM(-4550, 160, 0), + 6, 7, TLV_DB_SCALE_ITEM(-4140, 190, 0), + 8, 9, TLV_DB_SCALE_ITEM(-3650, 120, 0), + 10, 11, TLV_DB_SCALE_ITEM(-3330, 160, 0), + 12, 13, TLV_DB_SCALE_ITEM(-3040, 180, 0), + 14, 20, TLV_DB_SCALE_ITEM(-2710, 110, 0), + 21, 37, TLV_DB_SCALE_ITEM(-1960, 74, 0), + 38, 63, TLV_DB_SCALE_ITEM(-720, 45, 0), +}; + +static const struct snd_kcontrol_new tpa6130a2_controls[] = { + SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, + tpa6130a2_get_reg, tpa6130a2_set_reg, + tpa6130_tlv), +}; + +/* + * Enable or disable channel (left or right) + * The bit number for mute and amplifier are the same per channel: + * bit 6: Right channel + * bit 7: Left channel + * in both registers. + */ +static void tpa6130a2_channel_enable(u8 channel, int enable) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (enable) { + /* Enable channel */ + /* Enable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + + /* Unmute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + } else { + /* Disable channel */ + /* Mute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + + /* Disable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } +} + +static int tpa6130a2_left_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 0); + break; + } + return 0; +} + +static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 0); + break; + } + return 0; +} + +static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_power(1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_power(0); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { + SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_left_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_right_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM, + 0, 0, tpa6130a2_supply_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + /* Outputs */ + SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), + SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"}, + + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"}, +}; + +int tpa6130a2_add_controls(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, + ARRAY_SIZE(tpa6130a2_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); + +} +EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); + +static int tpa6130a2_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tpa6130a2_data *data; + struct tpa6130a2_platform_data *pdata; + int ret; + + dev = &client->dev; + + if (client->dev.platform_data == NULL) { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (data == NULL) { + dev_err(dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + tpa6130a2_client = client; + + i2c_set_clientdata(tpa6130a2_client, data); + + pdata = (struct tpa6130a2_platform_data *)client->dev.platform_data; + data->power_gpio = pdata->power_gpio; + + mutex_init(&data->mutex); + + /* Set default register values */ + data->regs[TPA6130A2_REG_CONTROL] = TPA6130A2_SWS; + data->regs[TPA6130A2_REG_VOL_MUTE] = TPA6130A2_MUTE_R | + TPA6130A2_MUTE_L; + + if (data->power_gpio >= 0) { + ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + if (ret < 0) { + dev_err(dev, "Failed to request power GPIO (%d)\n", + data->power_gpio); + goto fail; + } + gpio_direction_output(data->power_gpio, 0); + } else { + data->power_state = 1; + tpa6130a2_initialize(); + } + + tpa6130a2_power(1); + + /* Read version */ + ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & + TPA6130A2_VERSION_MASK; + if ((ret != 1) && (ret != 2)) + dev_warn(dev, "UNTESTED version detected (%d)\n", ret); + + /* Disable the chip */ + tpa6130a2_power(0); + + return 0; +fail: + kfree(data); + i2c_set_clientdata(tpa6130a2_client, NULL); + tpa6130a2_client = 0; + + return ret; +} + +static int tpa6130a2_remove(struct i2c_client *client) +{ + struct tpa6130a2_data *data = i2c_get_clientdata(client); + + tpa6130a2_power(0); + + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); + kfree(data); + tpa6130a2_client = 0; + + return 0; +} + +static const struct i2c_device_id tpa6130a2_id[] = { + { "tpa6130a2", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); + +static struct i2c_driver tpa6130a2_i2c_driver = { + .driver = { + .name = "tpa6130a2", + .owner = THIS_MODULE, + }, + .probe = tpa6130a2_probe, + .remove = __devexit_p(tpa6130a2_remove), + .id_table = tpa6130a2_id, +}; + +static int __init tpa6130a2_init(void) +{ + return i2c_add_driver(&tpa6130a2_i2c_driver); +} + +static void __exit tpa6130a2_exit(void) +{ + i2c_del_driver(&tpa6130a2_i2c_driver); +} + +MODULE_AUTHOR("Peter Ujfalusi"); +MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); +MODULE_LICENSE("GPL"); + +module_init(tpa6130a2_init); +module_exit(tpa6130a2_exit); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h new file mode 100644 index 0000000..6a794f1 --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.h @@ -0,0 +1,62 @@ +/* + * ALSA SoC TPA6130A2 amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TPA6130A2_H__ +#define __TPA6130A2_H__ + +/* Register addresses */ +#define TPA6130A2_REG_CONTROL 0x01 +#define TPA6130A2_REG_VOL_MUTE 0x02 +#define TPA6130A2_REG_OUT_IMPEDANCE 0x03 +#define TPA6130A2_REG_VERSION 0x04 + +#define TPA6130A2_CACHEREGNUM (TPA6130A2_REG_VERSION + 1) + +/* Register bits */ +/* TPA6130A2_REG_CONTROL (0x01) */ +#define TPA6130A2_SWS (0x01 << 0) +#define TPA6130A2_TERMAL (0x01 << 1) +#define TPA6130A2_MODE(x) (x << 4) +#define TPA6130A2_MODE_STEREO (0x00) +#define TPA6130A2_MODE_DUAL_MONO (0x01) +#define TPA6130A2_MODE_BRIDGE (0x02) +#define TPA6130A2_MODE_MASK (0x03) +#define TPA6130A2_HP_EN_R (0x01 << 6) +#define TPA6130A2_HP_EN_L (0x01 << 7) + +/* TPA6130A2_REG_VOL_MUTE (0x02) */ +#define TPA6130A2_VOLUME(x) ((x & 0x3f) << 0) +#define TPA6130A2_MUTE_R (0x01 << 6) +#define TPA6130A2_MUTE_L (0x01 << 7) + +/* TPA6130A2_REG_OUT_IMPEDANCE (0x03) */ +#define TPA6130A2_HIZ_R (0x01 << 0) +#define TPA6130A2_HIZ_L (0x01 << 1) + +/* TPA6130A2_REG_VERSION (0x04) */ +#define TPA6130A2_VERSION_MASK (0x0f) + +extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); +extern void tpa6130a2_power(int power); + +#endif /* __TPA6130A2_H__ */ -- 1.6.5.rc2

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[alsa-devel] [PATCH] [snd-usb-audio] Add match for JMTek USB sound card
by Lubomir Rintel 12 Oct '09

12 Oct '09

Entry for JMTek LLC., SSS USB Headphone Set in the quirk table. --- sound/usb/usbquirks.h | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201e..e98ff4c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -80,6 +80,19 @@ .bInterfaceClass = USB_CLASS_AUDIO, }, +/* JMTek, LLC. SSS USB Headphone Set */ +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_DEV_CLASS | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x0c76, + .idProduct = 0x1605, + .bDeviceClass = USB_CLASS_PER_INTERFACE, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, + /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. -- 1.6.5.rc2

3 3

Re: [alsa-devel] [PATCH] Add match for JMTek USB sound card
by Takashi Iwai 12 Oct '09

12 Oct '09

At Mon, 12 Oct 2009 08:42:40 +0200, Lubomir Rintel wrote: > > Entry for JMTek LLC., SSS USB Headphone Set in the quirk table. > > Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> Thanks, applied now. Takashi > --- > sound/usb/usbquirks.h | 13 +++++++++++++ > 1 files changed, 13 insertions(+), 0 deletions(-) > > diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h > index f6f201e..e98ff4c 100644 > --- a/sound/usb/usbquirks.h > +++ b/sound/usb/usbquirks.h > @@ -80,6 +80,19 @@ > .bInterfaceClass = USB_CLASS_AUDIO, > }, > > +/* JMTek, LLC. SSS USB Headphone Set */ > +{ > + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | > + USB_DEVICE_ID_MATCH_DEV_CLASS | > + USB_DEVICE_ID_MATCH_INT_CLASS | > + USB_DEVICE_ID_MATCH_INT_SUBCLASS, > + .idVendor = 0x0c76, > + .idProduct = 0x1605, > + .bDeviceClass = USB_CLASS_PER_INTERFACE, > + .bInterfaceClass = USB_CLASS_AUDIO, > + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL > +}, > + > /* > * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface > * class matches do not take effect without an explicit ID match. > -- > 1.6.2.5 >

1 0

[alsa-devel] Bt87x unknown card
by Adriano 12 Oct '09

12 Oct '09

Hi I've this Pinnacle PCTV card: 11bd:0012 with this multimedia audio controller: 02:02.1 Multimedia controller [0480]: Brooktree Corporation Bt878 Audio Capture [109e:0878] (rev 11) Subsystem: Pinnacle Systems Inc. PCTV pro (TV + FM stereo receiver, audio section) [11bd:0012] Flags: bus master, medium devsel, latency 32, IRQ 23 Memory at f7eff000 (32-bit, prefetchable) [size=4K] Capabilities: <access denied> Kernel driver in use: Bt87x I have no audio with alsa, when I loaded snd-bt87x, I received this error in dmesg: [ 1951.383480] unknown card 0x878-0x11bd:0x12 [ 1951.383484] please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> [ 1951.383518] Bt87x 0000:02:02.1: PCI INT A -> GSI 23 (level, low) -> IRQ 23 [ 1951.383741] bt87x0: Using board 0, analog, digital (rate 32000 Hz) Can you tell me the problem and any solution? Thanks you Adriano Foschi

2 1

[alsa-devel] Audo capture not working with HP 9000 laptop and 2.6.32 kernel
by Corey Boyle 12 Oct '09

12 Oct '09

Hi I have an HP 9000 series laptop with an Intel HD sound card and Realtek ALC298 codec. I am having a problem with audio capture ever since upgrading to the 2.6.32 kernel series from 2.6.31. I have git bisected to this commit: 33d78674586aeb6a623b1e612e6f92dd83015ed3 "ALSA: hda - Use auto model for HP laptops with ALC268 codec". After this commit I am not able to capture audio and using alsamixer I am not able to choose the input source. I was using "Front Mic" as the input source when this was working in 2.6.31. I have uploaded my configuration using alsa-info before and after the commit. Before change http://www.alsa-project.org/db/?f=4f5d3eb02d4d62fa044a4eb0a3aae2614064b208 After change http://www.alsa-project.org/db/?f=a9583da102419a9233cc469259c85ab81645d555 Could someone please provide some help in getting this working, presumably without rolling back the previously mentioned commit? I am not familiar with how the alsa code works but am willing to help out if given some guidance. Please CC me on all replies as I am not a subscriber to the list. Thanks Corey

2 1

[alsa-devel] [BUG] GPF in snd_hda_intel
by Luca Tettamanti 12 Oct '09

12 Oct '09

Hello, with current git kernel (bd381934) I see a GPF when the snd_hda_intel driver is loaded; the regression is recent, 0eca52a works fine. The machine is an ASUS laptop (F3Sa), this is the sound card: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) The codec is ALC660-VD (snd_hda_codec_realtek). The is the GFP message: general protection fault: 0000 [#1] PREEMPT SMP last sysfs file: /sys/devices/pci0000:00/0000:00:1c.1/0000:03:00.0/net/wlan0/ CPU 0 Modules linked in: arc4 snd_hda_codec_realtek ecb iwlagn snd_hda_intel(+) usbhid hid btusb snd_hda_codec iwlcore snd_hwdep bluetooth snd_pcm rtc_cmos rtc_core asus_laptop snd_seq snd_timer snd_seq_device snd psmouse mac80211 rtc_lib video soundcore cfg80211 evdev output snd_page_alloc rfkill battery button pcspkr ac processor ext4 mbcache jbd2 crc16 dm_mod sg sr_mod cdrom sd_mod ahci uhci_hcd ata_piix ohci1394 sdhci_pci sdhci mmc_core led_class ieee1394 libata ehci_hcd scsi_mod usbcore intel_agp thermal fan unix [last unloaded: scsi_wait_scan] Pid: 1027, comm: modprobe Not tainted 2.6.32-rc3-00337-gbd38193 #160 F3Sa RIP: 0010:[<ffffffffa033aebd>] [<ffffffffa033aebd>] snd_hda_sequence_write+0x27/0x34 [snd_hda_codec] RSP: 0018:ffff88013e0afbe8 EFLAGS: 00010207 RAX: 0000000000000000 RBX: 36434c4100000006 RCX: 000000000000009a RDX: ffff88013e0afb28 RSI: 36434c4100000006 RDI: ffff88013e259000 RBP: ffff88013e0afbf8 R08: 000000000000005e R09: ffff88013e0af9d8 R10: 0000000000000000 R11: 000000000000d2d0 R12: ffff88013e259000 R13: ffff88013e259000 R14: ffff88013d12c800 R15: ffff88013e0afd08 FS: 00007f423f9086f0(0000) GS:ffff880028200000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 000000000061d468 CR3: 000000013e568000 CR4: 00000000000006f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process modprobe (pid: 1027, threadinfo ffff88013e0ae000, task ffff88013ddbc290) Stack: 0000000000000006 ffff88013d12c830 ffff88013e0afc28 ffffffffa03bc117 <0> ffff88013e259000 0000000000000002 ffff88013dd68258 ffff88013de4f600 <0> ffff88013e0afc48 ffffffffa033aef5 ffff88013dd68258 ffff88013e259000 Call Trace: [<ffffffffa03bc117>] alc_init+0x2f8/0x31e [snd_hda_codec_realtek] [<ffffffffa033aef5>] snd_hda_codec_build_controls+0x2b/0x4c [snd_hda_codec] [<ffffffffa033c8ca>] snd_hda_build_controls+0x23/0x7f [snd_hda_codec] [<ffffffffa03899a0>] azx_probe+0x9d9/0xa97 [snd_hda_intel] [<ffffffffa0388e10>] ? azx_send_cmd+0x0/0x174 [snd_hda_intel] [<ffffffffa0388c19>] ? azx_get_response+0x0/0x1f7 [snd_hda_intel] [<ffffffffa0388276>] ? azx_attach_pcm_stream+0x0/0x209 [snd_hda_intel] [<ffffffffa0387ec4>] ? azx_bus_reset+0x0/0x79 [snd_hda_intel] [<ffffffffa0387d28>] ? azx_power_notify+0x0/0x5d [snd_hda_intel] [<ffffffff811829f3>] local_pci_probe+0x12/0x16 [<ffffffff811836b9>] pci_device_probe+0x5f/0x89 [<ffffffff81207f68>] ? driver_sysfs_add+0x4c/0x72 [<ffffffff812080b4>] driver_probe_device+0xad/0x15f [<ffffffff812081be>] __driver_attach+0x58/0x7b [<ffffffff81208166>] ? __driver_attach+0x0/0x7b [<ffffffff81207893>] bus_for_each_dev+0x4e/0x84 [<ffffffff81207f1a>] driver_attach+0x1c/0x1e [<ffffffff81207205>] bus_add_driver+0x126/0x26f [<ffffffff812084b2>] driver_register+0xb3/0x121 [<ffffffffa038f000>] ? alsa_card_azx_init+0x0/0x20 [snd_hda_intel] [<ffffffff811838fe>] __pci_register_driver+0x51/0xbc [<ffffffffa038f000>] ? alsa_card_azx_init+0x0/0x20 [snd_hda_intel] [<ffffffffa038f01e>] alsa_card_azx_init+0x1e/0x20 [snd_hda_intel] [<ffffffff81009060>] do_one_initcall+0x5a/0x14a [<ffffffff8107270a>] sys_init_module+0xd0/0x229 [<ffffffff8100baeb>] system_call_fastpath+0x16/0x1b Code: 31 c0 c9 c3 55 48 89 e5 41 54 49 89 fc 53 48 89 f3 eb 18 8b 4b 04 44 8b 43 08 0f b7 f0 31 d2 4c 89 e7 48 83 c3 0c e8 ce f4 ff ff <66> 8b 03 66 85 c0 75 e0 5b 41 5c c9 c3 55 48 89 e5 53 48 89 fb RIP [<ffffffffa033aebd>] snd_hda_sequence_write+0x27/0x34 [snd_hda_codec] RSP <ffff88013e0afbe8> ---[ end trace 91ec1e9b500145fa ]--- The disassembly: 1e96: 55 push %rbp 1e97: 48 89 e5 mov %rsp,%rbp 1e9a: 41 54 push %r12 1e9c: 49 89 fc mov %rdi,%r12 1e9f: 53 push %rbx 1ea0: 48 89 f3 mov %rsi,%rbx 1ea3: eb 18 jmp 1ebd <snd_hda_sequence_write+0x27> 1ea5: 8b 4b 04 mov 0x4(%rbx),%ecx 1ea8: 44 8b 43 08 mov 0x8(%rbx),%r8d 1eac: 0f b7 f0 movzwl %ax,%esi 1eaf: 31 d2 xor %edx,%edx 1eb1: 4c 89 e7 mov %r12,%rdi 1eb4: 48 83 c3 0c add $0xc,%rbx 1eb8: e8 00 00 00 00 callq 1ebd <snd_hda_sequence_write+0x27> 1ebd: 66 8b 03 mov (%rbx),%ax <-- fault here 1ec0: 66 85 c0 test %ax,%ax 1ec3: 75 e0 jne 1ea5 <snd_hda_sequence_write+0xf> 1ec5: 5b pop %rbx 1ec6: 41 5c pop %r12 1ec8: c9 leaveq 1ec9: c3 retq Luca

2 1

[alsa-devel] [PATCH] snd_dma_pointer workaround for chipsets with buggy DMA
by Krzysztof Helt 11 Oct '09

11 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The chipsets with the isa_dma_bridge_buggy set do not stop DMA during DMA counter reads. The DMA counter is read in two 8-bit read steps on x86 platform. Sometimes, such reads happen during higher byte change so the lower byte is already decremented (rolled over) but the higher byte is not. It introduces an error that position is moved 256 bytes ahead of the true position. Thus, the next DMA position read can return a lower value then the previous read. If the DMA position is decreased (reversed) the ALSA subsystem is tricked into the playback underrun error and resets the playback. It results in a "pop" during a playback. Work around the issue by reading the counter twice and choosing a higher value. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This issue can be tested with the pcm tool from the alsa-lib's test package. The command "pcm -c 2 -b 350000 -p 5800" can easily trigger the original error every ten seconds on my system. This work around greatly reduces the underrun errors for ogg123 and mpg321 on VIA Apollo (586) chipset + K6-2 533 MHz (from every few seconds to one per about 15 minutes - few songs). sound/core/isadma.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 79f0f16..950e19b 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -85,16 +85,24 @@ EXPORT_SYMBOL(snd_dma_disable); unsigned int snd_dma_pointer(unsigned long dma, unsigned int size) { unsigned long flags; - unsigned int result; + unsigned int result, result1; flags = claim_dma_lock(); clear_dma_ff(dma); if (!isa_dma_bridge_buggy) disable_dma(dma); result = get_dma_residue(dma); + /* + * HACK - read the counter again and choose higher value in order to + * avoid reading during counter lower byte roll over if the + * isa_dma_bridge_buggy is set. + */ + result1 = get_dma_residue(dma); if (!isa_dma_bridge_buggy) enable_dma(dma); release_dma_lock(flags); + if (unlikely(result < result1)) + result = result1; #ifdef CONFIG_SND_DEBUG if (result > size) snd_printk(KERN_ERR "pointer (0x%x) for DMA #%ld is greater than transfer size (0x%x)\n", result, dma, size); -- 1.6.0.3 --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

2 1

[alsa-devel] [PATCH] wss: reuse CS4231 controls for AD1848
by Krzysztof Helt 11 Oct '09

11 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The C4231 control set is a superset of the AD1848 control set so reuse the CS4231 controls definitions for the AD1848. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/wss/wss_lib.c | 79 +++++++++++++--------------------------------- 1 files changed, 23 insertions(+), 56 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 754a208..2ba1897 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2200,49 +2200,12 @@ static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); -static struct snd_kcontrol_new snd_ad1848_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, - 7, 7, 1, 1), -WSS_DOUBLE_TLV("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), -WSS_DOUBLE("Aux Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, - 0, 0, 15, 0, db_scale_rec_gain), -{ - .name = "Capture Source", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -}, -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, - db_scale_6bit), -}; - static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, db_scale_6bit), -WSS_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, - db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Aux Playback Volume", 0, @@ -2253,15 +2216,6 @@ WSS_DOUBLE("Aux Playback Switch", 1, WSS_DOUBLE_TLV("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, - CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1, - db_scale_4bit), -WSS_SINGLE("Mono Output Playback Switch", 0, - CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, - CS4231_MONO_CTRL, 5, 1, 0), WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0, db_scale_rec_gain), { @@ -2277,6 +2231,20 @@ WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_SINGLE("Mono Playback Switch", 0, + CS4231_MONO_CTRL, 7, 1, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), +WSS_SINGLE("Mono Output Playback Switch", 0, + CS4231_MONO_CTRL, 6, 1, 1), +WSS_SINGLE("Mono Output Playback Bypass", 0, + CS4231_MONO_CTRL, 5, 1, 0), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { @@ -2343,22 +2311,21 @@ int snd_wss_mixer(struct snd_wss *chip) if (err < 0) return err; } - else if (chip->hardware & WSS_HW_AD1848_MASK) - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_ad1848_controls[idx], - chip)); - if (err < 0) - return err; - } - else - for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) { + else { + int count = ARRAY_SIZE(snd_wss_controls); + + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + + for (idx = 0; idx < count; idx++) { err = snd_ctl_add(card, snd_ctl_new1(&snd_wss_controls[idx], chip)); if (err < 0) return err; } + } return 0; } EXPORT_SYMBOL(snd_wss_mixer); -- 1.6.0.3 ---------------------------------------------------------------------- Szukasz pracy? Chcesz lepiej zarabia�? Sprawd� oferty na http://link.interia.pl/f23ba

2 1

[alsa-devel] PATCH: ESI Juli@ fails to set sample rate
by David Henningsson 11 Oct '09

11 Oct '09

This one-line patch fixes bug https://bugs.launchpad.net/ubuntu/+source/linux/+bug/441446 and is likely to fix bug https://bugzilla.redhat.com/show_bug.cgi?id=525595 as well. The problem for me is that if two streams are started immediately after one another (such as a playback and a recording stream, e g pulseaudio does that on startup), the call to hw_params fail with EBUSY. This patch makes the call succeed, and playback and recording works. I do not have any other ice1724 soundcard than the ESI Juli@ so I don't know if/how this patch will affect them. // David

2 3

[alsa-devel] [PATCH 1/1] ALSA: ice1724 - Make call to set hw params succeed on ESI Juli@
by David Henningsson 11 Oct '09

11 Oct '09

From: David Henningsson <launchpad.web(a)epost.diwic.se> If two streams are started immediately after one another (such as a playback and a recording stream), the call to set hw params fails with EBUSY. This patch makes the call succeed, so playback and recording will work properly. Signed-off-by: David Henningsson <launchpad.web(a)epost.diwic.se> --- sound/pci/ice1712/ice1724.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) * This is likely a regression since 2009-06-05, commit 92d71005e2f305d6dca126d8b8497e885b842dba. * Tested on ESI Juli@. * Affected bugs: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/441446 Probably this one as well: https://bugzilla.redhat.com/show_bug.cgi?id=525595 --- diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index cc84a83..56d1bd8 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -643,7 +643,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); - return -EBUSY; + return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY; } if (!force && is_pro_rate_locked(ice)) { spin_unlock_irqrestore(&ice->reg_lock, flags);

1 0

[alsa-devel] [PATCH 1/2] ASoC: AD1980/AD73311: make sure to set dev member
by Mike Frysinger 10 Oct '09

10 Oct '09

The codec driver should initialize snd_soc_codec's dev member to the appropriate device when setting up the device, but these codecs weren't. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/ad73311.c | 1 + 2 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index d7440a9..0477a91 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -203,6 +203,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ad1980_reg); codec->reg_cache_step = 2; codec->name = "AD1980"; + codec->dev = &pdev->dev; codec->owner = THIS_MODULE; codec->dai = &ad1980_dai; codec->num_dai = 1; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index e61dac5..6e18149 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -50,6 +50,7 @@ static int ad73311_soc_probe(struct platform_device *pdev) return -ENOMEM; mutex_init(&codec->mutex); codec->name = "AD73311"; + codec->dev = &pdev->dev; codec->owner = THIS_MODULE; codec->dai = &ad73311_dai; codec->num_dai = 1; -- 1.6.5.rc2

4 19

[alsa-devel] [PATCH 30/30] ALSA: HDA VIA: Only cosmetic changes
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Only cosmetic changes Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 64 +++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 31 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -22,26 +22,26 @@ */ /* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */ -/* */ +/* */ /* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */ -/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ -/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ +/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ +/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ -/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ -/* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ /* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ -/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ -/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ -/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ -/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ /* 2009-02-16 Logan Li Add support for VT1718S */ /* 2009-03-13 Logan Li Add support for VT1716S */ /* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ /* 2009-07-08 Lydia Wang Add support for VT2002P */ /* 2009-07-21 Lydia Wang Add support for VT1812 */ -/* */ +/* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -486,7 +486,7 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -1545,7 +1545,7 @@ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* Setup default input MW0 to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ @@ -1887,8 +1887,10 @@ codec->pcm_info = info; info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -2133,7 +2135,7 @@ } else /* enable SPDIF-input pin */ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - return 0; + return 0; } #ifdef SND_HDA_NEEDS_RESUME @@ -2178,8 +2180,8 @@ spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -2217,7 +2219,7 @@ if (!nid) continue; - + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { @@ -2246,7 +2248,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2260,7 +2262,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2360,7 +2362,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -2593,7 +2595,7 @@ "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -2605,7 +2607,7 @@ spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -2792,11 +2794,11 @@ spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ - switch(i) { + switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ spec->multiout.dac_nids[i] = 0x10; @@ -2831,7 +2833,7 @@ for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; nid_vol = nid_vols[i]; @@ -2862,7 +2864,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2876,7 +2878,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2972,7 +2974,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -3081,7 +3083,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -3175,7 +3177,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);

1 0

[alsa-devel] [PATCH 29/30] ALSA: HDA VIA: comments: update copyright, changeset, etc.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: comments: update copyright, changeset, etc. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1,10 +1,10 @@ /* * Universal Interface for Intel High Definition Audio Codec * - * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec + * HD audio interface patch for VIA VT17xx/VT18xx/VT20xx codec * - * Copyright (c) 2006-2008 Lydia Wang <lydiawang(a)viatech.com> - * Takashi Iwai <tiwai(a)suse.de> + * (C) 2006-2009 VIA Technology, Inc. + * (C) 2006-2008 Takashi Iwai <tiwai(a)suse.de> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -36,6 +36,11 @@ /* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ /* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2009-02-16 Logan Li Add support for VT1718S */ +/* 2009-03-13 Logan Li Add support for VT1716S */ +/* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ +/* 2009-07-08 Lydia Wang Add support for VT2002P */ +/* 2009-07-21 Lydia Wang Add support for VT1812 */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */

1 0

[alsa-devel] [PATCH 28/30] ALSA: HDA VIA: Change PW4 connect select default to to MW0.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Change PW4 connect select default to to MW0. According to customer request, hp should be default to redirected mode, i.e. PW4 connect select default to to MW0. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1541,8 +1541,8 @@ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Setup default input to PW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Setup default input MW0 to PW4 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -2685,8 +2685,8 @@ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set input of PW4 as AOW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Set input of PW4 as MW0 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -3239,7 +3239,7 @@ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Setup default input to PW4 */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* PW10 Input enable */

1 0

[alsa-devel] [PATCH 27/30] ALSA: HDA VIA: rename vt1708_control_templates[].
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: rename vt1708_control_templates[]. To via_control_templates[]. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -367,7 +367,7 @@ .put = bind_pin_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -static struct snd_kcontrol_new vt1708_control_templates[] = { +static struct snd_kcontrol_new via_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, @@ -430,7 +430,7 @@ knew = snd_array_new(&spec->kctls); if (!knew) return -ENOMEM; - *knew = vt1708_control_templates[type]; + *knew = via_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM;

1 0

[alsa-devel] [PATCH 26/30] ALSA: HDA VIA: Add VT1812 support.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1812 support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 494 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 491 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -89,6 +89,7 @@ VT1718S, VT1716S, VT2002P, + VT1812, CODEC_TYPES, }; @@ -187,6 +188,8 @@ codec_type = VT1718S; else if (dev_id == 0x0438 || dev_id == 0x4438) codec_type = VT2002P; + else if (dev_id == 0x0448) + codec_type = VT1812; else codec_type = UNKNOWN; return codec_type; @@ -411,6 +414,12 @@ 0x10, 0x11 }; +static hda_nid_t vt1812_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -904,6 +913,120 @@ snd_hda_codec_write( codec, 0x21, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else if (spec->codec_type == VT1812) { + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_codec_read( + codec, 0x28, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -974,6 +1097,9 @@ case VT2002P: nid = 0x35; break; + case VT1812: + nid = 0x3d; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -1049,6 +1175,9 @@ case VT2002P: nid = 0x35; break; + case VT1812: + nid = 0x3d; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -1066,7 +1195,8 @@ || spec->codec_type == VT1702 || spec->codec_type == VT1718S || spec->codec_type == VT1716S - || spec->codec_type == VT2002P) { + || spec->codec_type == VT2002P + || spec->codec_type == VT1812) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -1307,6 +1437,7 @@ end_idx = 3; break; case VT2002P: + case VT1812: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -1370,6 +1501,7 @@ parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; case VT2002P: + case VT1812: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -1900,7 +2032,7 @@ unsigned int hp_present; struct via_spec *spec = codec->spec; - if (spec->codec_type != VT2002P) + if (spec->codec_type != VT2002P && spec->codec_type != VT1812) return; hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, @@ -2381,7 +2513,7 @@ via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P) { + if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5671,6 +5803,361 @@ return 0; } + +/* for vt1812 */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1812_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1812_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/13/15 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0xa8}, + { } +}; + + +static struct hda_verb vt1812_uniwill_init_verbs[] = { + {0x33, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT }, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1812_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1812_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + + +/* add playback controls from the parsed DAC table */ +static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 6; + imux->num_items++; + + return 0; +} + +static int vt1812_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + fill_dig_outs(codec); + err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs) + return 0; /* can't find valid BIOS pin config */ + + err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1812_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt1812 */ +static int patch_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1812_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + + spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs; + + spec->stream_name_analog = "VT1812 Analog"; + spec->stream_analog_playback = &vt1812_pcm_analog_playback; + spec->stream_analog_capture = &vt1812_pcm_analog_capture; + + spec->stream_name_digital = "VT1812 Digital"; + spec->stream_digital_playback = &vt1812_pcm_digital_playback; + + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1812_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1812_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1812_loopbacks; +#endif + + return 0; +} + /* * patch entries */ @@ -5757,6 +6244,7 @@ .patch = patch_vt1716S}, { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 25/30] ALSA: HDA VIA: Add VT2002P support.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT2002P support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 665 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 660 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -88,6 +88,7 @@ VT1702, VT1718S, VT1716S, + VT2002P, CODEC_TYPES, }; @@ -184,6 +185,8 @@ codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; + else if (dev_id == 0x0438 || dev_id == 0x4438) + codec_type = VT2002P; else codec_type = UNKNOWN; return codec_type; @@ -193,11 +196,14 @@ #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 #define VIA_MONO_EVENT 0x08 +#define VIA_SPEAKER_EVENT 0x10 +#define VIA_BIND_HP_EVENT 0x20 enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, VIA_CTL_WIDGET_ANALOG_MUTE, + VIA_CTL_WIDGET_BIND_PIN_MUTE, }; enum { @@ -235,6 +241,7 @@ flush_scheduled_work(); } + static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -262,13 +269,108 @@ .put = analog_input_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } +static void via_hp_bind_automute(struct hda_codec *codec); + +static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int i; + int change = 0; + + long *valp = ucontrol->value.integer.value; + int lmute, rmute; + if (strstr(kcontrol->id.name, "Switch") == NULL) { + snd_printd("Invalid control!\n"); + return change; + } + change = snd_hda_mixer_amp_switch_put(kcontrol, + ucontrol); + /* Get mute value */ + lmute = *valp ? 0 : HDA_AMP_MUTE; + valp++; + rmute = *valp ? 0 : HDA_AMP_MUTE; + + /* Set hp pins */ + if (!spec->hp_independent_mode) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + + if (!lmute && !rmute) { + /* Line Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* unmute */ + via_hp_bind_automute(codec); + + } else { + if (lmute) { + /* Mute all left channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + } + if (rmute) { + /* mute all right channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + return change; +} + +#define BIND_PIN_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = bind_pin_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, + BIND_PIN_MUTE, }; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -304,6 +406,11 @@ 0x13, 0x14 }; +static hda_nid_t vt2002P_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -386,10 +493,13 @@ { struct via_spec *spec = codec->spec; hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + } } static void via_auto_init_analog_input(struct hda_codec *codec) @@ -693,6 +803,107 @@ imux_is_smixer ? AC_PWRST_D0 : parm); snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, mono_out ? AC_PWRST_D0 : parm); + } else if (spec->codec_type == VT2002P) { + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, + 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Class-D */ + /* PW0 (24h), MW0(18h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* Mono Out */ + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + present = snd_hda_codec_read( + codec, 0x26, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -760,6 +971,9 @@ case VT1718S: nid = 0x34; break; + case VT2002P: + nid = 0x35; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -832,6 +1046,9 @@ pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ spec->multiout.num_dacs = 4; break; + case VT2002P: + nid = 0x35; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -848,7 +1065,8 @@ if (spec->codec_type == VT1708S || spec->codec_type == VT1702 || spec->codec_type == VT1718S - || spec->codec_type == VT1716S) { + || spec->codec_type == VT1716S + || spec->codec_type == VT2002P) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -1088,6 +1306,11 @@ start_idx = 1; end_idx = 3; break; + case VT2002P: + nid_mixer = 0x21; + start_idx = 0; + end_idx = 2; + break; default: return 0; } @@ -1146,6 +1369,10 @@ verb = 0xf73; parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; + case VT2002P: + verb = 0xf93; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ + break; default: return; /* other codecs are not supported */ } @@ -1667,6 +1894,66 @@ } } +/* mute Internal-Speaker if HP is plugged */ +static void via_speaker_automute(struct hda_codec *codec) +{ + unsigned int hp_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT2002P) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Speaker Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +/* mute line-out and internal speaker if HP is plugged */ +static void via_hp_bind_automute(struct hda_codec *codec) +{ + unsigned int hp_present, present = 0; + struct via_spec *spec = codec->spec; + int i; + + if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0]) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present = snd_hda_codec_read(codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + /* Mute Line-Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + if (hp_present) + present = hp_present; + } + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1680,6 +1967,10 @@ set_jack_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); + if (res & VIA_SPEAKER_EVENT) + via_speaker_automute(codec); + if (res & VIA_BIND_HP_EVENT) + via_hp_bind_automute(codec); } static int via_init(struct hda_codec *codec) @@ -2084,10 +2375,19 @@ /* init callback for auto-configuration model -- overriding the default init */ static int via_auto_init(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; + via_init(codec); via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); + if (spec->codec_type == VT2002P) { + via_hp_bind_automute(codec); + } else { + via_hp_automute(codec); + via_speaker_automute(codec); + } + return 0; } @@ -5018,6 +5318,359 @@ return 0; } + +/* for vt2002P */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt2002P_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt2002P_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x37, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3b, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; + + +static struct hda_verb vt2002P_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt2002P_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 3; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 4; + imux->num_items++; + + return 0; +} + +static int vt2002P_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt2002P_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt2002P */ +static int patch_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt2002P_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + + spec->stream_name_analog = "VT2002P Analog"; + spec->stream_analog_playback = &vt2002P_pcm_analog_playback; + spec->stream_analog_capture = &vt2002P_pcm_analog_capture; + + spec->stream_name_digital = "VT2002P Digital"; + spec->stream_digital_playback = &vt2002P_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt2002P_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt2002P_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt2002P_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -5102,6 +5755,8 @@ .patch = patch_vt1716S}, { .id = 0x1106a721, .name = "VT1716S", .patch = patch_vt1716S}, + { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 24/30] ALSA: HDA VIA: Add VT1716S support.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1716S support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 648 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 644 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -87,12 +87,13 @@ VT1708BCE, VT1702, VT1718S, + VT1716S, CODEC_TYPES, }; struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; + struct snd_kcontrol_new *mixers[6]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -135,7 +136,7 @@ unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; unsigned int smart51_enabled; - + unsigned int dmic_enabled; enum VIA_HDA_CODEC codec_type; /* work to check hp jack state */ @@ -179,6 +180,8 @@ else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0433 || dev_id == 0xa721) + codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; else @@ -189,6 +192,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 +#define VIA_MONO_EVENT 0x08 enum { VIA_CTL_WIDGET_VOL, @@ -295,6 +299,11 @@ 0x10, 0x11 }; +static hda_nid_t vt1716S_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -584,6 +593,106 @@ snd_hda_codec_write(codec, 0xc, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1716S) { + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write( + codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_codec_read( + codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + mono_out = 0; + else { + present = snd_hda_codec_read( + codec, 0x1d, 0, AC_VERB_GET_PIN_SENSE, 0) + & 0x80000000; + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); } } @@ -738,7 +847,8 @@ /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S || spec->codec_type == VT1702 - || spec->codec_type == VT1718S) { + || spec->codec_type == VT1718S + || spec->codec_type == VT1716S) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -797,6 +907,7 @@ case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -898,7 +1009,8 @@ HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { - if (spec->codec_type == VT1708S) { + if (spec->codec_type == VT1708S + || spec->codec_type == VT1716S) { /* input = index 1 (AOW3) */ snd_hda_codec_write( codec, nid, 0, @@ -961,6 +1073,7 @@ case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -1025,6 +1138,7 @@ break; case VT1708S: case VT1718S: + case VT1716S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1475,6 +1589,36 @@ } } +/* mute mono out if HP or Line out is plugged */ +static void via_mono_automute(struct hda_codec *codec) +{ + unsigned int hp_present, lineout_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1716S) + return; + + lineout_present = snd_hda_codec_read( + codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Mute Mono Out if Line Out is plugged */ + if (lineout_present) { + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); + return; + } + + hp_present = snd_hda_codec_read( + codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, + hp_present ? HDA_AMP_MUTE : 0); +} + static void via_gpio_control(struct hda_codec *codec) { unsigned int gpio_data; @@ -1534,6 +1678,8 @@ via_gpio_control(codec); if (res & VIA_JACK_EVENT) set_jack_power_state(codec); + if (res & VIA_MONO_EVENT) + via_mono_automute(codec); } static int via_init(struct hda_codec *codec) @@ -4382,6 +4528,496 @@ return 0; } + +/* Patch for VT1716S */ + +static int vt1716s_dmic_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int vt1716s_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int index = 0; + + index = snd_hda_codec_read(codec, 0x26, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (index != -1) + *ucontrol->value.integer.value = index; + + return 0; +} + +static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index = *ucontrol->value.integer.value; + + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_CONNECT_SEL, index); + spec->dmic_enabled = index; + set_jack_power_state(codec); + + return 1; +} + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1716S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Mic Capture Switch", + .count = 1, + .info = vt1716s_dmic_info, + .get = vt1716s_dmic_get, + .put = vt1716s_dmic_put, + }, + {} /* end */ +}; + + +/* mono-out mixer elements */ +static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb vt1716S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Stereo Mixer = 5 */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x5}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + + /* Setup default input of SW1 as MW0 */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* Setup default input of SW4 as AOW0 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* PW9 PW10 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute SW1, PW12 */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* PW12 Output enable */ + {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf8a, 0x80}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, + /* Enable mono output */ + {0x1, 0xf90, 0x08}, + { } +}; + + +static struct hda_verb vt1716S_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1716S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 3; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x25; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x11; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i <= AUTO_SEQ_CENLFE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x25; /* AOW3 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1716S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1716S_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1716S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1716S Analog"; + spec->stream_analog_playback = &vt1716S_pcm_analog_playback; + spec->stream_analog_capture = &vt1716S_pcm_analog_capture; + + spec->stream_name_digital = "VT1716S Digital"; + spec->stream_digital_playback = &vt1716S_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1716S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1716S_capture_mixer; + spec->num_mixers++; + } + + spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer; + spec->num_mixers++; + + spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1716S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -4462,6 +5098,10 @@ .patch = patch_vt1718S}, { .id = 0x11064441, .name = "VT1828S", .patch = patch_vt1718S}, + { .id = 0x11060433, .name = "VT1716S", + .patch = patch_vt1716S}, + { .id = 0x1106a721, .name = "VT1716S", + .patch = patch_vt1716S}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 23/30] ALSA: HDA VIA: Add VT1828S and VT2020 support.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1828S and VT2020 support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -179,6 +179,8 @@ else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0441 || dev_id == 0x4441) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -4340,21 +4342,31 @@ spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; - spec->stream_name_analog = "VT1718S Analog"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_analog = "VT2020 Analog"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_analog = "VT1828S Analog"; + else + spec->stream_name_analog = "VT1718S Analog"; spec->stream_analog_playback = &vt1718S_pcm_analog_playback; spec->stream_analog_capture = &vt1718S_pcm_analog_capture; - spec->stream_name_digital = "VT1718S Digital"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_digital = "VT2020 Digital"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_digital = "VT1828S Digital"; + else + spec->stream_name_digital = "VT1718S Digital"; spec->stream_digital_playback = &vt1718S_pcm_digital_playback; - if (codec->vendor_id == 0x11060428) + if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441) spec->stream_digital_capture = &vt1718S_pcm_digital_capture; if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1718S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); get_mux_nids(codec); - override_mic_boost(codec, 0x1a, 0, 3, 40); - override_mic_boost(codec, 0x1e, 0, 3, 40); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; spec->num_mixers++; } @@ -4446,6 +4458,10 @@ .patch = patch_vt1718S}, { .id = 0x11064428, .name = "VT1718S", .patch = patch_vt1718S}, + { .id = 0x11060441, .name = "VT2020", + .patch = patch_vt1718S}, + { .id = 0x11064441, .name = "VT1828S", + .patch = patch_vt1718S}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 22/30] ALSA: HDA VIA: Add VT1718S support.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1718S support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 554 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 545 insertions(+), 9 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -86,6 +86,7 @@ VT1708S, VT1708BCE, VT1702, + VT1718S, CODEC_TYPES, }; @@ -175,6 +176,9 @@ else if ((dev_id & 0xfff) == 0x398 && (dev_id >> 12) < 8) codec_type = VT1702; + else if ((dev_id & 0xfff) == 0x428 + && (dev_id >> 12) < 8) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -284,6 +288,11 @@ 0x12, 0x20, 0x1F }; +static hda_nid_t vt1718S_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -512,6 +521,67 @@ snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1718S) { + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW0/1 (24h/25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + set_pin_power_state(codec, 0x25, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, + parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } } } @@ -572,11 +642,21 @@ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; - unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_SEL, - 0x00); + hda_nid_t nid; + unsigned int pinsel; + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + /* use !! to translate conn sel 2 for VT1718S */ + pinsel = !!snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, + 0x00); ucontrol->value.enumerated.item[0] = pinsel; return 0; @@ -635,6 +715,16 @@ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ + spec->multiout.num_dacs = 4; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid @@ -645,7 +735,8 @@ update_side_mute_status(codec); /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S - || spec->codec_type == VT1702) { + || spec->codec_type == VT1702 + || spec->codec_type == VT1718S) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -758,7 +849,8 @@ AC_VERB_GET_PIN_WIDGET_CONTROL, 0); if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* ignore FMic for independent HP */ if (ctl & AC_PINCTL_IN_EN && !(ctl & AC_PINCTL_OUT_EN)) @@ -782,7 +874,8 @@ for (i = 0; i < ARRAY_SIZE(index); i++) { hda_nid_t nid = spec->autocfg.input_pins[index[i]]; if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* don't retask FMic for independent HP */ if (nid) { unsigned int parm = snd_hda_codec_read( @@ -797,6 +890,10 @@ mute_aa_path(codec, 1); notify_aa_path_ctls(codec); } + if (spec->codec_type == VT1718S) + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, + HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { if (spec->codec_type == VT1708S) { @@ -871,6 +968,11 @@ start_idx = 1; end_idx = 3; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return 0; } @@ -920,6 +1022,7 @@ parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ break; case VT1708S: + case VT1718S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1026,8 +1129,8 @@ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && - !spec->hp_independent_mode) + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] + && !spec->hp_independent_mode) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); @@ -3838,6 +3941,435 @@ return 0; } +/* Patch for VT1718S */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1718S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1718S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + + /* Setup default input of Front HP to MW9 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW9 PW10 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + /* PW11 Input enable */ + {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf88, 0x8}, + /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Unmute MW4's index 0 */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + + +static struct hda_verb vt1718S_uniwill_init_verbs[] = { + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 10, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1718S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x8; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0xa; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x9; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0xb; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; + hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; + hda_nid_t nid, nid_vol, nid_mute = 0; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0xc; /* AOW4 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 1; + break; + + case 0x2a: /* Line In */ + idx = 2; + break; + + case 0x29: /* Front Mic */ + idx = 3; + break; + + case 0x2c: /* CD */ + idx = 0; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + return 0; +} + +static int vt1718S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + + if (err < 0) + return err; + err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428) + spec->dig_in_nid = 0x13; + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1718S_loopbacks[] = { + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { 0x21, HDA_INPUT, 3 }, + { 0x21, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1718S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1718S Analog"; + spec->stream_analog_playback = &vt1718S_pcm_analog_playback; + spec->stream_analog_capture = &vt1718S_pcm_analog_capture; + + spec->stream_name_digital = "VT1718S Digital"; + spec->stream_digital_playback = &vt1718S_pcm_digital_playback; + if (codec->vendor_id == 0x11060428) + spec->stream_digital_capture = &vt1718S_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1718S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1718S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -3910,6 +4442,10 @@ .patch = patch_vt1702}, { .id = 0x11067398, .name = "VT1702", .patch = patch_vt1702}, + { .id = 0x11060428, .name = "VT1718S", + .patch = patch_vt1718S}, + { .id = 0x11064428, .name = "VT1718S", + .patch = patch_vt1718S}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 21/30] ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb As init verbs, vt17xx_volume_init_verb is a better place to hold them. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3085,6 +3085,8 @@ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* Enable Mic Boost Volume backdoor */ {0x1, 0xf98, 0x1}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, { } }; @@ -3544,6 +3546,10 @@ /* PW6 PW7 Output enable */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* mixer enable */ + {0x1, 0xF88, 0x3}, + /* GPIO 0~2 */ + {0x1, 0xF82, 0x3F}, { } }; @@ -3785,8 +3791,6 @@ { struct via_spec *spec; int err; - unsigned int response; - unsigned char control; /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -3831,18 +3835,6 @@ spec->loopback.amplist = vt1702_loopbacks; #endif - /* Open backdoor */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0); - control = (unsigned char)(response & 0xff); - control |= 0x3; - snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control); - - /* Enable GPIO 0&1 for volume&mute control */ - /* Enable GPIO 2 for DMIC-DATA */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0); - control = (unsigned char)((response >> 16) & 0x3f); - snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control); - return 0; }

1 0

[alsa-devel] [PATCH 20/30] ALSA: HDA VIA: Replace MIC_BOOST_VOLUME.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Replace MIC_BOOST_VOLUME. With snd_hda_override_amp_caps. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 72 +++++++++------------------------------------- 1 file changed, 15 insertions(+), 57 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -197,46 +197,6 @@ AUTO_SEQ_SIDE }; -/* Some VT1708S based boards gets the micboost setting wrong, so we have - * to apply some brute-force and re-write the TLV's by software. */ -static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *_tlv) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - if (size < 4 * sizeof(unsigned int)) - return -ENOMEM; - if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */ - return -EFAULT; - if (put_user(2 * sizeof(unsigned int), _tlv + 1)) - return -EFAULT; - if (put_user(0, _tlv + 2)) /* offset = 0 */ - return -EFAULT; - if (put_user(1000, _tlv + 3)) /* step size = 10 dB */ - return -EFAULT; - } - return 0; -} - -static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 3; - } - return 0; -} - static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); @@ -3080,29 +3040,15 @@ /* Patch for VT1708S */ -/* VT1708S software backdoor based override for buggy hardware micboost - * setting */ -#define MIC_BOOST_VOLUME(xname, nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ - .info = mic_boost_volume_info, \ - .get = snd_hda_mixer_amp_volume_get, \ - .put = snd_hda_mixer_amp_volume_put, \ - .tlv = { .c = mic_boost_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) } - /* capture mixer elements */ static struct snd_kcontrol_new vt1708S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), - MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A), - MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -3474,6 +3420,16 @@ }; #endif +static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, + int offset, int num_steps, int step_size) +{ + snd_hda_override_amp_caps(codec, pin, HDA_INPUT, + (offset << AC_AMPCAP_OFFSET_SHIFT) | + (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | + (step_size << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + static int patch_vt1708S(struct hda_codec *codec) { struct via_spec *spec; @@ -3510,6 +3466,8 @@ spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; }

1 0

[alsa-devel] [PATCH 19/30] ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2177,7 +2177,7 @@ { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -2186,43 +2186,45 @@ if (!nid) continue; + nid_vol = nid_vols[i]; + if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT){ - /* add control to mixer index 0 */ + /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -2243,26 +2245,26 @@ } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err;

1 0

[alsa-devel] [PATCH 18/30] ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1570,7 +1570,7 @@ { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid, nid_vol = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -1579,8 +1579,7 @@ if (!nid) continue; - if (i != AUTO_SEQ_FRONT) - nid_vol = 0x18 + i; + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -1612,13 +1611,13 @@ /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err;

1 0

[alsa-devel] [PATCH 17/30] ALSA: HDA VIA: Add 2nd S/PDIF out for VT1708S and VT1702.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add 2nd S/PDIF out for VT1708S and VT1702. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -107,7 +107,7 @@ /* playback */ struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; + hda_nid_t second_dig_out; /* capture */ unsigned int num_adc_nids; @@ -1179,8 +1179,23 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, format, substream); + hda_nid_t nid = substream->number == 0 + ? spec->multiout.dig_out_nid : spec->second_dig_out; + mutex_lock(&codec->spdif_mutex); + /* copy from hda_codec.c: setup_dig_out_stream */ + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write_cache( + codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); + /* turn on again (if needed) */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write_cache( + codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + mutex_unlock(&codec->spdif_mutex); + return 0; } static int via_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -1298,6 +1313,13 @@ if (err < 0) return err; spec->multiout.share_spdif = 1; + /* add second spdif out */ + if (spec->second_dig_out) { + err = snd_hda_create_spdif_out_ctls( + codec, spec->second_dig_out); + if (err < 0) + return err; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -1472,11 +1494,6 @@ } else /* enable SPDIF-input pin */ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - - /* assign slave outs */ - if (spec->slave_dig_outs[0]) - codec->slave_dig_outs = spec->slave_dig_outs; - return 0; } @@ -3164,7 +3181,7 @@ }; static struct hda_pcm_stream vt1708S_pcm_digital_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 2, /* NID is set in via_build_pcms */ @@ -3400,7 +3417,7 @@ if (!spec->multiout.dig_out_nid) spec->multiout.dig_out_nid = nid; else { - spec->slave_dig_outs[0] = nid; + spec->second_dig_out = nid; break; /* at most two dig outs */ } }

1 0

[alsa-devel] [PATCH 16/30] ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup Replaced with via_playback_multi_pcm_prepare/cleanup to support multi-stream operations Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 40 +++++++++------------------------------- 1 file changed, 9 insertions(+), 31 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1022,28 +1022,6 @@ hinfo); } -static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - vt1708_start_hp_work(spec); - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - vt1708_stop_hp_work(spec); - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - - static void playback_multi_pcm_prep_0(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -1252,7 +1230,7 @@ }; static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ @@ -1263,8 +1241,8 @@ .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -2062,8 +2040,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -2074,8 +2052,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -3166,8 +3144,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, .close = via_pcm_open_close }, };

1 0

[alsa-devel] [PATCH 15/30] ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function. like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port Connectivity field into 'AC_JACK_PORT_COMPLEX' Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1768,11 +1768,10 @@ def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { - if (seqassoc == 0xff) { - def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_set_pincfg(codec, nid, def_conf); - } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE + && (seqassoc == 0xf0 || seqassoc == 0xff)) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } return;

1 0

[alsa-devel] [PATCH 14/30] ALSA: HDA VIA: Add Jack detect feature for VT1708.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add Jack detect feature for VT1708. VT1708 does not support unsolicited response, but we need hp detect to automute speaker. Implemented in workqueue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 230 ++++++++++++++++++++++++++++++++++------------ 1 file changed, 173 insertions(+), 57 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -89,6 +89,64 @@ CODEC_TYPES, }; +struct via_spec { + /* codec parameterization */ + struct snd_kcontrol_new *mixers[4]; + unsigned int num_mixers; + + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; + + char *stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char *stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + hda_nid_t slave_dig_outs[2]; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; + hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* PCM information */ + struct hda_pcm pcm_rec[3]; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + struct snd_array kctls; + struct hda_input_mux private_imux[2]; + hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; + + enum VIA_HDA_CODEC codec_type; + + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work vt1708_hp_work; + int vt1708_jack_detectect; + int vt1708_hp_present; +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif +}; + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -181,6 +239,31 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); +static int is_aa_path_mute(struct hda_codec *codec); + +static void vt1708_start_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + if (!delayed_work_pending(&spec->vt1708_hp_work)) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); +} + +static void vt1708_stop_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 + && !is_aa_path_mute(spec->codec)) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + cancel_delayed_work(&spec->vt1708_hp_work); + flush_scheduled_work(); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -190,6 +273,12 @@ set_jack_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { + if (is_aa_path_mute(codec)) + vt1708_start_hp_work(codec->spec); + else + vt1708_stop_hp_work(codec->spec); + } return change; } @@ -210,59 +299,6 @@ }; -struct via_spec { - /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; - - struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - - char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - - char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; - - /* capture */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t mux_nids[3]; - hda_nid_t dig_in_nid; - hda_nid_t dig_in_pin; - - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; - - /* dynamic controls, init_verbs and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_imux[2]; - hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; - - /* HP mode source */ - const struct hda_input_mux *hp_mux; - unsigned int hp_independent_mode; - unsigned int hp_independent_mode_index; - unsigned int smart51_enabled; - - enum VIA_HDA_CODEC codec_type; - -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_loopback_check loopback; -#endif -}; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -981,7 +1017,6 @@ struct via_spec *spec = codec->spec; int idle = substream->pstr->substream_opened == 1 && substream->ref_count == 0; - analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); @@ -994,6 +1029,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_start_hp_work(spec); return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1003,6 +1039,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } @@ -1094,7 +1131,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); } - + vt1708_start_hp_work(spec); return 0; } @@ -1134,7 +1171,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); } - + vt1708_stop_hp_work(spec); return 0; } @@ -1345,6 +1382,7 @@ return; via_free_kctls(codec); + vt1708_stop_hp_work(spec); kfree(codec->spec); } @@ -1464,6 +1502,15 @@ return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static int via_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_POWER_SAVE static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -1479,6 +1526,9 @@ .build_pcms = via_build_pcms, .init = via_init, .free = via_free, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = via_suspend, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif @@ -1728,6 +1778,51 @@ return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1755,6 +1850,10 @@ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1788,6 +1887,22 @@ return 0; } +static void vt1708_update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + vt1708_hp_work.work); + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (spec->vt1708_hp_present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + spec->vt1708_hp_present ^= 1; + via_hp_automute(spec->codec); + } + vt1708_start_hp_work(spec); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1864,7 +1979,8 @@ #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; }

1 0

[alsa-devel] [PATCH 13/30] ALSA: HDA VIA: Refresh front playback mute in via_hp_automute.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Refresh front playback mute in via_hp_automute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1351,14 +1351,25 @@ /* mute internal speaker if HP is plugged */ static void via_hp_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = 0; struct via_spec *spec = codec->spec; present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], - HDA_OUTPUT, 0, HDA_AMP_MUTE, - present ? HDA_AMP_MUTE : 0); + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + /* auto mute */ + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Front Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } } static void via_gpio_control(struct hda_codec *codec)

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[alsa-devel] [PATCH 12/30] ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 38 +++++++++++++++++++++++++++++++------- 1 file changed, 31 insertions(+), 7 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -124,6 +124,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 +#define VIA_JACK_EVENT 0x04 enum { VIA_CTL_WIDGET_VOL, @@ -1413,10 +1414,12 @@ unsigned int res) { res >>= 26; - if (res == VIA_HP_EVENT) + if (res & VIA_HP_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) + if (res & VIA_GPIO_EVENT) via_gpio_control(codec); + if (res & VIA_JACK_EVENT) + set_jack_power_state(codec); } static int via_init(struct hda_codec *codec) @@ -1878,7 +1881,8 @@ }; static struct hda_verb vt1709_uniwill_init_verbs[] = { - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } }; @@ -2514,7 +2518,15 @@ }; static struct hda_verb vt1708B_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3009,7 +3021,15 @@ }; static struct hda_verb vt1708S_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3448,8 +3468,12 @@ }; static struct hda_verb vt1702_uniwill_init_verbs[] = { - {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } };

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[alsa-devel] [PATCH 11/30] ALSA: HDA VIA: When changing input source, update power state.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: When changing input source, update power state. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -549,6 +549,14 @@ if (!spec->mux_nids[adc_idx]) return -EINVAL; + /* switch to D0 beofre change index */ + if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) + snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* update jack power state */ + set_jack_power_state(codec); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);

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[alsa-devel] [PATCH 10/30] ALSA: HDA VIA: Add smart5.1 function.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add smart5.1 function. Smart 5.1 is for 3-jacks model, to reuse input pins as outputs. While off, they act as "line out" / "line in" / "mic in". While on, they acts as "line out" / "back left/right" / "center/lfe". Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 177 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 173 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -211,7 +211,7 @@ struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; + struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -253,6 +253,7 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; @@ -390,6 +391,8 @@ } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -400,9 +403,10 @@ & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ unsigned present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) >> 31; - - if ((no_presence || present) && get_defcfg_connect(def_conf) - != AC_JACK_PORT_NONE) { + struct via_spec *spec = codec->spec; + if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) + || ((no_presence || present) + && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { *affected_parm = AC_PWRST_D0; /* if it's connected */ parm = AC_PWRST_D0; } else @@ -657,6 +661,167 @@ { } /* end */ }; +static void notify_aa_path_ctls(struct hda_codec *codec) +{ + int i; + struct snd_ctl_elem_id id; + const char *labels[] = {"Mic", "Front Mic", "Line"}; + + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + for (i = 0; i < ARRAY_SIZE(labels); i++) { + sprintf(id.name, "%s Playback Volume", labels[i]); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +static void mute_aa_path(struct hda_codec *codec, int mute) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708: + nid_mixer = 0x17; + start_idx = 2; + end_idx = 4; + break; + case VT1709_10CH: + case VT1709_6CH: + nid_mixer = 0x18; + start_idx = 2; + end_idx = 4; + break; + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + default: + return; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE; + snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i, + HDA_AMP_MUTE, val); + } +} +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) +{ + int res = 0; + int index; + for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { + if (pin == spec->autocfg.input_pins[index]) { + res = 1; + break; + } + } + return res; +} + +static int via_smart51_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int via_smart51_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int on = 1; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + int ctl = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* ignore FMic for independent HP */ + if (ctl & AC_PINCTL_IN_EN + && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; + } + } + *ucontrol->value.integer.value = on; + return 0; +} + +static int via_smart51_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int out_in = *ucontrol->value.integer.value + ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* don't retask FMic for independent HP */ + if (nid) { + unsigned int parm = snd_hda_codec_read( + codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + } + if (i == AUTO_PIN_FRONT_MIC) { + if (spec->codec_type == VT1708S) { + /* input = index 1 (AOW3) */ + snd_hda_codec_write( + codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, 1); + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, + 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); + } + } + } + spec->smart51_enabled = *ucontrol->value.integer.value; + set_jack_power_state(codec); + return 1; +} + +static struct snd_kcontrol_new via_smart51_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Smart 5.1", + .count = 1, + .info = via_smart51_info, + .get = via_smart51_get, + .put = via_smart51_put, + }, + {} /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1587,6 +1752,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2087,6 +2253,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2649,6 +2816,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -3142,6 +3310,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; }

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[alsa-devel] [PATCH 9/30] ALSA: HDA VIA: Rewrite via_independent_hp_put
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Rewrite via_independent_hp_put Use hp_independent_mode_index to store hp index, and simplify function via_independent_hp_put with it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 87 ++++++++++++++++++++++++---------------------- 1 file changed, 47 insertions(+), 40 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -252,6 +252,7 @@ /* HP mode source */ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; enum VIA_HDA_CODEC codec_type; @@ -584,6 +585,36 @@ } } +static int update_side_mute_status(struct hda_codec *codec) +{ + /* mute side channel */ + struct via_spec *spec = codec->spec; + unsigned int parm = spec->hp_independent_mode + ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + hda_nid_t sw3; + + switch (spec->codec_type) { + case VT1708: + sw3 = 0x1b; + break; + case VT1709_10CH: + sw3 = 0x29; + break; + case VT1708B_8CH: + case VT1708S: + sw3 = 0x27; + break; + default: + sw3 = 0; + break; + } + + if (sw3) + snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, + parm); + return 0; +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -591,47 +622,18 @@ struct via_spec *spec = codec->spec; hda_nid_t nid = spec->autocfg.hp_pins[0]; unsigned int pinsel = ucontrol->value.enumerated.item[0]; - unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - - if (con_nid == spec->multiout.hp_nid) { - if (pinsel == 0) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } else if (pinsel == 1) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } - } else { - if (pinsel == 0) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } else if (pinsel == 1) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - pinsel); - - if (spec->multiout.hp_nid && - spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) - snd_hda_codec_setup_stream(codec, - spec->multiout.hp_nid, - 0, 0, 0); + /* Get Independent Mode index of headphone pin widget */ + spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel + ? 1 : 0; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + + if (spec->multiout.hp_nid && spec->multiout.hp_nid + != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, + 0, 0, 0); + update_side_mute_status(codec); /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S || spec->codec_type == VT1702) { @@ -1447,6 +1449,7 @@ return 0; spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1982,6 +1985,7 @@ spec->multiout.hp_nid = VT1709_HP_DAC_NID; else if (spec->multiout.num_dacs == 3) /* 6 channels */ spec->multiout.hp_nid = 0; + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2541,6 +2545,7 @@ return 0; spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3011,6 +3016,7 @@ return 0; spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3368,6 +3374,7 @@ if (!pin) return 0; spec->multiout.hp_nid = 0x1D; + spec->hp_independent_mode_index = 0; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume",

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[alsa-devel] [PATCH 8/30] ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls For VT1708S and VT1702, deactivate "Headphone Playback Volume" and "Headphone Playback Mute" control if "Independent HP" mode is OFF. and rename VT1702 "Independent HP" text. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 38 ++++++++++++++++++++++++++++++++++---- 1 file changed, 34 insertions(+), 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -572,6 +572,18 @@ return 0; } +static void activate_ctl(struct hda_codec *codec, const char *name, int active) +{ + struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); + if (ctl) { + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access |= active + ? 0 : SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id); + } +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -620,6 +632,14 @@ spec->multiout.hp_nid, 0, 0, 0); + /* update HP volume/swtich active state */ + if (spec->codec_type == VT1708S + || spec->codec_type == VT1702) { + activate_ctl(codec, "Headphone Playback Volume", + spec->hp_independent_mode); + activate_ctl(codec, "Headphone Playback Switch", + spec->hp_independent_mode); + } return 0; } @@ -3342,11 +3362,11 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { - int err; - + int err, i; + struct hda_input_mux *imux; + static const char *texts[] = { "ON", "OFF", NULL}; if (!pin) return 0; - spec->multiout.hp_nid = 0x1D; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -3361,8 +3381,18 @@ if (err < 0) return err; - create_hp_imux(spec); + imux = &spec->private_imux[1]; + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; return 0; }

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[alsa-devel] [PATCH 7/30] ALSA: HDA VIA: Remove unused argument of via_new_analog_input
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove unused argument of via_new_analog_input Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -317,8 +317,8 @@ } /* create input playback/capture controls for the given pin */ -static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, - const char *ctlname, int idx, int mix_nid) +static int via_new_analog_input(struct via_spec *spec, const char *ctlname, + int idx, int mix_nid) { char name[32]; int err; @@ -1480,8 +1480,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x17); + err = via_new_analog_input(spec, labels[i], idx, 0x17); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2014,8 +2013,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x18); + err = via_new_analog_input(spec, labels[i], idx, 0x18); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2576,8 +2574,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3048,8 +3045,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3402,8 +3398,7 @@ idx = 3; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], - labels[i], idx, 0x1A); + err = via_new_analog_input(spec, labels[i], idx, 0x1A); if (err < 0) return err; imux->items[imux->num_items].label = labels[i];

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[alsa-devel] [PATCH 6/30] ALSA: HDA VIA: Add low current mode for power saving.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add low current mode for power saving. For VT1708B, VT1708S and VT1702, enter low current mode if no analog stream is opened and all aa path mute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 41 +++++++++++++++++++++++++++++++++++------ 1 file changed, 35 insertions(+), 6 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -783,6 +783,10 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -1089,6 +1093,11 @@ if (err < 0) return err; } + + /* init power states */ + set_jack_power_state(codec); + analog_low_current_mode(codec, 1); + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -2312,6 +2321,17 @@ { } }; +static int via_pcm_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); + return 0; +} + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -2320,7 +2340,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2342,8 +2363,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2800,7 +2823,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .cleanup = via_playback_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2810,8 +2834,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3236,7 +3262,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3246,8 +3273,10 @@ .channels_max = 2, .nid = 0x12, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, };

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[alsa-devel] [PATCH 5/30] ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 240 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 239 insertions(+), 1 deletion(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -128,6 +128,7 @@ enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, + VIA_CTL_WIDGET_ANALOG_MUTE, }; enum { @@ -177,9 +178,34 @@ return 0; } +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + return change; +} + +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, }; @@ -303,7 +329,7 @@ if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -362,6 +388,131 @@ } } + +static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int *affected_parm) +{ + unsigned parm; + unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); + unsigned no_presence = (def_conf & AC_DEFCFG_MISC) + >> AC_DEFCFG_MISC_SHIFT + & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ + unsigned present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31; + + if ((no_presence || present) && get_defcfg_connect(def_conf) + != AC_JACK_PORT_NONE) { + *affected_parm = AC_PWRST_D0; /* if it's connected */ + parm = AC_PWRST_D0; + } else + parm = AC_PWRST_D3; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + +static void set_jack_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + + if (spec->codec_type == VT1702) { + imux_is_smixer = snd_hda_codec_read( + codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x16, &parm); + set_pin_power_state(codec, 0x17, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, + parm); + } else if (spec->codec_type == VT1708B_8CH + || spec->codec_type == VT1708B_4CH + || spec->codec_type == VT1708S) { + /* SW0 (17h) = stereo mixer */ + int is_8ch = spec->codec_type != VT1708B_4CH; + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } +} + /* * input MUX handling */ @@ -504,6 +655,93 @@ }, { } /* end */ }; + +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) +{ + struct via_spec *spec = codec->spec; + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; + case VT1708S: + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ + break; + case VT1702: + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ + break; + default: + return; /* other codecs are not supported */ + } + /* send verb */ + snd_hda_codec_write(codec, codec->afg, 0, verb, parm); +} + /* * generic initialization of ADC, input mixers and output mixers */

1 0

[alsa-devel] [PATCH 4/30] ALSA: HDA VIA: Limit VT1702 AA-Path max volume
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Limit VT1702 AA-Path max volume according to customer request, VT1702 AA-Path max volume (12 dB) is too high, so limit to 0 dB. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 6 ++++++ 1 file changed, 6 insertions(+) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3166,6 +3166,12 @@ err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; + /* limit AA path volume to 0 dB */ + snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err;

1 0

[alsa-devel] [PATCH 3/30] ALSA: HDA VIA: Add VT1708B-CE codec support.
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1708B-CE codec support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -84,6 +84,7 @@ VT1708B_8CH, VT1708B_4CH, VT1708S, + VT1708BCE, VT1702, CODEC_TYPES, }; @@ -104,9 +105,11 @@ codec_type = VT1709_10CH; else if (dev_id >= 0xe714 && dev_id <= 0xe717) codec_type = VT1709_6CH; - else if (dev_id >= 0xe720 && dev_id <= 0xe723) + else if (dev_id >= 0xe720 && dev_id <= 0xe723) { codec_type = VT1708B_8CH; - else if (dev_id >= 0xe724 && dev_id <= 0xe727) + if (snd_hda_param_read(codec, 0x16, AC_PAR_CONNLIST_LEN) == 0x7) + codec_type = VT1708BCE; + } else if (dev_id >= 0xe724 && dev_id <= 0xe727) codec_type = VT1708B_4CH; else if ((dev_id & 0xfff) == 0x397 && (dev_id >> 12) < 8) @@ -224,6 +227,8 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + enum VIA_HDA_CODEC codec_type; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -979,6 +984,10 @@ for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + spec->codec_type = get_codec_type(codec); + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost + same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { @@ -2369,12 +2378,14 @@ { } /* end */ }; #endif - +static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { struct via_spec *spec; int err; + if (get_codec_type(codec) == VT1708BCE) + return patch_vt1708S(codec); /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2906,6 +2917,16 @@ spec->loopback.amplist = vt1708S_loopbacks; #endif + /* correct names for VT1708BCE */ + if (get_codec_type(codec) == VT1708BCE) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + spec->stream_name_analog = "VT1708BCE Analog"; + spec->stream_name_digital = "VT1708BCE Digital"; + } return 0; }

1 0

[alsa-devel] [PATCH 2/30] ALSA: HDA VIA: Change get_codec_type argument to hda_codec type
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Change get_codec_type argument to hda_codec type Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -88,8 +88,9 @@ CODEC_TYPES, }; -static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { + u32 vendor_id = codec->vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -141,7 +142,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { if (size < 4 * sizeof(unsigned int)) return -ENOMEM; @@ -163,7 +164,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2;

1 0

[alsa-devel] [PATCH 1/30] ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro
by Logan Li 10 Oct '09

10 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro IS_VT17*_VENDORID macros are used nowhere, so clean them up. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 8 -------- 1 file changed, 8 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -76,14 +76,6 @@ #define VT1702_HP_NID 0x17 #define VT1702_DIGOUT_NID 0x11 -#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) -#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) -#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) -#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) -#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) -#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) -#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) - enum VIA_HDA_CODEC { UNKNOWN = -1, VT1708,

1 0

[alsa-devel] [PATCH] sscape: coding style fixes
by Krzysztof Helt 10 Oct '09

10 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix coding style errors in the driver. Also, add missing argument for CMD_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The third trial. I hope I got it right this time. sound/isa/sscape.c | 169 +++++++++++++++++++++++++-------------------------- 1 files changed, 83 insertions(+), 86 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be50..0a706c4 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -190,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -329,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) * The AD1845 detection fails if we *don't* do this, so I * think that this is a good idea ... */ -static inline void activate_ad1845_unsafe(unsigned io_base) +static void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -350,24 +354,27 @@ static void soundscape_free(struct snd_card *c) * Tell the SoundScape to begin a DMA tranfer using the given channel. * All locking issues are left to the caller. */ -static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) +static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -427,13 +434,13 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) /* * Upload a byte-stream into the SoundScape using DMA channel A. */ -static int upload_dma_data(struct soundscape *s, - const unsigned char *data, - size_t size) +static int upload_dma_data(struct soundscape *s, const unsigned char *data, + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +450,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +482,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +499,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -591,7 +602,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +612,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +626,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +650,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +752,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +776,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +792,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +803,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +817,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +880,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +929,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1034,7 +1035,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1055,6 +1055,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1094,11 +1098,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* * Initialize mixer */ spin_lock_irqsave(&sscape->lock, flags); @@ -1155,7 +1154,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1183,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1237,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1294,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- Przygotuj si� na przysz�o��! Kliknij >>> http://link.interia.pl/f238a

2 1

[alsa-devel] [PATCH] wss: convert CS4231 mixer to dB scale
by Krzysztof Helt 10 Oct '09

10 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Convert CS4231 mixer to dB scale after AD1848 mixer. Also, add missing microphone boost control for the AD1848 and correct wrong bits for loopback volume on the AD1848. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I wonder if it is a good idea to reorder the snd_wss_controls table so the first part is identical to the snd_ad1848_controls. The snd_ad1848_controls can be removed completely and the first part of the snd_wss_controls can be used for the AD1848 chips. sound/isa/wss/wss_lib.c | 43 ++++++++++++++++++++++++++----------------- 1 files changed, 26 insertions(+), 17 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5d2ba1b..754a208 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2198,6 +2198,7 @@ EXPORT_SYMBOL(snd_wss_put_double); static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); static struct snd_kcontrol_new snd_ad1848_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, @@ -2224,38 +2225,45 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, +WSS_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), }; static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), WSS_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -2267,15 +2275,16 @@ WSS_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE("Loopback Capture Volume", 0, - CS4231_LOOPBACK, 2, 63, 1) +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, + db_scale_6bit), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Playback Volume", 0, -- 1.6.0.3 ---------------------------------------------------------------------- Szukasz pracy? Chcesz lepiej zarabia�? Sprawd� oferty na http://link.interia.pl/f23ba

2 1

[alsa-devel] [PATCH] ice1724: Fix surround on Chaintech AV-710
by Robert Hancock 10 Oct '09

10 Oct '09

Fix the num_total_dacs setting for Chaintech AV710. The existing comment that only PSDOUT0 is connected is correct, but since the card is using packed AC97 mode to send 6 channels to the codec, num_total_dacs should be set to 6 and not 2. This allows 6-channel surround to work. Also clarify a comment regarding the additional WM8728 codec on this card (it's connected to the SPDIF output and always receives the same data). Signed-off-by: Robert Hancock <hancockrwd(a)gmail.com> diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 3756430..bfcf989 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) /* only use basic functionality for now */ - ice->num_total_dacs = 2; /* only PSDOUT0 is connected */ + /* VT1616 6ch codec connected to PSDOUT0 using packed mode */ + ice->num_total_dacs = 6; ice->num_total_adcs = 2; - /* Chaintech AV-710 has another codecs, which need initialization */ - /* initialize WM8728 codec */ + /* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4 + (shared with the SPDIF output). Mixer control for this codec + is not yet supported. */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) { for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) wm_put(ice, wm_inits[i], wm_inits[i+1]);

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 10 Oct '09

10 Oct '09

The following changes since commit 907bc6c7fc7071b00083fc11e510e47dd93df45d: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Mark Brown (5): ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSI ASoC: WM8350 capture PGA mutes are inverted Merge branch 'upstream/wm8350' into for-2.6.32 Merge branch 'for-2.6.32' into for-2.6.33 ASoC: Minor fixups to tpa6130a2 driver Nicolas Ferre (1): ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek Peter Ujfalusi (1): ASoC: TPA6130A2 amplifier driver include/sound/tpa6130a2-plat.h | 30 +++ sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tpa6130a2.c | 463 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tpa6130a2.h | 62 +++++ sound/soc/codecs/wm8350.c | 4 +- sound/soc/imx/mxc-ssi.c | 8 - 8 files changed, 564 insertions(+), 11 deletions(-) create mode 100644 include/sound/tpa6130a2-plat.h create mode 100644 sound/soc/codecs/tpa6130a2.c create mode 100644 sound/soc/codecs/tpa6130a2.h

2 1

[alsa-devel] VIA HDA patch problem (was [PATCH 17/31] HDA patch_via.c: Add second S/PDIF out control for VT1708S and VT1702.)
by Robert Hancock 10 Oct '09

10 Oct '09

It looks like there's some problem somewhere in the patch set posted on Oct. 7th that wasn't in the original patch set from the 5th: I can't open the mixer for the HDA device: alsamixer -c 1 cannot load mixer controls: Operation not permitted _From strace: open("/dev/snd/controlC1", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0x7fff338e57fc) = 0 fcntl(3, F_GETFL) = 0x8002 (flags O_RDWR|O_LARGEFILE) fcntl(3, F_SETFL, O_RDWR|O_NONBLOCK|O_LARGEFILE) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0x7fff338e5990) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0x7fff338e5990) = 0 ioctl(3, USBDEVFS_CONNECTINFO, 0x7fff338e5670) = 0 ioctl(3, USBDEVFS_IOCTL, 0x7fff338e4ea0) = -1 EPERM (Operation not permitted) Presumably something is creating a control that isn't readable, or writable or something, and this is causing the mixer to fail to load? Also, with regard to the SPDIF problem I'm seeing on my Asus P7P55D (VT1828S), I was able to dig up a little bit more info: by using hda_analyzer to set the ENABLE flag on the Digital Converter, I was able to get the optical SPDIF port to light up and my receiver indicates PCM input. Unfortunately there's still no actual audio coming through on the receiver, just silence. However, obviously that ENABLE flag is going to have to get set somehow to have any hope of SPDIF working, and currently it seems like it's not..

2 2

[alsa-devel] ascenario: Last batch of fixes.
by Stefan Schmidt 09 Oct '09

09 Oct '09

Hello. This is the last set of fixes I have in my queue. [PATCH 1/3] ascenario: Add check for set scenario in snd_scenario_get_control_id() Debugged some hours that I should test it after _set() and not only after _list(). Better let the user know that no scenario is set instead of silently using a non existing scenario and fail. [PATCH 2/3] ascenario: Convert last sprintf() to snprintf() Some more sprintf() replaced with the more secure snprintf(). [PATCH 3/3] ascenario: Make sequences unique for every scenario. That one actually got me some grey hairs. I'm pretty sure I tested it earlier and it worked as it should, but I can find now traces of this. Fixed for good now. [PATCH] ascenario: Add dump and configure utils for ascneario. Reference patch against alsa-utils for some small utilities using the new scenario API. Only change from the last time is that _dump() now has two parameters even if the new one is still unused. regards Stefan Schmidt

1 4

[alsa-devel] [PATCH 0/30] ALSA HDA VIA patches
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add Jack detect feature for VT1708. VT1708 does not support unsolicited response, but we need hp detect to automute speaker. Implemented in workqueue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 230 +++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 173 insertions(+), 57 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -89,6 +89,64 @@ CODEC_TYPES, }; +struct via_spec { + /* codec parameterization */ + struct snd_kcontrol_new *mixers[4]; + unsigned int num_mixers; + + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; + + char *stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char *stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + hda_nid_t slave_dig_outs[2]; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; + hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* PCM information */ + struct hda_pcm pcm_rec[3]; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + struct snd_array kctls; + struct hda_input_mux private_imux[2]; + hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; + + enum VIA_HDA_CODEC codec_type; + + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work vt1708_hp_work; + int vt1708_jack_detectect; + int vt1708_hp_present; +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif +}; + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -181,6 +239,31 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); +static int is_aa_path_mute(struct hda_codec *codec); + +static void vt1708_start_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + if (!delayed_work_pending(&spec->vt1708_hp_work)) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); +} + +static void vt1708_stop_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 + && !is_aa_path_mute(spec->codec)) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + cancel_delayed_work(&spec->vt1708_hp_work); + flush_scheduled_work(); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -190,6 +273,12 @@ set_jack_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { + if (is_aa_path_mute(codec)) + vt1708_start_hp_work(codec->spec); + else + vt1708_stop_hp_work(codec->spec); + } return change; } @@ -210,59 +299,6 @@ }; -struct via_spec { - /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; - - struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - - char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - - char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; - - /* capture */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t mux_nids[3]; - hda_nid_t dig_in_nid; - hda_nid_t dig_in_pin; - - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; - - /* dynamic controls, init_verbs and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_imux[2]; - hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; - - /* HP mode source */ - const struct hda_input_mux *hp_mux; - unsigned int hp_independent_mode; - unsigned int hp_independent_mode_index; - unsigned int smart51_enabled; - - enum VIA_HDA_CODEC codec_type; - -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_loopback_check loopback; -#endif -}; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -981,7 +1017,6 @@ struct via_spec *spec = codec->spec; int idle = substream->pstr->substream_opened == 1 && substream->ref_count == 0; - analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); @@ -994,6 +1029,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_start_hp_work(spec); return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1003,6 +1039,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } @@ -1094,7 +1131,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); } - + vt1708_start_hp_work(spec); return 0; } @@ -1134,7 +1171,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); } - + vt1708_stop_hp_work(spec); return 0; } @@ -1345,6 +1382,7 @@ return; via_free_kctls(codec); + vt1708_stop_hp_work(spec); kfree(codec->spec); } @@ -1464,6 +1502,15 @@ return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static int via_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_POWER_SAVE static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -1479,6 +1526,9 @@ .build_pcms = via_build_pcms, .init = via_init, .free = via_free, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = via_suspend, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif @@ -1728,6 +1778,51 @@ return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1755,6 +1850,10 @@ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1788,6 +1887,22 @@ return 0; } +static void vt1708_update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + vt1708_hp_work.work); + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (spec->vt1708_hp_present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + spec->vt1708_hp_present ^= 1; + via_hp_automute(spec->codec); + } + vt1708_start_hp_work(spec); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1863,7 +1978,8 @@ #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; }

2 1

[alsa-devel] [PATCH] sound: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek
by Nicolas Ferre 09 Oct '09

09 Oct '09

The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file to extend the machine ID checking. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 885ba01..e028744 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static int __init at91sam9g20ek_init(void) struct clk *pllb; int ret; - if (!machine_is_at91sam9g20ek()) + if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) return -ENODEV; /* -- 1.5.6.5

2 1

[alsa-devel] [PATCH 30/30] ALSA: HDA VIA: Only cosmetic changes
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Only cosmetic changes Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 64 ++++++++++++++++++++++++++-------------------------- 1 file changed, 33 insertions(+), 31 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -22,26 +22,26 @@ */ /* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */ -/* */ +/* */ /* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */ -/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ -/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ +/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ +/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ -/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ -/* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ /* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ -/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ -/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ -/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ -/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ /* 2009-02-16 Logan Li Add support for VT1718S */ /* 2009-03-13 Logan Li Add support for VT1716S */ /* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ /* 2009-07-08 Lydia Wang Add support for VT2002P */ /* 2009-07-21 Lydia Wang Add support for VT1812 */ -/* */ +/* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -486,7 +486,7 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -1545,7 +1545,7 @@ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* Setup default input MW0 to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ @@ -1887,8 +1887,10 @@ codec->pcm_info = info; info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -2133,7 +2135,7 @@ } else /* enable SPDIF-input pin */ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - return 0; + return 0; } #ifdef SND_HDA_NEEDS_RESUME @@ -2178,8 +2180,8 @@ spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -2217,7 +2219,7 @@ if (!nid) continue; - + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { @@ -2246,7 +2248,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2260,7 +2262,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2360,7 +2362,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -2593,7 +2595,7 @@ "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -2605,7 +2607,7 @@ spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -2791,11 +2793,11 @@ spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ - switch(i) { + switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ spec->multiout.dac_nids[i] = 0x10; @@ -2830,7 +2832,7 @@ for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; nid_vol = nid_vols[i]; @@ -2861,7 +2863,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2875,7 +2877,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2971,7 +2973,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -3080,7 +3082,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -3174,7 +3176,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);

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[alsa-devel] [PATCH 29/30] ALSA: HDA VIA: Comments: update copyright, changeset, etc.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: comments: update copyright, changeset, etc. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -1,10 +1,10 @@ /* * Universal Interface for Intel High Definition Audio Codec * - * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec + * HD audio interface patch for VIA VT17xx/VT18xx/VT20xx codec * - * Copyright (c) 2006-2008 Lydia Wang <lydiawang(a)viatech.com> - * Takashi Iwai <tiwai(a)suse.de> + * (C) 2006-2009 VIA Technology, Inc. + * (C) 2006-2008 Takashi Iwai <tiwai(a)suse.de> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -36,6 +36,11 @@ /* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ /* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2009-02-16 Logan Li Add support for VT1718S */ +/* 2009-03-13 Logan Li Add support for VT1716S */ +/* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ +/* 2009-07-08 Lydia Wang Add support for VT2002P */ +/* 2009-07-21 Lydia Wang Add support for VT1812 */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */

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[alsa-devel] [PATCH 28/30] ALSA: HDA VIA: Change PW4 connect select default to to MW0.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Change PW4 connect select default to to MW0. According to customer request, hp should be default to redirected mode, i.e. PW4 connect select default to to MW0. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -1541,8 +1541,8 @@ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Setup default input to PW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Setup default input MW0 to PW4 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -2684,8 +2684,8 @@ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set input of PW4 as AOW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Set input of PW4 as MW0 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -3238,7 +3238,7 @@ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Setup default input to PW4 */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* PW10 Input enable */

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[alsa-devel] [PATCH 27/30] ALSA: HDA VIA: rename vt1708_control_templates[].
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: rename vt1708_control_templates[]. To via_control_templates[]. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -367,7 +367,7 @@ .put = bind_pin_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -static struct snd_kcontrol_new vt1708_control_templates[] = { +static struct snd_kcontrol_new via_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, @@ -430,7 +430,7 @@ knew = snd_array_new(&spec->kctls); if (!knew) return -ENOMEM; - *knew = vt1708_control_templates[type]; + *knew = via_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM;

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[alsa-devel] [PATCH 26/30] ALSA: HDA VIA: Add VT1812 support.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1812 support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 494 +++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 491 insertions(+), 3 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -89,6 +89,7 @@ VT1718S, VT1716S, VT2002P, + VT1812, CODEC_TYPES, }; @@ -187,6 +188,8 @@ codec_type = VT1718S; else if (dev_id == 0x0438 || dev_id == 0x4438) codec_type = VT2002P; + else if (dev_id == 0x0448) + codec_type = VT1812; else codec_type = UNKNOWN; return codec_type; @@ -411,6 +414,12 @@ 0x10, 0x11 }; +static hda_nid_t vt1812_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -904,6 +913,120 @@ snd_hda_codec_write( codec, 0x21, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else if (spec->codec_type == VT1812) { + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_codec_read( + codec, 0x28, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -974,6 +1097,9 @@ case VT2002P: nid = 0x35; break; + case VT1812: + nid = 0x3d; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -1049,6 +1175,9 @@ case VT2002P: nid = 0x35; break; + case VT1812: + nid = 0x3d; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -1066,7 +1195,8 @@ || spec->codec_type == VT1702 || spec->codec_type == VT1718S || spec->codec_type == VT1716S - || spec->codec_type == VT2002P) { + || spec->codec_type == VT2002P + || spec->codec_type == VT1812) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -1307,6 +1437,7 @@ end_idx = 3; break; case VT2002P: + case VT1812: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -1370,6 +1501,7 @@ parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; case VT2002P: + case VT1812: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -1900,7 +2032,7 @@ unsigned int hp_present; struct via_spec *spec = codec->spec; - if (spec->codec_type != VT2002P) + if (spec->codec_type != VT2002P && spec->codec_type != VT1812) return; hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, @@ -2381,7 +2513,7 @@ via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P) { + if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5671,6 +5803,361 @@ return 0; } + +/* for vt1812 */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1812_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1812_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/13/15 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0xa8}, + { } +}; + + +static struct hda_verb vt1812_uniwill_init_verbs[] = { + {0x33, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT }, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1812_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1812_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + + +/* add playback controls from the parsed DAC table */ +static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 6; + imux->num_items++; + + return 0; +} + +static int vt1812_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + fill_dig_outs(codec); + err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs) + return 0; /* can't find valid BIOS pin config */ + + err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1812_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt1812 */ +static int patch_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1812_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + + spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs; + + spec->stream_name_analog = "VT1812 Analog"; + spec->stream_analog_playback = &vt1812_pcm_analog_playback; + spec->stream_analog_capture = &vt1812_pcm_analog_capture; + + spec->stream_name_digital = "VT1812 Digital"; + spec->stream_digital_playback = &vt1812_pcm_digital_playback; + + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1812_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1812_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1812_loopbacks; +#endif + + return 0; +} + /* * patch entries */ @@ -5757,6 +6244,7 @@ .patch = patch_vt1716S}, { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 25/30] ALSA: HDA VIA: Add VT2002P support.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT2002P support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 665 +++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 660 insertions(+), 5 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -88,6 +88,7 @@ VT1702, VT1718S, VT1716S, + VT2002P, CODEC_TYPES, }; @@ -184,6 +185,8 @@ codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; + else if (dev_id == 0x0438 || dev_id == 0x4438) + codec_type = VT2002P; else codec_type = UNKNOWN; return codec_type; @@ -193,11 +196,14 @@ #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 #define VIA_MONO_EVENT 0x08 +#define VIA_SPEAKER_EVENT 0x10 +#define VIA_BIND_HP_EVENT 0x20 enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, VIA_CTL_WIDGET_ANALOG_MUTE, + VIA_CTL_WIDGET_BIND_PIN_MUTE, }; enum { @@ -235,6 +241,7 @@ flush_scheduled_work(); } + static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -262,13 +269,108 @@ .put = analog_input_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } +static void via_hp_bind_automute(struct hda_codec *codec); + +static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int i; + int change = 0; + + long *valp = ucontrol->value.integer.value; + int lmute, rmute; + if (strstr(kcontrol->id.name, "Switch") == NULL) { + snd_printd("Invalid control!\n"); + return change; + } + change = snd_hda_mixer_amp_switch_put(kcontrol, + ucontrol); + /* Get mute value */ + lmute = *valp ? 0 : HDA_AMP_MUTE; + valp++; + rmute = *valp ? 0 : HDA_AMP_MUTE; + + /* Set hp pins */ + if (!spec->hp_independent_mode) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + + if (!lmute && !rmute) { + /* Line Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* unmute */ + via_hp_bind_automute(codec); + + } else { + if (lmute) { + /* Mute all left channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + } + if (rmute) { + /* mute all right channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + return change; +} + +#define BIND_PIN_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = bind_pin_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, + BIND_PIN_MUTE, }; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -304,6 +406,11 @@ 0x13, 0x14 }; +static hda_nid_t vt2002P_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -386,10 +493,13 @@ { struct via_spec *spec = codec->spec; hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + } } static void via_auto_init_analog_input(struct hda_codec *codec) @@ -693,6 +803,107 @@ imux_is_smixer ? AC_PWRST_D0 : parm); snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, mono_out ? AC_PWRST_D0 : parm); + } else if (spec->codec_type == VT2002P) { + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, + 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Class-D */ + /* PW0 (24h), MW0(18h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* Mono Out */ + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + present = snd_hda_codec_read( + codec, 0x26, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -760,6 +971,9 @@ case VT1718S: nid = 0x34; break; + case VT2002P: + nid = 0x35; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -832,6 +1046,9 @@ pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ spec->multiout.num_dacs = 4; break; + case VT2002P: + nid = 0x35; + break; default: nid = spec->autocfg.hp_pins[0]; break; @@ -848,7 +1065,8 @@ if (spec->codec_type == VT1708S || spec->codec_type == VT1702 || spec->codec_type == VT1718S - || spec->codec_type == VT1716S) { + || spec->codec_type == VT1716S + || spec->codec_type == VT2002P) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -1088,6 +1306,11 @@ start_idx = 1; end_idx = 3; break; + case VT2002P: + nid_mixer = 0x21; + start_idx = 0; + end_idx = 2; + break; default: return 0; } @@ -1146,6 +1369,10 @@ verb = 0xf73; parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; + case VT2002P: + verb = 0xf93; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ + break; default: return; /* other codecs are not supported */ } @@ -1667,6 +1894,66 @@ } } +/* mute Internal-Speaker if HP is plugged */ +static void via_speaker_automute(struct hda_codec *codec) +{ + unsigned int hp_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT2002P) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Speaker Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +/* mute line-out and internal speaker if HP is plugged */ +static void via_hp_bind_automute(struct hda_codec *codec) +{ + unsigned int hp_present, present = 0; + struct via_spec *spec = codec->spec; + int i; + + if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0]) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present = snd_hda_codec_read(codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + /* Mute Line-Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + if (hp_present) + present = hp_present; + } + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1680,6 +1967,10 @@ set_jack_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); + if (res & VIA_SPEAKER_EVENT) + via_speaker_automute(codec); + if (res & VIA_BIND_HP_EVENT) + via_hp_bind_automute(codec); } static int via_init(struct hda_codec *codec) @@ -2084,10 +2375,19 @@ /* init callback for auto-configuration model -- overriding the default init */ static int via_auto_init(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; + via_init(codec); via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); + if (spec->codec_type == VT2002P) { + via_hp_bind_automute(codec); + } else { + via_hp_automute(codec); + via_speaker_automute(codec); + } + return 0; } @@ -5018,6 +5318,359 @@ return 0; } + +/* for vt2002P */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt2002P_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt2002P_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x37, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3b, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; + + +static struct hda_verb vt2002P_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt2002P_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 3; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 4; + imux->num_items++; + + return 0; +} + +static int vt2002P_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt2002P_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt2002P */ +static int patch_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt2002P_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + + spec->stream_name_analog = "VT2002P Analog"; + spec->stream_analog_playback = &vt2002P_pcm_analog_playback; + spec->stream_analog_capture = &vt2002P_pcm_analog_capture; + + spec->stream_name_digital = "VT2002P Digital"; + spec->stream_digital_playback = &vt2002P_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt2002P_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt2002P_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt2002P_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -5102,6 +5755,8 @@ .patch = patch_vt1716S}, { .id = 0x1106a721, .name = "VT1716S", .patch = patch_vt1716S}, + { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 24/30] ALSA: HDA VIA: Add VT1716S support.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1716S support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 648 +++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 644 insertions(+), 4 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -87,12 +87,13 @@ VT1708BCE, VT1702, VT1718S, + VT1716S, CODEC_TYPES, }; struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; + struct snd_kcontrol_new *mixers[6]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -135,7 +136,7 @@ unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; unsigned int smart51_enabled; - + unsigned int dmic_enabled; enum VIA_HDA_CODEC codec_type; /* work to check hp jack state */ @@ -179,6 +180,8 @@ else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0433 || dev_id == 0xa721) + codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; else @@ -189,6 +192,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 +#define VIA_MONO_EVENT 0x08 enum { VIA_CTL_WIDGET_VOL, @@ -295,6 +299,11 @@ 0x10, 0x11 }; +static hda_nid_t vt1716S_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -584,6 +593,106 @@ snd_hda_codec_write(codec, 0xc, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1716S) { + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write( + codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_codec_read( + codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + mono_out = 0; + else { + present = snd_hda_codec_read( + codec, 0x1d, 0, AC_VERB_GET_PIN_SENSE, 0) + & 0x80000000; + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); } } @@ -738,7 +847,8 @@ /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S || spec->codec_type == VT1702 - || spec->codec_type == VT1718S) { + || spec->codec_type == VT1718S + || spec->codec_type == VT1716S) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -797,6 +907,7 @@ case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -898,7 +1009,8 @@ HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { - if (spec->codec_type == VT1708S) { + if (spec->codec_type == VT1708S + || spec->codec_type == VT1716S) { /* input = index 1 (AOW3) */ snd_hda_codec_write( codec, nid, 0, @@ -961,6 +1073,7 @@ case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -1025,6 +1138,7 @@ break; case VT1708S: case VT1718S: + case VT1716S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1475,6 +1589,36 @@ } } +/* mute mono out if HP or Line out is plugged */ +static void via_mono_automute(struct hda_codec *codec) +{ + unsigned int hp_present, lineout_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1716S) + return; + + lineout_present = snd_hda_codec_read( + codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Mute Mono Out if Line Out is plugged */ + if (lineout_present) { + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); + return; + } + + hp_present = snd_hda_codec_read( + codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, + hp_present ? HDA_AMP_MUTE : 0); +} + static void via_gpio_control(struct hda_codec *codec) { unsigned int gpio_data; @@ -1534,6 +1678,8 @@ via_gpio_control(codec); if (res & VIA_JACK_EVENT) set_jack_power_state(codec); + if (res & VIA_MONO_EVENT) + via_mono_automute(codec); } static int via_init(struct hda_codec *codec) @@ -4382,6 +4528,496 @@ return 0; } + +/* Patch for VT1716S */ + +static int vt1716s_dmic_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int vt1716s_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int index = 0; + + index = snd_hda_codec_read(codec, 0x26, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (index != -1) + *ucontrol->value.integer.value = index; + + return 0; +} + +static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index = *ucontrol->value.integer.value; + + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_CONNECT_SEL, index); + spec->dmic_enabled = index; + set_jack_power_state(codec); + + return 1; +} + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1716S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Mic Capture Switch", + .count = 1, + .info = vt1716s_dmic_info, + .get = vt1716s_dmic_get, + .put = vt1716s_dmic_put, + }, + {} /* end */ +}; + + +/* mono-out mixer elements */ +static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb vt1716S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Stereo Mixer = 5 */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x5}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + + /* Setup default input of SW1 as MW0 */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* Setup default input of SW4 as AOW0 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* PW9 PW10 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute SW1, PW12 */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* PW12 Output enable */ + {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf8a, 0x80}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, + /* Enable mono output */ + {0x1, 0xf90, 0x08}, + { } +}; + + +static struct hda_verb vt1716S_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1716S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 3; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x25; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x11; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i <= AUTO_SEQ_CENLFE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x25; /* AOW3 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1716S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1716S_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1716S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1716S Analog"; + spec->stream_analog_playback = &vt1716S_pcm_analog_playback; + spec->stream_analog_capture = &vt1716S_pcm_analog_capture; + + spec->stream_name_digital = "VT1716S Digital"; + spec->stream_digital_playback = &vt1716S_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1716S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1716S_capture_mixer; + spec->num_mixers++; + } + + spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer; + spec->num_mixers++; + + spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1716S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -4462,6 +5098,10 @@ .patch = patch_vt1718S}, { .id = 0x11064441, .name = "VT1828S", .patch = patch_vt1718S}, + { .id = 0x11060433, .name = "VT1716S", + .patch = patch_vt1716S}, + { .id = 0x1106a721, .name = "VT1716S", + .patch = patch_vt1716S}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 23/30] ALSA: HDA VIA: Add VT1828S and VT2020 support.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1828S and VT2020 support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -179,6 +179,8 @@ else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0441 || dev_id == 0x4441) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -4340,21 +4342,31 @@ spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; - spec->stream_name_analog = "VT1718S Analog"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_analog = "VT2020 Analog"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_analog = "VT1828S Analog"; + else + spec->stream_name_analog = "VT1718S Analog"; spec->stream_analog_playback = &vt1718S_pcm_analog_playback; spec->stream_analog_capture = &vt1718S_pcm_analog_capture; - spec->stream_name_digital = "VT1718S Digital"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_digital = "VT2020 Digital"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_digital = "VT1828S Digital"; + else + spec->stream_name_digital = "VT1718S Digital"; spec->stream_digital_playback = &vt1718S_pcm_digital_playback; - if (codec->vendor_id == 0x11060428) + if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441) spec->stream_digital_capture = &vt1718S_pcm_digital_capture; if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1718S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); get_mux_nids(codec); - override_mic_boost(codec, 0x1a, 0, 3, 40); - override_mic_boost(codec, 0x1e, 0, 3, 40); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; spec->num_mixers++; } @@ -4446,6 +4458,10 @@ .patch = patch_vt1718S}, { .id = 0x11064428, .name = "VT1718S", .patch = patch_vt1718S}, + { .id = 0x11060441, .name = "VT2020", + .patch = patch_vt1718S}, + { .id = 0x11064441, .name = "VT1828S", + .patch = patch_vt1718S}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 22/30] ALSA: HDA VIA: Add VT1718S support.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1718S support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 555 +++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 546 insertions(+), 9 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -86,6 +86,7 @@ VT1708S, VT1708BCE, VT1702, + VT1718S, CODEC_TYPES, }; @@ -175,6 +176,9 @@ else if ((dev_id & 0xfff) == 0x398 && (dev_id >> 12) < 8) codec_type = VT1702; + else if ((dev_id & 0xfff) == 0x428 + && (dev_id >> 12) < 8) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -284,6 +288,11 @@ 0x12, 0x20, 0x1F }; +static hda_nid_t vt1718S_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -512,6 +521,67 @@ snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1718S) { + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW0/1 (24h/25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + set_pin_power_state(codec, 0x25, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, + parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } } } @@ -572,11 +642,21 @@ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; - unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_SEL, - 0x00); + hda_nid_t nid; + unsigned int pinsel; + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + /* use !! to translate conn sel 2 for VT1718S */ + pinsel = !!snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, + 0x00); ucontrol->value.enumerated.item[0] = pinsel; return 0; @@ -635,6 +715,16 @@ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ + spec->multiout.num_dacs = 4; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid @@ -645,7 +735,8 @@ update_side_mute_status(codec); /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S - || spec->codec_type == VT1702) { + || spec->codec_type == VT1702 + || spec->codec_type == VT1718S) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -758,7 +849,8 @@ AC_VERB_GET_PIN_WIDGET_CONTROL, 0); if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* ignore FMic for independent HP */ if (ctl & AC_PINCTL_IN_EN && !(ctl & AC_PINCTL_OUT_EN)) @@ -782,7 +874,8 @@ for (i = 0; i < ARRAY_SIZE(index); i++) { hda_nid_t nid = spec->autocfg.input_pins[index[i]]; if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* don't retask FMic for independent HP */ if (nid) { unsigned int parm = snd_hda_codec_read( @@ -797,6 +890,10 @@ mute_aa_path(codec, 1); notify_aa_path_ctls(codec); } + if (spec->codec_type == VT1718S) + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, + HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { if (spec->codec_type == VT1708S) { @@ -871,6 +968,11 @@ start_idx = 1; end_idx = 3; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return 0; } @@ -920,6 +1022,7 @@ parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ break; case VT1708S: + case VT1718S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1026,8 +1129,8 @@ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && - !spec->hp_independent_mode) + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] + && !spec->hp_independent_mode) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); @@ -3837,6 +3940,436 @@ return 0; } +/* Patch for VT1718S */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1718S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1718S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + + /* Setup default input of Front HP to MW9 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW9 PW10 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + /* PW11 Input enable */ + {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf88, 0x8}, + /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Unmute MW4's index 0 */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + + +static struct hda_verb vt1718S_uniwill_init_verbs[] = { + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 10, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1718S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x8; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0xa; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x9; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0xb; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; + hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; + hda_nid_t nid, nid_vol, nid_mute = 0; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0xc; /* AOW4 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 1; + break; + + case 0x2a: /* Line In */ + idx = 2; + break; + + case 0x29: /* Front Mic */ + idx = 3; + break; + + case 0x2c: /* CD */ + idx = 0; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + return 0; +} + +static int vt1718S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + static hda_nid_t vt1718s_ignore[] = {0x2e, 0}; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + vt1718s_ignore); + if (err < 0) + return err; + err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428) + spec->dig_in_nid = 0x13; + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1718S_loopbacks[] = { + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { 0x21, HDA_INPUT, 3 }, + { 0x21, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1718S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1718S Analog"; + spec->stream_analog_playback = &vt1718S_pcm_analog_playback; + spec->stream_analog_capture = &vt1718S_pcm_analog_capture; + + spec->stream_name_digital = "VT1718S Digital"; + spec->stream_digital_playback = &vt1718S_pcm_digital_playback; + if (codec->vendor_id == 0x11060428) + spec->stream_digital_capture = &vt1718S_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1718S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1718S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -3909,6 +4442,10 @@ .patch = patch_vt1702}, { .id = 0x11067398, .name = "VT1702", .patch = patch_vt1702}, + { .id = 0x11060428, .name = "VT1718S", + .patch = patch_vt1718S}, + { .id = 0x11064428, .name = "VT1718S", + .patch = patch_vt1718S}, {} /* terminator */ };

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[alsa-devel] [PATCH 21/30] ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb As init verbs, vt17xx_volume_init_verb is a better place to hold them. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -3084,6 +3084,8 @@ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* Enable Mic Boost Volume backdoor */ {0x1, 0xf98, 0x1}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, { } }; @@ -3543,6 +3545,10 @@ /* PW6 PW7 Output enable */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* mixer enable */ + {0x1, 0xF88, 0x3}, + /* GPIO 0~2 */ + {0x1, 0xF82, 0x3F}, { } }; @@ -3784,8 +3790,6 @@ { struct via_spec *spec; int err; - unsigned int response; - unsigned char control; /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -3830,18 +3834,6 @@ spec->loopback.amplist = vt1702_loopbacks; #endif - /* Open backdoor */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0); - control = (unsigned char)(response & 0xff); - control |= 0x3; - snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control); - - /* Enable GPIO 0&1 for volume&mute control */ - /* Enable GPIO 2 for DMIC-DATA */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0); - control = (unsigned char)((response >> 16) & 0x3f); - snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control); - return 0; }

1 0

[alsa-devel] [PATCH 20/30] ALSA: HDA VIA: Replace MIC_BOOST_VOLUME.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Replace MIC_BOOST_VOLUME. With snd_hda_override_amp_caps. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 72 ++++++++++------------------------------------------ 1 file changed, 15 insertions(+), 57 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -197,46 +197,6 @@ AUTO_SEQ_SIDE }; -/* Some VT1708S based boards gets the micboost setting wrong, so we have - * to apply some brute-force and re-write the TLV's by software. */ -static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *_tlv) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - if (size < 4 * sizeof(unsigned int)) - return -ENOMEM; - if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */ - return -EFAULT; - if (put_user(2 * sizeof(unsigned int), _tlv + 1)) - return -EFAULT; - if (put_user(0, _tlv + 2)) /* offset = 0 */ - return -EFAULT; - if (put_user(1000, _tlv + 3)) /* step size = 10 dB */ - return -EFAULT; - } - return 0; -} - -static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 3; - } - return 0; -} - static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); @@ -3079,29 +3039,15 @@ /* Patch for VT1708S */ -/* VT1708S software backdoor based override for buggy hardware micboost - * setting */ -#define MIC_BOOST_VOLUME(xname, nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ - .info = mic_boost_volume_info, \ - .get = snd_hda_mixer_amp_volume_get, \ - .put = snd_hda_mixer_amp_volume_put, \ - .tlv = { .c = mic_boost_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) } - /* capture mixer elements */ static struct snd_kcontrol_new vt1708S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), - MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A), - MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -3473,6 +3419,16 @@ }; #endif +static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, + int offset, int num_steps, int step_size) +{ + snd_hda_override_amp_caps(codec, pin, HDA_INPUT, + (offset << AC_AMPCAP_OFFSET_SHIFT) | + (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | + (step_size << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + static int patch_vt1708S(struct hda_codec *codec) { struct via_spec *spec; @@ -3509,6 +3465,8 @@ spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; }

1 0

[alsa-devel] [PATCH 19/30] ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Modify vt1709_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -2176,7 +2176,7 @@ { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -2185,43 +2185,45 @@ if (!nid) continue; + nid_vol = nid_vols[i]; + if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT){ - /* add control to mixer index 0 */ + /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -2242,26 +2244,26 @@ } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err;

1 0

[alsa-devel] [PATCH 18/30] ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Modify vt1708_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -1570,7 +1570,7 @@ { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid, nid_vol = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -1579,8 +1579,7 @@ if (!nid) continue; - if (i != AUTO_SEQ_FRONT) - nid_vol = 0x18 + i; + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -1612,13 +1611,13 @@ /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err;

1 0

[alsa-devel] [PATCH 17/30] ALSA: HDA VIA: Add 2nd S/PDIF out for VT1708S and VT1702.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add 2nd S/PDIF out for VT1708S and VT1702. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -107,7 +107,7 @@ /* playback */ struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; + hda_nid_t second_dig_out; /* capture */ unsigned int num_adc_nids; @@ -1179,8 +1179,23 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, format, substream); + hda_nid_t nid = substream->number == 0 + ? spec->multiout.dig_out_nid : spec->second_dig_out; + mutex_lock(&codec->spdif_mutex); + /* copy from hda_codec.c: setup_dig_out_stream */ + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write_cache( + codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); + /* turn on again (if needed) */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write_cache( + codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + mutex_unlock(&codec->spdif_mutex); + return 0; } static int via_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -1298,6 +1313,13 @@ if (err < 0) return err; spec->multiout.share_spdif = 1; + /* add second spdif out */ + if (spec->second_dig_out) { + err = snd_hda_create_spdif_out_ctls( + codec, spec->second_dig_out); + if (err < 0) + return err; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -1472,11 +1494,6 @@ } else /* enable SPDIF-input pin */ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - - /* assign slave outs */ - if (spec->slave_dig_outs[0]) - codec->slave_dig_outs = spec->slave_dig_outs; - return 0; } @@ -3163,7 +3180,7 @@ }; static struct hda_pcm_stream vt1708S_pcm_digital_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 2, /* NID is set in via_build_pcms */ @@ -3399,7 +3416,7 @@ if (!spec->multiout.dig_out_nid) spec->multiout.dig_out_nid = nid; else { - spec->slave_dig_outs[0] = nid; + spec->second_dig_out = nid; break; /* at most two dig outs */ } }

1 0

[alsa-devel] [PATCH 16/30] ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Replace via_playback_pcm_prepare/cleanup Replaced with via_playback_multi_pcm_prepare/cleanup to support multi-stream operations Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 40 +++++++++------------------------------- 1 file changed, 9 insertions(+), 31 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -1022,28 +1022,6 @@ hinfo); } -static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - vt1708_start_hp_work(spec); - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - vt1708_stop_hp_work(spec); - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - - static void playback_multi_pcm_prep_0(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -1252,7 +1230,7 @@ }; static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ @@ -1263,8 +1241,8 @@ .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -2061,8 +2039,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -2073,8 +2051,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -3165,8 +3143,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, .close = via_pcm_open_close }, };

1 0

[alsa-devel] [PATCH 14/30] ALSA: HDA VIA: Add Jack detect feature for VT1708.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add Jack detect feature for VT1708. VT1708 does not support unsolicited response, but we need hp detect to automute speaker. Implemented in workqueue. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 230 +++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 173 insertions(+), 57 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -89,6 +89,64 @@ CODEC_TYPES, }; +struct via_spec { + /* codec parameterization */ + struct snd_kcontrol_new *mixers[4]; + unsigned int num_mixers; + + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; + + char *stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char *stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + hda_nid_t slave_dig_outs[2]; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; + hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* PCM information */ + struct hda_pcm pcm_rec[3]; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + struct snd_array kctls; + struct hda_input_mux private_imux[2]; + hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; + + enum VIA_HDA_CODEC codec_type; + + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work vt1708_hp_work; + int vt1708_jack_detectect; + int vt1708_hp_present; +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif +}; + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -181,6 +239,31 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); +static int is_aa_path_mute(struct hda_codec *codec); + +static void vt1708_start_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + if (!delayed_work_pending(&spec->vt1708_hp_work)) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); +} + +static void vt1708_stop_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 + && !is_aa_path_mute(spec->codec)) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + cancel_delayed_work(&spec->vt1708_hp_work); + flush_scheduled_work(); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -190,6 +273,12 @@ set_jack_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { + if (is_aa_path_mute(codec)) + vt1708_start_hp_work(codec->spec); + else + vt1708_stop_hp_work(codec->spec); + } return change; } @@ -210,59 +299,6 @@ }; -struct via_spec { - /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; - - struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - - char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - - char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; - - /* capture */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t mux_nids[3]; - hda_nid_t dig_in_nid; - hda_nid_t dig_in_pin; - - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; - - /* dynamic controls, init_verbs and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_imux[2]; - hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; - - /* HP mode source */ - const struct hda_input_mux *hp_mux; - unsigned int hp_independent_mode; - unsigned int hp_independent_mode_index; - unsigned int smart51_enabled; - - enum VIA_HDA_CODEC codec_type; - -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_loopback_check loopback; -#endif -}; - static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -981,7 +1017,6 @@ struct via_spec *spec = codec->spec; int idle = substream->pstr->substream_opened == 1 && substream->ref_count == 0; - analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); @@ -994,6 +1029,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_start_hp_work(spec); return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1003,6 +1039,7 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } @@ -1094,7 +1131,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); } - + vt1708_start_hp_work(spec); return 0; } @@ -1134,7 +1171,7 @@ snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); } - + vt1708_stop_hp_work(spec); return 0; } @@ -1345,6 +1382,7 @@ return; via_free_kctls(codec); + vt1708_stop_hp_work(spec); kfree(codec->spec); } @@ -1464,6 +1502,15 @@ return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static int via_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_POWER_SAVE static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -1479,6 +1526,9 @@ .build_pcms = via_build_pcms, .init = via_init, .free = via_free, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = via_suspend, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif @@ -1728,6 +1778,51 @@ return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1755,6 +1850,10 @@ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1788,6 +1887,22 @@ return 0; } +static void vt1708_update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + vt1708_hp_work.work); + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (spec->vt1708_hp_present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + spec->vt1708_hp_present ^= 1; + via_hp_automute(spec->codec); + } + vt1708_start_hp_work(spec); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1863,7 +1978,8 @@ #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; }

1 0

[alsa-devel] [PATCH 15/30] ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Modify vt1708_set_pinconfig_connect function. like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port Connectivity field into 'AC_JACK_PORT_COMPLEX' Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -1768,11 +1768,10 @@ def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { - if (seqassoc == 0xff) { - def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_set_pincfg(codec, nid, def_conf); - } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE + && (seqassoc == 0xf0 || seqassoc == 0xff)) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } return;

1 0

[alsa-devel] [PATCH 13/30] ALSA: HDA VIA: Refresh front playback mute in via_hp_automute.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Refresh front playback mute in via_hp_automute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -1351,14 +1351,25 @@ /* mute internal speaker if HP is plugged */ static void via_hp_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = 0; struct via_spec *spec = codec->spec; present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], - HDA_OUTPUT, 0, HDA_AMP_MUTE, - present ? HDA_AMP_MUTE : 0); + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + /* auto mute */ + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Front Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } } static void via_gpio_control(struct hda_codec *codec)

1 0

[alsa-devel] [PATCH 12/30] ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 38 +++++++++++++++++++++++++++++++------- 1 file changed, 31 insertions(+), 7 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -124,6 +124,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 +#define VIA_JACK_EVENT 0x04 enum { VIA_CTL_WIDGET_VOL, @@ -1413,10 +1414,12 @@ unsigned int res) { res >>= 26; - if (res == VIA_HP_EVENT) + if (res & VIA_HP_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) + if (res & VIA_GPIO_EVENT) via_gpio_control(codec); + if (res & VIA_JACK_EVENT) + set_jack_power_state(codec); } static int via_init(struct hda_codec *codec) @@ -1877,7 +1880,8 @@ }; static struct hda_verb vt1709_uniwill_init_verbs[] = { - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } }; @@ -2513,7 +2517,15 @@ }; static struct hda_verb vt1708B_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3008,7 +3020,15 @@ }; static struct hda_verb vt1708S_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3447,8 +3467,12 @@ }; static struct hda_verb vt1702_uniwill_init_verbs[] = { - {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } };

1 0

[alsa-devel] [PATCH 11/30] ALSA: HDA VIA: When changing input source, update power state.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: When changing input source, update power state. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -549,6 +549,14 @@ if (!spec->mux_nids[adc_idx]) return -EINVAL; + /* switch to D0 beofre change index */ + if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) + snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* update jack power state */ + set_jack_power_state(codec); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);

1 0

[alsa-devel] [PATCH 10/30] ALSA: HDA VIA: Add smart5.1 function.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add smart5.1 function. Smart 5.1 is for 3-jacks model, to reuse input pins as outputs. While off, they act as "line out" / "line in" / "mic in". While on, they acts as "line out" / "back left/right" / "center/lfe". Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 177 ++++++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 173 insertions(+), 4 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -211,7 +211,7 @@ struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; + struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -253,6 +253,7 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; @@ -390,6 +391,8 @@ } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -400,9 +403,10 @@ & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ unsigned present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) >> 31; - - if ((no_presence || present) && get_defcfg_connect(def_conf) - != AC_JACK_PORT_NONE) { + struct via_spec *spec = codec->spec; + if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) + || ((no_presence || present) + && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { *affected_parm = AC_PWRST_D0; /* if it's connected */ parm = AC_PWRST_D0; } else @@ -657,6 +661,167 @@ { } /* end */ }; +static void notify_aa_path_ctls(struct hda_codec *codec) +{ + int i; + struct snd_ctl_elem_id id; + const char *labels[] = {"Mic", "Front Mic", "Line"}; + + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + for (i = 0; i < ARRAY_SIZE(labels); i++) { + sprintf(id.name, "%s Playback Volume", labels[i]); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +static void mute_aa_path(struct hda_codec *codec, int mute) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708: + nid_mixer = 0x17; + start_idx = 2; + end_idx = 4; + break; + case VT1709_10CH: + case VT1709_6CH: + nid_mixer = 0x18; + start_idx = 2; + end_idx = 4; + break; + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + default: + return; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE; + snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i, + HDA_AMP_MUTE, val); + } +} +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) +{ + int res = 0; + int index; + for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { + if (pin == spec->autocfg.input_pins[index]) { + res = 1; + break; + } + } + return res; +} + +static int via_smart51_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int via_smart51_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int on = 1; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + int ctl = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* ignore FMic for independent HP */ + if (ctl & AC_PINCTL_IN_EN + && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; + } + } + *ucontrol->value.integer.value = on; + return 0; +} + +static int via_smart51_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int out_in = *ucontrol->value.integer.value + ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* don't retask FMic for independent HP */ + if (nid) { + unsigned int parm = snd_hda_codec_read( + codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + } + if (i == AUTO_PIN_FRONT_MIC) { + if (spec->codec_type == VT1708S) { + /* input = index 1 (AOW3) */ + snd_hda_codec_write( + codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, 1); + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, + 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); + } + } + } + spec->smart51_enabled = *ucontrol->value.integer.value; + set_jack_power_state(codec); + return 1; +} + +static struct snd_kcontrol_new via_smart51_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Smart 5.1", + .count = 1, + .info = via_smart51_info, + .get = via_smart51_get, + .put = via_smart51_put, + }, + {} /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1587,6 +1752,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2086,6 +2252,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2648,6 +2815,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -3141,6 +3309,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; }

1 0

[alsa-devel] [PATCH 9/30] ALSA: HDA VIA: Rewrite via_independent_hp_put
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Rewrite via_independent_hp_put Use hp_independent_mode_index to store hp index, and simplify function via_independent_hp_put with it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 87 ++++++++++++++++++++++++++++------------------------ 1 file changed, 47 insertions(+), 40 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -252,6 +252,7 @@ /* HP mode source */ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; enum VIA_HDA_CODEC codec_type; @@ -584,6 +585,36 @@ } } +static int update_side_mute_status(struct hda_codec *codec) +{ + /* mute side channel */ + struct via_spec *spec = codec->spec; + unsigned int parm = spec->hp_independent_mode + ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + hda_nid_t sw3; + + switch (spec->codec_type) { + case VT1708: + sw3 = 0x1b; + break; + case VT1709_10CH: + sw3 = 0x29; + break; + case VT1708B_8CH: + case VT1708S: + sw3 = 0x27; + break; + default: + sw3 = 0; + break; + } + + if (sw3) + snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, + parm); + return 0; +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -591,47 +622,18 @@ struct via_spec *spec = codec->spec; hda_nid_t nid = spec->autocfg.hp_pins[0]; unsigned int pinsel = ucontrol->value.enumerated.item[0]; - unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - - if (con_nid == spec->multiout.hp_nid) { - if (pinsel == 0) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } else if (pinsel == 1) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } - } else { - if (pinsel == 0) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } else if (pinsel == 1) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - pinsel); - - if (spec->multiout.hp_nid && - spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) - snd_hda_codec_setup_stream(codec, - spec->multiout.hp_nid, - 0, 0, 0); + /* Get Independent Mode index of headphone pin widget */ + spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel + ? 1 : 0; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + + if (spec->multiout.hp_nid && spec->multiout.hp_nid + != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, + 0, 0, 0); + update_side_mute_status(codec); /* update HP volume/swtich active state */ if (spec->codec_type == VT1708S || spec->codec_type == VT1702) { @@ -1447,6 +1449,7 @@ return 0; spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1981,6 +1984,7 @@ spec->multiout.hp_nid = VT1709_HP_DAC_NID; else if (spec->multiout.num_dacs == 3) /* 6 channels */ spec->multiout.hp_nid = 0; + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2540,6 +2544,7 @@ return 0; spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3010,6 +3015,7 @@ return 0; spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3367,6 +3373,7 @@ if (!pin) return 0; spec->multiout.hp_nid = 0x1D; + spec->hp_independent_mode_index = 0; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume",

1 0

[alsa-devel] [PATCH 8/30] ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls For VT1708S and VT1702, deactivate "Headphone Playback Volume" and "Headphone Playback Mute" control if "Independent HP" mode is OFF. and rename VT1702 "Independent HP" text. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 38 ++++++++++++++++++++++++++++++++++---- 1 file changed, 34 insertions(+), 4 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -572,6 +572,18 @@ return 0; } +static void activate_ctl(struct hda_codec *codec, const char *name, int active) +{ + struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); + if (ctl) { + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access |= active + ? 0 : SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id); + } +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -620,6 +632,14 @@ spec->multiout.hp_nid, 0, 0, 0); + /* update HP volume/swtich active state */ + if (spec->codec_type == VT1708S + || spec->codec_type == VT1702) { + activate_ctl(codec, "Headphone Playback Volume", + spec->hp_independent_mode); + activate_ctl(codec, "Headphone Playback Switch", + spec->hp_independent_mode); + } return 0; } @@ -3341,11 +3361,11 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { - int err; - + int err, i; + struct hda_input_mux *imux; + static const char *texts[] = { "ON", "OFF", NULL}; if (!pin) return 0; - spec->multiout.hp_nid = 0x1D; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -3360,8 +3380,18 @@ if (err < 0) return err; - create_hp_imux(spec); + imux = &spec->private_imux[1]; + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; return 0; }

1 0

[alsa-devel] [PATCH 7/30] ALSA: HDA VIA: Remove unused argument of via_new_analog_input
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove unused argument of via_new_analog_input Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -317,8 +317,8 @@ } /* create input playback/capture controls for the given pin */ -static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, - const char *ctlname, int idx, int mix_nid) +static int via_new_analog_input(struct via_spec *spec, const char *ctlname, + int idx, int mix_nid) { char name[32]; int err; @@ -1480,8 +1480,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x17); + err = via_new_analog_input(spec, labels[i], idx, 0x17); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2013,8 +2012,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x18); + err = via_new_analog_input(spec, labels[i], idx, 0x18); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2575,8 +2573,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3047,8 +3044,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3401,8 +3397,7 @@ idx = 3; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], - labels[i], idx, 0x1A); + err = via_new_analog_input(spec, labels[i], idx, 0x1A); if (err < 0) return err; imux->items[imux->num_items].label = labels[i];

1 0

[alsa-devel] [PATCH 6/30] ALSA: HDA VIA: Add low current mode for power saving.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add low current mode for power saving. For VT1708B, VT1708S and VT1702, enter low current mode if no analog stream is opened and all aa path mute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 41 +++++++++++++++++++++++++++++++++++------ 1 file changed, 35 insertions(+), 6 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -783,6 +783,10 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -1089,6 +1093,11 @@ if (err < 0) return err; } + + /* init power states */ + set_jack_power_state(codec); + analog_low_current_mode(codec, 1); + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -2311,6 +2320,17 @@ { } }; +static int via_pcm_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); + return 0; +} + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -2319,7 +2339,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2341,8 +2362,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2799,7 +2822,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .cleanup = via_playback_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2809,8 +2833,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3235,7 +3261,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3245,8 +3272,10 @@ .channels_max = 2, .nid = 0x12, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, };

1 0

[alsa-devel] [PATCH 5/30] ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 240 +++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 239 insertions(+), 1 deletion(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -128,6 +128,7 @@ enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, + VIA_CTL_WIDGET_ANALOG_MUTE, }; enum { @@ -177,9 +178,34 @@ return 0; } +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + return change; +} + +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, }; @@ -303,7 +329,7 @@ if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -362,6 +388,131 @@ } } + +static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int *affected_parm) +{ + unsigned parm; + unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); + unsigned no_presence = (def_conf & AC_DEFCFG_MISC) + >> AC_DEFCFG_MISC_SHIFT + & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ + unsigned present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31; + + if ((no_presence || present) && get_defcfg_connect(def_conf) + != AC_JACK_PORT_NONE) { + *affected_parm = AC_PWRST_D0; /* if it's connected */ + parm = AC_PWRST_D0; + } else + parm = AC_PWRST_D3; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + +static void set_jack_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + + if (spec->codec_type == VT1702) { + imux_is_smixer = snd_hda_codec_read( + codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x16, &parm); + set_pin_power_state(codec, 0x17, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, + parm); + } else if (spec->codec_type == VT1708B_8CH + || spec->codec_type == VT1708B_4CH + || spec->codec_type == VT1708S) { + /* SW0 (17h) = stereo mixer */ + int is_8ch = spec->codec_type != VT1708B_4CH; + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } +} + /* * input MUX handling */ @@ -504,6 +655,93 @@ }, { } /* end */ }; + +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) +{ + struct via_spec *spec = codec->spec; + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; + case VT1708S: + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ + break; + case VT1702: + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ + break; + default: + return; /* other codecs are not supported */ + } + /* send verb */ + snd_hda_codec_write(codec, codec->afg, 0, verb, parm); +} + /* * generic initialization of ADC, input mixers and output mixers */

1 0

[alsa-devel] [PATCH 4/30] ALSA: HDA VIA: Limit VT1702 AA-Path max volume
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Limit VT1702 AA-Path max volume according to customer request, VT1702 AA-Path max volume (12 dB) is too high, so limit to 0 dB. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 6 ++++++ 1 file changed, 6 insertions(+) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -3165,6 +3165,12 @@ err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; + /* limit AA path volume to 0 dB */ + snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err;

1 0

[alsa-devel] [PATCH 3/30] ALSA: HDA VIA: Add VT1708B-CE codec support.
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Add VT1708B-CE codec support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -84,6 +84,7 @@ VT1708B_8CH, VT1708B_4CH, VT1708S, + VT1708BCE, VT1702, CODEC_TYPES, }; @@ -104,9 +105,11 @@ codec_type = VT1709_10CH; else if (dev_id >= 0xe714 && dev_id <= 0xe717) codec_type = VT1709_6CH; - else if (dev_id >= 0xe720 && dev_id <= 0xe723) + else if (dev_id >= 0xe720 && dev_id <= 0xe723) { codec_type = VT1708B_8CH; - else if (dev_id >= 0xe724 && dev_id <= 0xe727) + if (snd_hda_param_read(codec, 0x16, AC_PAR_CONNLIST_LEN) == 0x7) + codec_type = VT1708BCE; + } else if (dev_id >= 0xe724 && dev_id <= 0xe727) codec_type = VT1708B_4CH; else if ((dev_id & 0xfff) == 0x397 && (dev_id >> 12) < 8) @@ -224,6 +227,8 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + enum VIA_HDA_CODEC codec_type; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -979,6 +984,10 @@ for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + spec->codec_type = get_codec_type(codec); + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost + same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { @@ -2368,12 +2377,14 @@ { } /* end */ }; #endif - +static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { struct via_spec *spec; int err; + if (get_codec_type(codec) == VT1708BCE) + return patch_vt1708S(codec); /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2905,6 +2916,16 @@ spec->loopback.amplist = vt1708S_loopbacks; #endif + /* correct names for VT1708BCE */ + if (get_codec_type(codec) == VT1708BCE) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + spec->stream_name_analog = "VT1708BCE Analog"; + spec->stream_name_digital = "VT1708BCE Digital"; + } return 0; }

1 0

[alsa-devel] [PATCH 2/30] HDA VIA: Change get_codec_type argument to hda_codec type
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Change get_codec_type argument to hda_codec type Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -88,8 +88,9 @@ CODEC_TYPES, }; -static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { + u32 vendor_id = codec->vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -141,7 +142,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { if (size < 4 * sizeof(unsigned int)) return -ENOMEM; @@ -163,7 +164,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2;

1 0

[alsa-devel] [PATCH 1/30] ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro
by LoganLi＠viatech.com.cn 09 Oct '09

09 Oct '09

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove unused IS_VT17xx_VENDORID macro IS_VT17*_VENDORID macros are used nowhere, so clean them up. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- pci/hda/patch_via.c | 8 -------- 1 file changed, 8 deletions(-) --- a/pci/hda/patch_via.c +++ b/pci/hda/patch_via.c @@ -76,14 +76,6 @@ #define VT1702_HP_NID 0x17 #define VT1702_DIGOUT_NID 0x11 -#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) -#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) -#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) -#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) -#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) -#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) -#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) - enum VIA_HDA_CODEC { UNKNOWN = -1, VT1708,

1 0

[alsa-devel] [PATCH 1/31] HDA patch_via.c: Remove IS_VT17xx_VENDORID macro
by Li Bo 09 Oct '09

09 Oct '09

[ALSA] HDA VIA: Remove IS_VT17xx_VENDORID macro Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-09-29 18:01:12.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-09-29 18:01:16.000000000 +0800 @@ -76,14 +76,6 @@ #define VT1702_HP_NID 0x17 #define VT1702_DIGOUT_NID 0x11 -#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) -#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) -#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) -#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) -#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) -#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) -#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) - enum VIA_HDA_CODEC { UNKNOWN = -1, VT1708,

3 13

Re: [alsa-devel] sscape: make more code __devinit and other improvements
by krzysztof.h1＠poczta.fm 08 Oct '09

08 Oct '09

"Takashi Iwai" <tiwai(a)suse.de> pisze: > At Wed, 7 Oct 2009 18:45:37 +0200, > Krzysztof Helt wrote: > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Make code to upload firmware marked as __devinit. > > I guess the mark would be removed again if you implement suspend/resume > :) > If you are not going to implement PM, I'm willing to apply this patch, > though. > I haven't thought about this. Drop the patch then. > > General code and coding style improvements. > > > > Add a missing argument for CMD_XXX_MIDI_VOL command. > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > --- > > This is the last my patch for the sscape driver. > > > > I will examine if the OSS aedsp16 driver can be removed as well. The > corresponding > > ALSA driver is snd-sc6000. > > Yeah, that'll be nice. > > I have a strange problem. The sscape and aedsp16 drivers do not work correctly with OSS drivers (DMA/IRQ problem) but they work correctly with ALSA + OSS emulation. I will check more cards (e.g. SB16). It may a new bug in the OSS layer. Regards, Krzysztof ---------------------------------------------------------------------- Zobacz najwiekszy samolot na swiecie! Kliknij >>> http://link.interia.pl/f238f

2 1

[alsa-devel] ascenario: Second batch
by Stefan Schmidt 08 Oct '09

08 Oct '09

Hello. Two more patches for ascenario. To be applied on top of the batch from yesterday. [PATCH 1/2] ascenario: Bring back a bugfix and config token change from latest ascenario That one got missed as Jaroslav applied the initial ascenario patch. There was a later one which added this changes. [PATCH 2/2] ascenario: Reduce SND_SCN_* defines to the basics. I still need feedback here how you guys want to handle the defines. Could be seen as an RFC if you like. For the better part of the day I'm looking into some bug regarding the control_id fucntion. Hopefully another fix I can submit tomorrow. regards Stefan Schmidt

1 2

[alsa-devel] ALSA: ctxfi: Swapped SURROUND-SIDE mute
by Frank Roth 08 Oct '09

08 Oct '09

Hello, I can confirm, that this patch is working fine with 6 Speakers. Mute-/unmute and volume control of all 6 channels are working fine. I used a Soundblaster X-Fi Titanium (EMU20k2) Best regards, Frank Roth

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 08 Oct '09

08 Oct '09

The following changes since commit 2a0f5cb32772e9a9560209e241a80bfbbc31dbc3: Mark Brown (1): Merge branch 'for-2.6.32' of git://git.kernel.org/.../broonie/sound-2.6 into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (3): ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSI ASoC: WM8350 capture PGA mutes are inverted Merge branch 'upstream/wm8350' into for-2.6.32 sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/imx/mxc-ssi.c | 8 -------- 2 files changed, 2 insertions(+), 10 deletions(-)

2 1

[alsa-devel] [PATCH] ice1724: increase SPDIF and independent stereo buffer sizes
by Robert Hancock 08 Oct '09

08 Oct '09

Increase the default and maximum PCM buffer prellocation size for ice1724's SPDIF and independent stereo pair outputs to 256K, which is the hardware's maximum supported size. This allows a reduction in interrupt rate and potentially power usage when an application is not latency-critical. Signed-off-by: Robert Hancock <hancockrwd(a)gmail.com> diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index af6e001..3ae4407 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm = pcm; @@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm_ds = pcm;

2 1

[alsa-devel] [PATCH] opl3: circular locking in the snd_opl3_note_on() and snd_opl3_note_off()
by Krzysztof Helt 08 Oct '09

08 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix following circular locking in the opl3 driver. ======================================================= [ INFO: possible circular locking dependency detected ] 2.6.32-rc3 #87 ------------------------------------------------------- swapper/0 is trying to acquire lock: (&opl3->voice_lock){..-...}, at: [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] but task is already holding lock: (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth] which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&opl3->sys_timer_lock){..-...}: [<c02461d5>] validate_chain+0xa25/0x1040 [<c0246aca>] __lock_acquire+0x2da/0xab0 [<c024731a>] lock_acquire+0x7a/0xa0 [<c044c300>] _spin_lock_irqsave+0x40/0x60 [<cca75046>] snd_opl3_note_on+0x686/0x790 [snd_opl3_synth] [<cca68912>] snd_midi_process_event+0x322/0x590 [snd_seq_midi_emul] [<cca74245>] snd_opl3_synth_event_input+0x15/0x20 [snd_opl3_synth] [<cca4dcc0>] snd_seq_deliver_single_event+0x100/0x200 [snd_seq] [<cca4de07>] snd_seq_deliver_event+0x47/0x1f0 [snd_seq] [<cca4e50b>] snd_seq_dispatch_event+0x3b/0x140 [snd_seq] [<cca5008c>] snd_seq_check_queue+0x10c/0x120 [snd_seq] [<cca5037b>] snd_seq_enqueue_event+0x6b/0xe0 [snd_seq] [<cca4e0fd>] snd_seq_client_enqueue_event+0xdd/0x100 [snd_seq] [<cca4eb7a>] snd_seq_write+0xea/0x190 [snd_seq] [<c02827b6>] vfs_write+0x96/0x160 [<c0282c9d>] sys_write+0x3d/0x70 [<c0202c45>] syscall_call+0x7/0xb -> #0 (&opl3->voice_lock){..-...}: [<c02467e6>] validate_chain+0x1036/0x1040 [<c0246aca>] __lock_acquire+0x2da/0xab0 [<c024731a>] lock_acquire+0x7a/0xa0 [<c044c300>] _spin_lock_irqsave+0x40/0x60 [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth] [<c022ac46>] run_timer_softirq+0x166/0x1e0 [<c02269e8>] __do_softirq+0x78/0x110 [<c0226ac6>] do_softirq+0x46/0x50 [<c0226e26>] irq_exit+0x36/0x40 [<c0204bd2>] do_IRQ+0x42/0xb0 [<c020328e>] common_interrupt+0x2e/0x40 [<c021092f>] apm_cpu_idle+0x10f/0x290 [<c0201b11>] cpu_idle+0x21/0x40 [<c04443cd>] rest_init+0x4d/0x60 [<c055c835>] start_kernel+0x235/0x280 [<c055c066>] i386_start_kernel+0x66/0x70 other info that might help us debug this: 2 locks held by swapper/0: #0: (&opl3->tlist){+.-...}, at: [<c022abd0>] run_timer_softirq+0xf0/0x1e0 #1: (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth] stack backtrace: Pid: 0, comm: swapper Not tainted 2.6.32-rc3 #87 Call Trace: [<c0245188>] print_circular_bug+0xc8/0xd0 [<c02467e6>] validate_chain+0x1036/0x1040 [<c0247f14>] ? check_usage_forwards+0x54/0xd0 [<c0246aca>] __lock_acquire+0x2da/0xab0 [<c024731a>] lock_acquire+0x7a/0xa0 [<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<c044c300>] _spin_lock_irqsave+0x40/0x60 [<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<c044c307>] ? _spin_lock_irqsave+0x47/0x60 [<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth] [<c022ac46>] run_timer_softirq+0x166/0x1e0 [<c022abd0>] ? run_timer_softirq+0xf0/0x1e0 [<cca75150>] ? snd_opl3_timer_func+0x0/0xc0 [snd_opl3_synth] [<c02269e8>] __do_softirq+0x78/0x110 [<c044c0fd>] ? _spin_unlock+0x1d/0x20 [<c025915f>] ? handle_level_irq+0xaf/0xe0 [<c0226ac6>] do_softirq+0x46/0x50 [<c0226e26>] irq_exit+0x36/0x40 [<c0204bd2>] do_IRQ+0x42/0xb0 [<c024463c>] ? trace_hardirqs_on_caller+0x12c/0x180 [<c020328e>] common_interrupt+0x2e/0x40 [<c0208d88>] ? default_idle+0x38/0x50 [<c021092f>] apm_cpu_idle+0x10f/0x290 [<c0201b11>] cpu_idle+0x21/0x40 [<c04443cd>] rest_init+0x4d/0x60 [<c055c835>] start_kernel+0x235/0x280 [<c055c210>] ? unknown_bootoption+0x0/0x210 [<c055c066>] i386_start_kernel+0x66/0x70 Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I am not sure if this is the only possible way to fix my the problem. My idea is to change locking in the snd_opl3_timer_func() so the opl3 voices' part is locked with voice_lock and the timer's part is locked with sys_timer_lock. diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 6e7d09a..1e76f7b 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; extern int use_internal_drums; +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan); /* * The next table looks magical, but it certainly is not. Its values have * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception @@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data) int again = 0; int i; - spin_lock_irqsave(&opl3->sys_timer_lock, flags); + spin_lock_irqsave(&opl3->voice_lock, flags); for (i = 0; i < opl3->max_voices; i++) { struct snd_opl3_voice *vp = &opl3->voices[i]; if (vp->state > 0 && vp->note_off_check) { if (vp->note_off == jiffies) - snd_opl3_note_off(opl3, vp->note, 0, vp->chan); + snd_opl3_note_off_unsafe(opl3, vp->note, 0, + vp->chan); else again++; } } + spin_unlock_irqrestore(&opl3->voice_lock, flags); + + spin_lock_irqsave(&opl3->sys_timer_lock, flags); if (again) { opl3->tlist.expires = jiffies + 1; /* invoke again */ add_timer(&opl3->tlist); @@ -658,28 +664,22 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* * Release a note in response to a midi note off. */ -void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan) +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - int voice; struct snd_opl3_voice *vp; - unsigned long flags; - - opl3 = p; + struct snd_opl3 *opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Note off, ch %i, inst %i, note %i\n", chan->number, chan->midi_program, note); #endif - spin_lock_irqsave(&opl3->voice_lock, flags); - if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel && use_internal_drums) { snd_opl3_drum_switch(opl3, note, vel, 0, chan); - spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } /* this loop will hopefully kill all extra voices, because @@ -697,6 +697,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan snd_opl3_kill_voice(opl3, voice); } } +} + +void snd_opl3_note_off(void *p, int note, int vel, + struct snd_midi_channel *chan) +{ + struct snd_opl3 *opl3 = p; + unsigned long flags; + + spin_lock_irqsave(&opl3->voice_lock, flags); + snd_opl3_note_off_unsafe(p, note, vel, chan); spin_unlock_irqrestore(&opl3->voice_lock, flags); } @@ -867,3 +877,4 @@ void snd_opl3_sysex(void *p, unsigned char *buf, int len, snd_printk(KERN_DEBUG "SYSEX\n"); #endif } + ---------------------------------------------------------------------- Nie wiesz ktory bank wybrac? 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2 1

[alsa-devel] sscape: make more code __devinit and other improvements
by Krzysztof Helt 08 Oct '09

08 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Make code to upload firmware marked as __devinit. General code and coding style improvements. Add a missing argument for CMD_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the last my patch for the sscape driver. I will examine if the OSS aedsp16 driver can be removed as well. The corresponding ALSA driver is snd-sc6000. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be50..06aa116 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer __devinit *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -184,19 +186,19 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo /* * Release the DMA-able kernel memory ... */ -static void free_dmabuf(struct snd_dma_buffer *buf) +static void __devinit free_dmabuf(struct snd_dma_buffer *buf) { if (buf && buf->area) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void __devinit sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -260,7 +263,7 @@ static inline int host_read_unsafe(unsigned io_base) * a limited amount of busy-waiting if the register isn't ready. * Also leaves all locking-issues to the caller ... */ -static int host_read_ctrl_unsafe(unsigned io_base, unsigned timeout) +static int __devinit host_read_ctrl_unsafe(unsigned io_base, unsigned timeout) { int data; @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void __devinit initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -329,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) * The AD1845 detection fails if we *don't* do this, so I * think that this is a good idea ... */ -static inline void activate_ad1845_unsafe(unsigned io_base) +static void __devinit activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -350,24 +354,27 @@ static void soundscape_free(struct snd_card *c) * Tell the SoundScape to begin a DMA tranfer using the given channel. * All locking issues are left to the caller. */ -static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) +static __devinit void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int __devinit sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -377,7 +384,7 @@ static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned ti * This in turn means that we must not be holding any * spinlocks when we call this function. */ -static int obp_startup_ack(struct soundscape *s, unsigned timeout) +static int __devinit obp_startup_ack(struct soundscape *s, unsigned timeout) { unsigned long end_time = jiffies + msecs_to_jiffies(timeout); @@ -404,7 +411,7 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) * that we must not be holding any spinlocks when we call * this function. */ -static int host_startup_ack(struct soundscape *s, unsigned timeout) +static int __devinit host_startup_ack(struct soundscape *s, unsigned timeout) { unsigned long end_time = jiffies + msecs_to_jiffies(timeout); @@ -427,13 +434,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) /* * Upload a byte-stream into the SoundScape using DMA channel A. */ -static int upload_dma_data(struct soundscape *s, - const unsigned char *data, - size_t size) +static int __devinit upload_dma_data(struct soundscape *s, + const unsigned char *data, + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +451,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +483,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +500,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -523,7 +535,7 @@ _release_dma: * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. */ -static int sscape_upload_bootblock(struct snd_card *card) +static int __devinit sscape_upload_bootblock(struct snd_card *card) { struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; @@ -562,7 +574,7 @@ static int sscape_upload_bootblock(struct snd_card *card) /* * Upload the microcode into the SoundScape. */ -static int sscape_upload_microcode(struct snd_card *card, int version) +static int __devinit sscape_upload_microcode(struct snd_card *card, int version) { struct soundscape *sscape = get_card_soundscape(card); const struct firmware *init_fw = NULL; @@ -591,7 +603,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +613,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +627,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +651,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +753,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +777,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +793,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +804,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +818,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +881,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +930,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1034,7 +1036,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1055,6 +1056,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1094,11 +1099,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* * Initialize mixer */ spin_lock_irqsave(&sscape->lock, flags); @@ -1155,7 +1155,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1184,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1238,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1295,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } --------------------------------------------------------------- Zapytaj wr�k�! Kliknij >>> http://link.interia.pl/f238d

2 1

[alsa-devel] [PATCH 31/31] HDA patch_via.c: Only cosmetic changes
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Only cosmetic changes Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:40.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:42.000000000 +0800 @@ -22,26 +22,26 @@ */ /* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */ -/* */ +/* */ /* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */ -/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ -/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ +/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ +/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ -/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ -/* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ /* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ -/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ -/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ -/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ -/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ /* 2009-02-16 Logan Li Add support for VT1718S */ /* 2009-03-13 Logan Li Add support for VT1716S */ /* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ /* 2009-07-08 Lydia Wang Add support for VT2002P */ /* 2009-07-21 Lydia Wang Add support for VT1812 */ -/* */ +/* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -455,7 +455,7 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -1509,7 +1509,7 @@ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* Setup default input MW0 to PW4 */ {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ @@ -1841,8 +1841,10 @@ codec->pcm_info = info; info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -2097,7 +2099,7 @@ if (spec->slave_dig_outs[0]) codec->slave_dig_outs = spec->slave_dig_outs; - return 0; + return 0; } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -2130,8 +2132,8 @@ spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -2169,7 +2171,7 @@ if (!nid) continue; - + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { @@ -2198,7 +2200,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2212,7 +2214,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2312,7 +2314,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -2548,7 +2550,7 @@ "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -2560,7 +2562,7 @@ spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -2749,11 +2751,11 @@ spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ - switch(i) { + switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ spec->multiout.dac_nids[i] = 0x10; @@ -2788,7 +2790,7 @@ for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; nid_vol = nid_vols[i]; @@ -2819,7 +2821,7 @@ HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", @@ -2833,7 +2835,7 @@ HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -2929,7 +2931,7 @@ case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -3038,7 +3040,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -3132,7 +3134,7 @@ spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);

1 1

[alsa-devel] [PATCH 30/31] HDA patch_via.c: comments: update copyright, code description, changeset
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: comments: update copyright, code description, changeset Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:14:16.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:14:19.000000000 +0800 @@ -1,10 +1,10 @@ /* * Universal Interface for Intel High Definition Audio Codec * - * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec + * HD audio interface patch for VIA VT17xx/VT20xx codec * - * Copyright (c) 2006-2008 Lydia Wang <lydiawang(a)viatech.com> - * Takashi Iwai <tiwai(a)suse.de> + * (C) 2006-2009 VIA Technology, Inc. + * (C) 2006-2008 Takashi Iwai <tiwai(a)suse.de> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -36,6 +36,11 @@ /* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ /* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2009-02-16 Logan Li Add support for VT1718S */ +/* 2009-03-13 Logan Li Add support for VT1716S */ +/* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ +/* 2009-07-08 Lydia Wang Add support for VT2002P */ +/* 2009-07-21 Lydia Wang Add support for VT1812 */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */

1 1

[alsa-devel] [PATCH 29/31] HDA patch_via.c: Change PW4 connect select default to to MW0.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Change PW4 connect select default to to MW0. According to customer request, hp should be default to redirected mode, i.e. PW4 connect select default to to MW0. , i.e. PW4 should be default to MW0. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:14:10.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:14:16.000000000 +0800 @@ -1505,8 +1505,8 @@ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Setup default input to PW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Setup default input MW0 to PW4 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* power down jack detect function */ @@ -2641,8 +2641,8 @@ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set input of PW4 as AOW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Set input of PW4 as MW0 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -3196,7 +3196,7 @@ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Setup default input to PW4 */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* PW10 Input enable */

1 1

[alsa-devel] [PATCH 28/31] HDA patch_via.c: vt1708_control_templates[] -> via_control_templates[].
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: vt1708_control_templates[] -> via_control_templates[]. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:14:04.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:14:10.000000000 +0800 @@ -336,7 +336,7 @@ .put = bind_pin_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -static struct snd_kcontrol_new vt1708_control_templates[] = { +static struct snd_kcontrol_new via_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, @@ -399,7 +399,7 @@ knew = snd_array_new(&spec->kctls); if (!knew) return -ENOMEM; - *knew = vt1708_control_templates[type]; + *knew = via_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM;

1 1

[alsa-devel] [PATCH 27/31] HDA patch_via.c: Add VT1812 support.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VT1812 support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:30.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:32.000000000 +0800 @@ -89,6 +89,7 @@ VT1718S, VT1716S, VT2002P, + VT1812, CODEC_TYPES, }; @@ -129,6 +130,8 @@ codec_type = VT1718S; else if (dev_id == 0x0438 || dev_id == 0x4438) codec_type = VT2002P; + else if (dev_id == 0x0448) + codec_type = VT1812; else codec_type = UNKNOWN; return codec_type; @@ -380,6 +383,12 @@ 0x10, 0x11 }; +static hda_nid_t vt1812_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -541,6 +550,7 @@ end_idx = 3; break; case VT2002P: + case VT1812: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -932,6 +942,120 @@ snd_hda_codec_write( codec, 0x21, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else if (spec->codec_type == VT1812) { + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_codec_read( + codec, 0x28, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -1004,7 +1128,7 @@ codec, mux1, 0, AC_VERB_GET_CONNECT_SEL, 0x00); pinsel = conn_sel == 0 ? 0 : 1; } - if (spec->codec_type == VT2002P) { + if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { hda_nid_t mux1 = 0x35; pinsel = snd_hda_codec_read( codec, mux1, 0, AC_VERB_GET_CONNECT_SEL, 0); @@ -1015,7 +1139,8 @@ || spec->codec_type == VT1702 || spec->codec_type == VT1718S || spec->codec_type == VT1716S - || spec->codec_type == VT2002P) + || spec->codec_type == VT2002P + || spec->codec_type == VT1812) && hp_ctl_initialized == -1) { struct snd_card *card = codec->bus->card; struct snd_kcontrol *kctl; @@ -1050,6 +1175,7 @@ [VT1718S] = 0, [VT1716S] = 0, [VT2002P] = 0, + [VT1812] = 0, }; if (!nids_sw3[spec->codec_type]) @@ -1083,9 +1209,13 @@ hda_nid_t mux1 = 0x35; snd_hda_codec_write(codec, mux1, 0, AC_VERB_SET_CONNECT_SEL, pinsel); - } - snd_hda_codec_write(codec, nid, 0, + } else if (spec->codec_type == VT1812) { + /* mux15: index 0: AOW0, index 1: AOW1 */ + hda_nid_t mux1 = 0x3d; + snd_hda_codec_write(codec, mux1, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -1335,6 +1465,7 @@ parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; case VT2002P: + case VT1812: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -1860,7 +1991,7 @@ struct via_spec *spec = codec->spec; char name[32] = "Speaker Playback Switch"; - if (spec->codec_type != VT2002P) + if (spec->codec_type != VT2002P && spec->codec_type != VT1812) return; hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, @@ -2335,7 +2466,7 @@ via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P) { + if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5635,6 +5766,361 @@ return 0; } + +/* for vt1812 */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1812_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1812_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/13/15 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0xa8}, + { } +}; + + +static struct hda_verb vt1812_uniwill_init_verbs[] = { + {0x33, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT }, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1812_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1812_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + + +/* add playback controls from the parsed DAC table */ +static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 6; + imux->num_items++; + + return 0; +} + +static int vt1812_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; +fill_dig_outs(codec); + err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs) + return 0; /* can't find valid BIOS pin config */ + + err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1812_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt1812 */ +static int patch_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1812_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + + spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs; + + spec->stream_name_analog = "VT1812 Analog"; + spec->stream_analog_playback = &vt1812_pcm_analog_playback; + spec->stream_analog_capture = &vt1812_pcm_analog_capture; + + spec->stream_name_digital = "VT1812 Digital"; + spec->stream_digital_playback = &vt1812_pcm_digital_playback; + + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1812_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1812_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1812_loopbacks; +#endif + + return 0; +} + /* * patch entries */ @@ -5721,6 +6207,7 @@ .patch = patch_vt1716S}, { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, {} /* terminator */ };

1 1

[alsa-devel] [PATCH 26/31] HDA patch_via.c: Add VT2002P support.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VT2002P support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:12:39.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:13:44.000000000 +0800 @@ -88,6 +88,7 @@ VT1702, VT1718S, VT1716S, + VT2002P, CODEC_TYPES, }; @@ -126,6 +127,8 @@ codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; + else if (dev_id == 0x0438 || dev_id == 0x4438) + codec_type = VT2002P; else codec_type = UNKNOWN; return codec_type; @@ -135,11 +138,14 @@ #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 #define VIA_MONO_EVENT 0x08 +#define VIA_SPEAKER_EVENT 0x10 +#define VIA_BIND_HP_EVENT 0x20 enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, VIA_CTL_WIDGET_ANALOG_MUTE, + VIA_CTL_WIDGET_BIND_PIN_MUTE, }; enum { @@ -149,36 +155,6 @@ AUTO_SEQ_SIDE }; -static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); -static void set_jack_power_state(struct hda_codec *codec); - -static int analog_input_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - set_jack_power_state(codec); - analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); - return change; -} - -/* modify .put = snd_hda_mixer_amp_switch_put */ -#define ANALOG_INPUT_MUTE \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = NULL, \ - .index = 0, \ - .info = snd_hda_mixer_amp_switch_info, \ - .get = snd_hda_mixer_amp_switch_get, \ - .put = analog_input_switch_put, \ - .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } - -static struct snd_kcontrol_new vt1708_control_templates[] = { - HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), - ANALOG_INPUT_MUTE, -}; - struct via_spec { /* codec parameterization */ @@ -238,6 +214,132 @@ #endif }; +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + return change; +} + +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + +static void via_hp_bind_automute(struct hda_codec *codec); + +static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int i; + int change = 0; + + long *valp = ucontrol->value.integer.value; + int lmute, rmute; + if (strstr(kcontrol->id.name, "Switch") == NULL) { + snd_printd("Invalid control!\n"); + return change; + } + change = snd_hda_mixer_amp_switch_put(kcontrol, + ucontrol); + /* Get mute value */ + lmute = *valp ? 0 : HDA_AMP_MUTE; + valp++; + rmute = *valp ? 0 : HDA_AMP_MUTE; + + /* Set hp pins */ + if (!spec->hp_independent_mode) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + + if (!lmute && !rmute) { + /* Line Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* unmute */ + via_hp_bind_automute(codec); + + } else { + if (lmute) { + /* Mute all left channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + } + if (rmute) { + /* mute all right channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + return change; +} + +#define BIND_PIN_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = bind_pin_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + +static struct snd_kcontrol_new vt1708_control_templates[] = { + HDA_CODEC_VOLUME(NULL, 0, 0, 0), + HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, + BIND_PIN_MUTE, +}; + static hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 @@ -273,6 +375,11 @@ 0x13, 0x14 }; +static hda_nid_t vt2002P_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -355,10 +462,13 @@ { struct via_spec *spec = codec->spec; hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + } } static void via_auto_init_analog_input(struct hda_codec *codec) @@ -401,6 +511,65 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + case VT1716S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; + case VT2002P: + nid_mixer = 0x21; + start_idx = 0; + end_idx = 2; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + static void set_jack_power_state(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -662,6 +831,107 @@ imux_is_smixer ? AC_PWRST_D0 : parm); snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, mono_out ? AC_PWRST_D0 : parm); + } else if (spec->codec_type == VT2002P) { + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, + 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Class-D */ + /* PW0 (24h), MW0(18h), MUX0(34h) */ + present = snd_hda_codec_read( + codec, 0x25, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* Mono Out */ + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + present = snd_hda_codec_read( + codec, 0x26, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); } } @@ -734,13 +1004,18 @@ codec, mux1, 0, AC_VERB_GET_CONNECT_SEL, 0x00); pinsel = conn_sel == 0 ? 0 : 1; } - + if (spec->codec_type == VT2002P) { + hda_nid_t mux1 = 0x35; + pinsel = snd_hda_codec_read( + codec, mux1, 0, AC_VERB_GET_CONNECT_SEL, 0); + } ucontrol->value.enumerated.item[0] = pinsel; /* update HP controls on first time */ if ((spec->codec_type == VT1708S || spec->codec_type == VT1702 || spec->codec_type == VT1718S - || spec->codec_type == VT1716S) + || spec->codec_type == VT1716S + || spec->codec_type == VT2002P) && hp_ctl_initialized == -1) { struct snd_card *card = codec->bus->card; struct snd_kcontrol *kctl; @@ -774,6 +1049,7 @@ [VT1702] = 0, [VT1718S] = 0, [VT1716S] = 0, + [VT2002P] = 0, }; if (!nids_sw3[spec->codec_type]) @@ -802,10 +1078,14 @@ snd_hda_codec_write(codec, mux1, 0, AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); spec->multiout.num_dacs = 4; - } else + } else if (spec->codec_type == VT2002P) { + /* mux1: index 0: AOW0, index 1: AOW1 */ + hda_nid_t mux1 = 0x35; + snd_hda_codec_write(codec, mux1, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); - if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -1021,95 +1301,45 @@ { } /* end */ }; -/* check AA path's mute statue */ -static int is_aa_path_mute(struct hda_codec *codec) +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) { - int mute = 1; - hda_nid_t nid_mixer; - int start_idx; - int end_idx; - int i; struct via_spec *spec = codec->spec; - /* get nid of MW0 and start & end index */ + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ switch (spec->codec_type) { case VT1708B_8CH: case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; case VT1708S: + case VT1718S: case VT1716S: - nid_mixer = 0x16; - start_idx = 2; - end_idx = 4; + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; case VT1702: - nid_mixer = 0x1a; - start_idx = 1; - end_idx = 3; + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ break; - case VT1718S: - nid_mixer = 0x21; - start_idx = 1; - end_idx = 3; + case VT2002P: + verb = 0xf93; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; default: - return 0; - } - /* check AA path's mute status */ - for (i = start_idx; i <= end_idx; i++) { - unsigned int con_list = snd_hda_codec_read( - codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); - int shift = 8 * (i % 4); - hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; - unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); - if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { - /* check mute status while the pin is connected */ - int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, - HDA_INPUT, i) >> 7; - int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, - HDA_INPUT, i) >> 7; - if (!mute_l || !mute_r) { - mute = 0; - break; - } - } - } - return mute; -} - -/* enter/exit analog low-current mode */ -static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) -{ - struct via_spec *spec = codec->spec; - static int saved_stream_idle = 1; /* saved stream idle status */ - int enable = is_aa_path_mute(codec); - unsigned int verb = 0; - unsigned int parm = 0; - - if (stream_idle == -1) /* stream status did not change */ - enable = enable && saved_stream_idle; - else { - enable = enable && stream_idle; - saved_stream_idle = stream_idle; - } - - /* decide low current mode's verb & parameter */ - switch (spec->codec_type) { - case VT1708B_8CH: - case VT1708B_4CH: - verb = 0xf70; - parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ - break; - case VT1708S: - case VT1718S: - case VT1716S: - verb = 0xf73; - parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ - break; - case VT1702: - verb = 0xf73; - parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ - break; - default: - return; /* other codecs are not supported */ + return; /* other codecs are not supported */ } /* send verb */ snd_hda_codec_write(codec, codec->afg, 0, verb, parm); @@ -1623,6 +1853,67 @@ } } +/* mute Internal-Speaker if HP is plugged */ +static void via_speaker_automute(struct hda_codec *codec) +{ + unsigned int hp_present; + struct via_spec *spec = codec->spec; + char name[32] = "Speaker Playback Switch"; + + if (spec->codec_type != VT2002P) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + /* notify change */ + list_for_each_entry(kctl, &card->controls, list) + if (strcmp(kctl->id.name, name) == 0) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctl->id); + } +} + +/* mute line-out and internal speaker if HP is plugged */ +static void via_hp_bind_automute(struct hda_codec *codec) +{ + unsigned int hp_present, present = 0; + struct via_spec *spec = codec->spec; + int i; + + if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0]) + return; + + hp_present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present = snd_hda_codec_read(codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) { + /* Mute Line-Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + if (hp_present) + present = hp_present; + } + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1636,6 +1927,10 @@ set_jack_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); + if (res & VIA_SPEAKER_EVENT) + via_speaker_automute(codec); + if (res & VIA_BIND_HP_EVENT) + via_hp_bind_automute(codec); } static int via_init(struct hda_codec *codec) @@ -2034,10 +2329,19 @@ /* init callback for auto-configuration model -- overriding the default init */ static int via_auto_init(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; + via_init(codec); via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); + if (spec->codec_type == VT2002P) { + via_hp_bind_automute(codec); + } else { + via_hp_automute(codec); + via_speaker_automute(codec); + } + return 0; } @@ -4978,6 +5282,359 @@ return 0; } + +/* for vt2002P */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt2002P_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt2002P_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x37, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3b, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; + + +static struct hda_verb vt2002P_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt2002P_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 3; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 4; + imux->num_items++; + + return 0; +} + +static int vt2002P_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt2002P_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt2002P */ +static int patch_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt2002P_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + + spec->stream_name_analog = "VT2002P Analog"; + spec->stream_analog_playback = &vt2002P_pcm_analog_playback; + spec->stream_analog_capture = &vt2002P_pcm_analog_capture; + + spec->stream_name_digital = "VT2002P Digital"; + spec->stream_digital_playback = &vt2002P_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt2002P_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt2002P_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt2002P_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -5062,6 +5719,8 @@ .patch = patch_vt1716S}, { .id = 0x1106a721, .name = "VT1716S", .patch = patch_vt1716S}, + { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, {} /* terminator */ };

1 1

[alsa-devel] [PATCH 25/31] HDA patch_via.c: Add VT1716S support.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VT1716S support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:26.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:28.000000000 +0800 @@ -87,6 +87,7 @@ VT1708BCE, VT1702, VT1718S, + VT1716S, CODEC_TYPES, }; @@ -121,6 +122,8 @@ else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0433 || dev_id == 0xa721) + codec_type = VT1716S; else if (dev_id == 0x0441 || dev_id == 0x4441) codec_type = VT1718S; else @@ -131,6 +134,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 #define VIA_JACK_EVENT 0x04 +#define VIA_MONO_EVENT 0x08 enum { VIA_CTL_WIDGET_VOL, @@ -178,7 +182,7 @@ struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[4]; + struct snd_kcontrol_new *mixers[6]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -221,6 +225,7 @@ unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; unsigned int smart51_enabled; + unsigned int dmic_enabled; enum VIA_HDA_CODEC codec_type; /* work to check hp jack state */ @@ -263,6 +268,11 @@ 0x10, 0x11 }; +static hda_nid_t vt1716S_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -552,6 +562,106 @@ snd_hda_codec_write(codec, 0xc, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1716S) { + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write( + codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_codec_read( + codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + mono_out = 0; + else { + present = snd_hda_codec_read( + codec, 0x1d, 0, AC_VERB_GET_PIN_SENSE, 0) + & 0x80000000; + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); } } @@ -629,7 +739,8 @@ /* update HP controls on first time */ if ((spec->codec_type == VT1708S || spec->codec_type == VT1702 - || spec->codec_type == VT1718S) + || spec->codec_type == VT1718S + || spec->codec_type == VT1716S) && hp_ctl_initialized == -1) { struct snd_card *card = codec->bus->card; struct snd_kcontrol *kctl; @@ -662,6 +773,7 @@ [VT1708S] = 0x27, [VT1702] = 0, [VT1718S] = 0, + [VT1716S] = 0, }; if (!nids_sw3[spec->codec_type]) @@ -757,6 +869,7 @@ case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -858,7 +971,8 @@ HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { - if (spec->codec_type == VT1708S) { + if (spec->codec_type == VT1708S + || spec->codec_type == VT1716S) { /* input = index 1 (AOW3) */ snd_hda_codec_write( codec, nid, 0, @@ -921,6 +1035,7 @@ case VT1708B_8CH: case VT1708B_4CH: case VT1708S: + case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; @@ -985,6 +1100,7 @@ break; case VT1708S: case VT1718S: + case VT1716S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -1429,6 +1545,36 @@ } } +/* mute mono out if HP or Line out is plugged */ +static void via_mono_automute(struct hda_codec *codec) +{ + unsigned int hp_present, lineout_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1716S) + return; + + lineout_present = snd_hda_codec_read( + codec, spec->autocfg.line_out_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Mute Mono Out if Line Out is plugged */ + if (lineout_present) { + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); + return; + } + + hp_present = snd_hda_codec_read( + codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + if (!spec->hp_independent_mode) + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, + hp_present ? HDA_AMP_MUTE : 0); +} + static void via_gpio_control(struct hda_codec *codec) { unsigned int gpio_data; @@ -1488,6 +1634,8 @@ via_gpio_control(codec); if (res & VIA_JACK_EVENT) set_jack_power_state(codec); + if (res & VIA_MONO_EVENT) + via_mono_automute(codec); } static int via_init(struct hda_codec *codec) @@ -4340,6 +4488,496 @@ return 0; } + +/* Patch for VT1716S */ + +static int vt1716s_dmic_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int vt1716s_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int index = 0; + + index = snd_hda_codec_read(codec, 0x26, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (index != -1) + *ucontrol->value.integer.value = index; + + return 0; +} + +static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index = *ucontrol->value.integer.value; + + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_CONNECT_SEL, index); + spec->dmic_enabled = index; + set_jack_power_state(codec); + + return 1; +} + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1716S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Mic Capture Switch", + .count = 1, + .info = vt1716s_dmic_info, + .get = vt1716s_dmic_get, + .put = vt1716s_dmic_put, + }, + {} /* end */ +}; + + +/* mono-out mixer elements */ +static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb vt1716S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Stereo Mixer = 5 */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x5}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + + /* Setup default input of SW1 as MW0 */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* Setup default input of SW4 as AOW0 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* PW9 PW10 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute SW1, PW12 */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* PW12 Output enable */ + {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf8a, 0x80}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, + /* Enable mono output */ + {0x1, 0xf90, 0x08}, + { } +}; + + +static struct hda_verb vt1716S_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1716S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 3; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x25; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x11; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i <= AUTO_SEQ_CENLFE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x25; /* AOW3 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1716S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1716S_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1716S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1716S Analog"; + spec->stream_analog_playback = &vt1716S_pcm_analog_playback; + spec->stream_analog_capture = &vt1716S_pcm_analog_capture; + + spec->stream_name_digital = "VT1716S Digital"; + spec->stream_digital_playback = &vt1716S_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1716S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1716S_capture_mixer; + spec->num_mixers++; + } + + spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer; + spec->num_mixers++; + + spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1716S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -4420,6 +5058,10 @@ .patch = patch_vt1718S}, { .id = 0x11064441, .name = "VT1828S", .patch = patch_vt1718S}, + { .id = 0x11060433, .name = "VT1716S", + .patch = patch_vt1716S}, + { .id = 0x1106a721, .name = "VT1716S", + .patch = patch_vt1716S}, {} /* terminator */ };

1 1

[alsa-devel] [PATCH 24/31] HDA patch_via.c: Add VT1828S and VT2020 support.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VT1828S and VT2020 support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:24.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:26.000000000 +0800 @@ -121,6 +121,8 @@ else if ((dev_id & 0xfff) == 0x428 && (dev_id >> 12) < 8) codec_type = VT1718S; + else if (dev_id == 0x0441 || dev_id == 0x4441) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -4298,21 +4300,31 @@ spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; - spec->stream_name_analog = "VT1718S Analog"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_analog = "VT2020 Analog"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_analog = "VT1828S Analog"; + else + spec->stream_name_analog = "VT1718S Analog"; spec->stream_analog_playback = &vt1718S_pcm_analog_playback; spec->stream_analog_capture = &vt1718S_pcm_analog_capture; - spec->stream_name_digital = "VT1718S Digital"; + if (codec->vendor_id == 0x11060441) + spec->stream_name_digital = "VT2020 Digital"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_digital = "VT1828S Digital"; + else + spec->stream_name_digital = "VT1718S Digital"; spec->stream_digital_playback = &vt1718S_pcm_digital_playback; - if (codec->vendor_id == 0x11060428) + if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441) spec->stream_digital_capture = &vt1718S_pcm_digital_capture; if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1718S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); get_mux_nids(codec); - override_mic_boost(codec, 0x1a, 0, 3, 40); - override_mic_boost(codec, 0x1e, 0, 3, 40); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; spec->num_mixers++; } @@ -4404,6 +4416,10 @@ .patch = patch_vt1718S}, { .id = 0x11064428, .name = "VT1718S", .patch = patch_vt1718S}, + { .id = 0x11060441, .name = "VT2020", + .patch = patch_vt1718S}, + { .id = 0x11064441, .name = "VT1828S", + .patch = patch_vt1718S}, {} /* terminator */ };

1 1

[alsa-devel] [PATCH 23/31] HDA patch_via.c: Add VT1718S support.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VT1718S support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:22.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:24.000000000 +0800 @@ -86,6 +86,7 @@ VT1708S, VT1708BCE, VT1702, + VT1718S, CODEC_TYPES, }; @@ -117,6 +118,9 @@ else if ((dev_id & 0xfff) == 0x398 && (dev_id >> 12) < 8) codec_type = VT1702; + else if ((dev_id & 0xfff) == 0x428 + && (dev_id >> 12) < 8) + codec_type = VT1718S; else codec_type = UNKNOWN; return codec_type; @@ -252,6 +256,11 @@ 0x12, 0x20, 0x1F }; +static hda_nid_t vt1718S_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -480,6 +489,67 @@ snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); } + } else if (spec->codec_type == VT1718S) { + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW0/1 (24h/25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + set_pin_power_state(codec, 0x25, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, + parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } } } @@ -546,10 +616,18 @@ 0x00); static int hp_ctl_initialized = -1; + if (spec->codec_type == VT1718S) { + hda_nid_t mux1 = 0x34; + int conn_sel = snd_hda_codec_read( + codec, mux1, 0, AC_VERB_GET_CONNECT_SEL, 0x00); + pinsel = conn_sel == 0 ? 0 : 1; + } + ucontrol->value.enumerated.item[0] = pinsel; /* update HP controls on first time */ if ((spec->codec_type == VT1708S - || spec->codec_type == VT1702) + || spec->codec_type == VT1702 + || spec->codec_type == VT1718S) && hp_ctl_initialized == -1) { struct snd_card *card = codec->bus->card; struct snd_kcontrol *kctl; @@ -581,6 +659,7 @@ [VT1708B_4CH] = 0, [VT1708S] = 0x27, [VT1702] = 0, + [VT1718S] = 0, }; if (!nids_sw3[spec->codec_type]) @@ -603,7 +682,15 @@ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1718S) { + /* mux1: index 0: AOW0, index 2: AOW4 */ + hda_nid_t mux1 = 0x34; + snd_hda_codec_write(codec, mux1, 0, + AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + spec->multiout.num_dacs = 4; + } else + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) @@ -722,7 +809,8 @@ AC_VERB_GET_PIN_WIDGET_CONTROL, 0); if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* ignore FMic for independent HP */ if (ctl & AC_PINCTL_IN_EN && !(ctl & AC_PINCTL_OUT_EN)) @@ -746,7 +834,8 @@ for (i = 0; i < ARRAY_SIZE(index); i++) { hda_nid_t nid = spec->autocfg.input_pins[index[i]]; if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode) + && spec->hp_independent_mode + && spec->codec_type != VT1718S) continue; /* don't retask FMic for independent HP */ if (nid) { unsigned int parm = snd_hda_codec_read( @@ -761,6 +850,10 @@ mute_aa_path(codec, 1); notify_aa_path_ctls(codec); } + if (spec->codec_type == VT1718S) + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, + HDA_AMP_UNMUTE); } if (i == AUTO_PIN_FRONT_MIC) { if (spec->codec_type == VT1708S) { @@ -835,6 +928,11 @@ start_idx = 1; end_idx = 3; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return 0; } @@ -884,6 +982,7 @@ parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ break; case VT1708S: + case VT1718S: verb = 0xf73; parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ break; @@ -993,8 +1092,8 @@ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && - !spec->hp_independent_mode) + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] + && !spec->hp_independent_mode) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); @@ -3799,6 +3898,436 @@ return 0; } +/* Patch for VT1718S */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1718S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1718S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + + /* Setup default input of Front HP to MW9 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW9 PW10 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + /* PW11 Input enable */ + {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf88, 0x8}, + /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Unmute MW4's index 0 */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + + +static struct hda_verb vt1718S_uniwill_init_verbs[] = { + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 10, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1718S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x8; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0xa; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x9; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0xb; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; + hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; + hda_nid_t nid, nid_vol, nid_mute = 0; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0xc; /* AOW4 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 1; + break; + + case 0x2a: /* Line In */ + idx = 2; + break; + + case 0x29: /* Front Mic */ + idx = 3; + break; + + case 0x2c: /* CD */ + idx = 0; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + return 0; +} + +static int vt1718S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + static hda_nid_t vt1718s_ignore[] = {0x2e, 0}; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + vt1718s_ignore); + if (err < 0) + return err; + err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428) + spec->dig_in_nid = 0x13; + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1718S_loopbacks[] = { + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { 0x21, HDA_INPUT, 3 }, + { 0x21, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1718S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1718S Analog"; + spec->stream_analog_playback = &vt1718S_pcm_analog_playback; + spec->stream_analog_capture = &vt1718S_pcm_analog_capture; + + spec->stream_name_digital = "VT1718S Digital"; + spec->stream_digital_playback = &vt1718S_pcm_digital_playback; + if (codec->vendor_id == 0x11060428) + spec->stream_digital_capture = &vt1718S_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1718S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event, + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1718S_loopbacks; +#endif + + return 0; +} /* * patch entries */ @@ -3871,6 +4400,10 @@ .patch = patch_vt1702}, { .id = 0x11067398, .name = "VT1702", .patch = patch_vt1702}, + { .id = 0x11060428, .name = "VT1718S", + .patch = patch_vt1718S}, + { .id = 0x11064428, .name = "VT1718S", + .patch = patch_vt1718S}, {} /* terminator */ };

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[alsa-devel] [PATCH 22/31] HDA patch_via.c: Move backdoor verbs to vt17xx_volume_init_verb
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Move backdoor verbs to vt17xx_volume_init_verb As init verbs, vt17xx_volume_init_verb is a better place to hold them. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:17.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:22.000000000 +0800 @@ -3042,6 +3042,8 @@ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* Enable Mic Boost Volume backdoor */ {0x1, 0xf98, 0x1}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, { } }; @@ -3502,6 +3504,10 @@ /* PW6 PW7 Output enable */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* mixer enable */ + {0x1, 0xF88, 0x3}, + /* GPIO 0~2 */ + {0x1, 0xF82, 0x3F}, { } }; @@ -3746,8 +3752,6 @@ { struct via_spec *spec; int err; - unsigned int response; - unsigned char control; /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -3792,18 +3796,6 @@ spec->loopback.amplist = vt1702_loopbacks; #endif - /* Open backdoor */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0); - control = (unsigned char)(response & 0xff); - control |= 0x3; - snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control); - - /* Enable GPIO 0&1 for volume&mute control */ - /* Enable GPIO 2 for DMIC-DATA */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0); - control = (unsigned char)((response >> 16) & 0x3f); - snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control); - return 0; }

1 1

[alsa-devel] [PATCH 21/31] HDA patch_via.c: Substitue snd_hda_override_amp_caps for MIC_BOOST_VOLUME
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Substitue snd_hda_override_amp_caps for MIC_BOOST_VOLUME Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:08.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:17.000000000 +0800 @@ -139,46 +139,6 @@ AUTO_SEQ_SIDE }; -/* Some VT1708S based boards gets the micboost setting wrong, so we have - * to apply some brute-force and re-write the TLV's by software. */ -static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *_tlv) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - if (size < 4 * sizeof(unsigned int)) - return -ENOMEM; - if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */ - return -EFAULT; - if (put_user(2 * sizeof(unsigned int), _tlv + 1)) - return -EFAULT; - if (put_user(0, _tlv + 2)) /* offset = 0 */ - return -EFAULT; - if (put_user(1000, _tlv + 3)) /* step size = 10 dB */ - return -EFAULT; - } - return 0; -} - -static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - - if (get_codec_type(codec) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 3; - } - return 0; -} - static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); static void set_jack_power_state(struct hda_codec *codec); @@ -3037,29 +2997,15 @@ /* Patch for VT1708S */ -/* VT1708S software backdoor based override for buggy hardware micboost - * setting */ -#define MIC_BOOST_VOLUME(xname, nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ - .info = mic_boost_volume_info, \ - .get = snd_hda_mixer_amp_volume_get, \ - .put = snd_hda_mixer_amp_volume_put, \ - .tlv = { .c = mic_boost_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) } - /* capture mixer elements */ static struct snd_kcontrol_new vt1708S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), - MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A), - MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -3432,6 +3378,16 @@ }; #endif +static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, + int offset, int num_steps, int step_size) +{ + snd_hda_override_amp_caps(codec, pin, HDA_INPUT, + (offset << AC_AMPCAP_OFFSET_SHIFT) | + (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | + (step_size << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + static int patch_vt1708S(struct hda_codec *codec) { struct via_spec *spec; @@ -3468,6 +3424,8 @@ spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; }

1 1

[alsa-devel] [PATCH 20/31] HDA patch_via.c: Modify vt1709_auto_create_multi_out_ctls.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Modify vt1709_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:06.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:08.000000000 +0800 @@ -2133,7 +2133,7 @@ { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -2142,43 +2142,45 @@ if (!nid) continue; + nid_vol = nid_vols[i]; + if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT){ - /* add control to mixer index 0 */ + /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -2199,26 +2201,26 @@ } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err;

1 1

[alsa-devel] [PATCH 19/31] HDA patch_via.c: Modify vt1708_auto_create_multi_out_ctls.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Modify vt1708_auto_create_multi_out_ctls. Rewrite nid_vol/mute assignment for clearity, and check line connection before adding control for it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:04.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:06.000000000 +0800 @@ -1521,7 +1521,7 @@ { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid, nid_vol = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -1530,8 +1530,7 @@ if (!nid) continue; - if (i != AUTO_SEQ_FRONT) - nid_vol = 0x18 + i; + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -1563,13 +1562,13 @@ /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err;

1 1

[alsa-devel] [PATCH 16/31] HDA patch_via.c: Remove via_playback_pcm_prepare/cleanup
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Remove via_playback_pcm_prepare/cleanup replaced with via_playback_multi_pcm_prepare/cleanup to support multi-stream operations Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:49.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:59.000000000 +0800 @@ -989,26 +989,6 @@ hinfo); } -static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - - static void playback_multi_pcm_prep_0(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -1217,7 +1197,7 @@ }; static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ @@ -1228,8 +1208,8 @@ .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -2023,8 +2003,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -2035,8 +2015,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -3127,8 +3107,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, .close = via_pcm_open_close }, };

1 1

[alsa-devel] [PATCH 15/31] HDA patch_via.c: Modify vt1708_set_pinconfig_connect function.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Modify vt1708_set_pinconfig_connect function. like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port Connectivity field into 'AC_JACK_PORT_COMPLEX' Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:36.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:49.000000000 +0800 @@ -1725,11 +1725,10 @@ def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { - if (seqassoc == 0xff) { - def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_set_pincfg(codec, nid, def_conf); - } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE + && (seqassoc == 0xf0 || seqassoc == 0xff)) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } return;

1 1

[alsa-devel] [PATCH 12/31] HDA patch_via.c: Add VIA_JACK_EVENT process in via_unsol_event.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:05.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:13.000000000 +0800 @@ -124,6 +124,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 +#define VIA_JACK_EVENT 0x04 enum { VIA_CTL_WIDGET_VOL, @@ -1408,10 +1409,12 @@ unsigned int res) { res >>= 26; - if (res == VIA_HP_EVENT) + if (res & VIA_HP_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) + if (res & VIA_GPIO_EVENT) via_gpio_control(codec); + if (res & VIA_JACK_EVENT) + set_jack_power_state(codec); } static int via_init(struct hda_codec *codec) @@ -1873,7 +1876,8 @@ }; static struct hda_verb vt1709_uniwill_init_verbs[] = { - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } }; @@ -2509,7 +2513,15 @@ }; static struct hda_verb vt1708B_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3004,7 +3016,15 @@ }; static struct hda_verb vt1708S_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3443,8 +3463,12 @@ }; static struct hda_verb vt1702_uniwill_init_verbs[] = { - {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } };

1 1

[alsa-devel] [PATCH 11/31] HDA patch_via.c: When changing input source, update jack power state.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: When changing input source, update jack power state. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:55.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:05.000000000 +0800 @@ -548,6 +548,14 @@ if (!spec->mux_nids[adc_idx]) return -EINVAL; + /* switch to D0 beofre change index */ + if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) + snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* update jack power state */ + set_jack_power_state(codec); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);

1 1

[alsa-devel] [PATCH 10/31] HDA patch_via.c: Add smart5.1 function.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add smart5.1 function. Smart 5.1 is for 3-jacks model, to reuse input pins as outputs. While off, they act as "line out" / "line in" / "mic in". While on, they acts as "line out" / "back left/right" / "center/lfe". Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:46.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:55.000000000 +0800 @@ -211,7 +211,7 @@ struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; + struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -253,7 +253,7 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; - + unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -390,6 +390,8 @@ } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -400,9 +402,10 @@ & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ unsigned present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) >> 31; - - if ((no_presence || present) && get_defcfg_connect(def_conf) - != AC_JACK_PORT_NONE) { + struct via_spec *spec = codec->spec; + if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) + || ((no_presence || present) + && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { *affected_parm = AC_PWRST_D0; /* if it's connected */ parm = AC_PWRST_D0; } else @@ -649,6 +652,171 @@ { } /* end */ }; +static void notify_aa_path_ctls(struct hda_codec *codec) +{ + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + list_for_each_entry(kctl, &card->controls, list) { + int i; + static char *labels[] = {"Mic", "Front Mic", "Line"}; + for (i = 0; i < ARRAY_SIZE(labels); i++) { + char name[32]; + sprintf(name, "%s Playback Volume", labels[i]); + if (strcmp(kctl->id.name, name) == 0) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctl->id); + break; + } + } + } +} + +static void mute_aa_path(struct hda_codec *codec, int mute) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708: + nid_mixer = 0x17; + start_idx = 2; + end_idx = 4; + break; + case VT1709_10CH: + case VT1709_6CH: + nid_mixer = 0x18; + start_idx = 2; + end_idx = 4; + break; + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + default: + return; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE; + snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i, + HDA_AMP_MUTE, val); + } +} +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) +{ + int res = 0; + int index; + for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { + if (pin == spec->autocfg.input_pins[index]) { + res = 1; + break; + } + } + return res; +} + +static int via_smart51_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int via_smart51_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int on = 1; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + int ctl = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* ignore FMic for independent HP */ + if (ctl & AC_PINCTL_IN_EN + && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; + } + } + *ucontrol->value.integer.value = on; + return 0; +} + +static int via_smart51_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int out_in = *ucontrol->value.integer.value + ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* don't retask FMic for independent HP */ + if (nid) { + unsigned int parm = snd_hda_codec_read( + codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + } + if (i == AUTO_PIN_FRONT_MIC) { + if (spec->codec_type == VT1708S) { + /* input = index 1 (AOW3) */ + snd_hda_codec_write( + codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, 1); + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, + 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); + } + } + } + spec->smart51_enabled = *ucontrol->value.integer.value; + set_jack_power_state(codec); + return 1; +} + +static struct snd_kcontrol_new via_smart51_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Smart 5.1", + .count = 1, + .info = via_smart51_info, + .get = via_smart51_get, + .put = via_smart51_put, + }, + {} /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1579,6 +1747,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2079,6 +2248,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2641,6 +2811,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -3134,6 +3305,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; }

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[alsa-devel] [PATCH 9/31] HDA patch_via.c: Rewrite via_independent_hp_put
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Rewrite via_independent_hp_put Use hp_independent_mode_index to store hp index, and simplify function via_independent_hp_put with it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:33.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:46.000000000 +0800 @@ -252,6 +252,7 @@ /* HP mode source */ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; enum VIA_HDA_CODEC codec_type; @@ -589,6 +590,30 @@ return 0; } +static int update_side_mute_status(struct hda_codec *codec) +{ + /* mute side channel */ + struct via_spec *spec = codec->spec; + unsigned int parm = spec->hp_independent_mode + ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + hda_nid_t nids_sw3[CODEC_TYPES] = { + [VT1708] = 0x1b, + [VT1709_10CH] = 0x29, + [VT1709_6CH] = 0, + [VT1708B_8CH] = 0x27, + [VT1708B_4CH] = 0, + [VT1708S] = 0x27, + [VT1702] = 0, + }; + + if (!nids_sw3[spec->codec_type]) + return 0; + + snd_hda_codec_write(codec, nids_sw3[spec->codec_type], 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + return 0; +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -596,47 +621,19 @@ struct via_spec *spec = codec->spec; hda_nid_t nid = spec->autocfg.hp_pins[0]; unsigned int pinsel = ucontrol->value.enumerated.item[0]; - unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - if (con_nid == spec->multiout.hp_nid) { - if (pinsel == 0) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } else if (pinsel == 1) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } - } else { - if (pinsel == 0) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } else if (pinsel == 1) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - pinsel); - - if (spec->multiout.hp_nid && - spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) - snd_hda_codec_setup_stream(codec, - spec->multiout.hp_nid, - 0, 0, 0); + /* Get Independent Mode index of headphone pin widget */ + spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel + ? 1 : 0; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + + if (spec->multiout.hp_nid && spec->multiout.hp_nid + != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, + 0, 0, 0); + update_side_mute_status(codec); return 0; } @@ -1444,6 +1441,7 @@ return 0; spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1979,6 +1977,7 @@ spec->multiout.hp_nid = VT1709_HP_DAC_NID; else if (spec->multiout.num_dacs == 3) /* 6 channels */ spec->multiout.hp_nid = 0; + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2538,6 +2537,7 @@ return 0; spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3008,6 +3008,7 @@ return 0; spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3367,6 +3368,7 @@ if (!pin) return 0; spec->multiout.hp_nid = 0x1D; + spec->hp_independent_mode_index = 0; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume",

1 1

[alsa-devel] [PATCH 7/31] HDA patch_via.c: Remove unused argument of via_new_analog_input
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Remove unused argument of via_new_analog_input Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:07.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:18.000000000 +0800 @@ -317,8 +317,8 @@ } /* create input playback/capture controls for the given pin */ -static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, - const char *ctlname, int idx, int mix_nid) +static int via_new_analog_input(struct via_spec *spec, const char *ctlname, + int idx, int mix_nid) { char name[32]; int err; @@ -1480,8 +1480,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x17); + err = via_new_analog_input(spec, labels[i], idx, 0x17); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2014,8 +2013,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x18); + err = via_new_analog_input(spec, labels[i], idx, 0x18); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2576,8 +2574,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3048,8 +3045,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3402,8 +3398,7 @@ idx = 3; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], - labels[i], idx, 0x1A); + err = via_new_analog_input(spec, labels[i], idx, 0x1A); if (err < 0) return err; imux->items[imux->num_items].label = labels[i];

1 1

[alsa-devel] [PATCH 6/31] HDA patch_via.c: Add low current mode for power saving.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add low current mode for power saving. For VT1708B, VT1708S and VT1702, enter low current mode if no analog stream is opened and all aa path mute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:08:44.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:08:59.000000000 +0800 @@ -783,6 +783,10 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -1089,6 +1093,11 @@ if (err < 0) return err; } + + /* init power states */ + set_jack_power_state(codec); + analog_low_current_mode(codec, 1); + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -2312,6 +2321,17 @@ { } }; +static int via_pcm_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); + return 0; +} + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -2320,7 +2340,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2342,8 +2363,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2800,7 +2823,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .cleanup = via_playback_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2810,8 +2834,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3236,7 +3262,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3246,8 +3273,10 @@ .channels_max = 2, .nid = 0x12, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, };

1 1

[alsa-devel] [PATCH 5/31] HDA patch_via.c: Add VIA_CTL_WIDGET_ANALOG_MUTE control type
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:08:20.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:02.000000000 +0800 @@ -128,6 +128,7 @@ enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, + VIA_CTL_WIDGET_ANALOG_MUTE, }; enum { @@ -177,9 +178,34 @@ return 0; } +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + return change; +} + +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, }; @@ -303,7 +329,7 @@ if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -362,6 +388,131 @@ } } + +static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int *affected_parm) +{ + unsigned parm; + unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); + unsigned no_presence = (def_conf & AC_DEFCFG_MISC) + >> AC_DEFCFG_MISC_SHIFT + & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ + unsigned present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31; + + if ((no_presence || present) && get_defcfg_connect(def_conf) + != AC_JACK_PORT_NONE) { + *affected_parm = AC_PWRST_D0; /* if it's connected */ + parm = AC_PWRST_D0; + } else + parm = AC_PWRST_D3; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + +static void set_jack_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + + if (spec->codec_type == VT1702) { + imux_is_smixer = snd_hda_codec_read( + codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x16, &parm); + set_pin_power_state(codec, 0x17, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, + parm); + } else if (spec->codec_type == VT1708B_8CH + || spec->codec_type == VT1708B_4CH + || spec->codec_type == VT1708S) { + /* SW0 (17h) = stereo mixer */ + int is_8ch = spec->codec_type != VT1708B_4CH; + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } +} + /* * input MUX handling */ @@ -504,6 +655,93 @@ }, { } /* end */ }; + +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) +{ + struct via_spec *spec = codec->spec; + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; + case VT1708S: + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ + break; + case VT1702: + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ + break; + default: + return; /* other codecs are not supported */ + } + /* send verb */ + snd_hda_codec_write(codec, codec->afg, 0, verb, parm); +} + /* * generic initialization of ADC, input mixers and output mixers */

1 1

[alsa-devel] [PATCH 4/31] HDA patch_via.c: Modify VT1702 AA-Path max volume
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Modify VT1702 AA-Path max volume according to customer request, VT1702 AA-Path max volume (12 dB) is too high, so limit to 0 dB. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-04 23:05:19.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 00:24:34.000000000 +0800 @@ -3166,6 +3166,12 @@ err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; + /* limit AA path volume to 0 dB */ + snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err;

1 1

[alsa-devel] [PATCH 3/31] HDA patch_via.c: Add VT1708B-CE codec support.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VT1708B-CE codec support. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-04 23:05:10.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-04 23:05:19.000000000 +0800 @@ -84,6 +84,7 @@ VT1708B_8CH, VT1708B_4CH, VT1708S, + VT1708BCE, VT1702, CODEC_TYPES, }; @@ -104,9 +105,11 @@ codec_type = VT1709_10CH; else if (dev_id >= 0xe714 && dev_id <= 0xe717) codec_type = VT1709_6CH; - else if (dev_id >= 0xe720 && dev_id <= 0xe723) + else if (dev_id >= 0xe720 && dev_id <= 0xe723) { codec_type = VT1708B_8CH; - else if (dev_id >= 0xe724 && dev_id <= 0xe727) + if (snd_hda_param_read(codec, 0x16, AC_PAR_CONNLIST_LEN) == 0x7) + codec_type = VT1708BCE; + } else if (dev_id >= 0xe724 && dev_id <= 0xe727) codec_type = VT1708B_4CH; else if ((dev_id & 0xfff) == 0x397 && (dev_id >> 12) < 8) @@ -224,6 +227,8 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + enum VIA_HDA_CODEC codec_type; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -979,6 +984,10 @@ for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + spec->codec_type = get_codec_type(codec); + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost + same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { @@ -2369,12 +2378,14 @@ { } /* end */ }; #endif - +static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { struct via_spec *spec; int err; + if (get_codec_type(codec) == VT1708BCE) + return patch_vt1708S(codec); /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2906,6 +2917,16 @@ spec->loopback.amplist = vt1708S_loopbacks; #endif + /* correct names for VT1708BCE */ + if (get_codec_type(codec) == VT1708BCE) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + spec->stream_name_analog = "VT1708BCE Analog"; + spec->stream_name_digital = "VT1708BCE Digital"; + } return 0; }

1 1

[alsa-devel] [PATCH 2/31] HDA patch_via.c: Change get_codec_type argument to hda_codec type
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Change get_codec_type argument to hda_codec type Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-04 23:01:14.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-04 23:05:10.000000000 +0800 @@ -88,8 +88,9 @@ CODEC_TYPES, }; -static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { + u32 vendor_id = codec->vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -141,7 +142,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { if (size < 4 * sizeof(unsigned int)) return -ENOMEM; @@ -163,7 +164,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2;

1 1

[alsa-devel] [PATCH 13/31] HDA patch_via.c: Refresh front playback mute if hp plug-in mutes speaker.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Refresh front playback mute if hp plug-in mutes speaker. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:13.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:22.000000000 +0800 @@ -1346,14 +1346,26 @@ /* mute internal speaker if HP is plugged */ static void via_hp_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = 0; struct via_spec *spec = codec->spec; + char name[32] = "Front Playback Switch"; present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], - HDA_OUTPUT, 0, HDA_AMP_MUTE, - present ? HDA_AMP_MUTE : 0); + + if (!spec->hp_independent_mode) { + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + /* auto mute */ + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + /* notify change */ + list_for_each_entry(kctl, &card->controls, list) + if (strcmp(kctl->id.name, name) == 0) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctl->id); + } } static void via_gpio_control(struct hda_codec *codec)

2 5

[alsa-devel] [PATCH 17/31] HDA patch_via.c: Add second S/PDIF out control for VT1708S and VT1702.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add second S/PDIF out control for VT1708S and VT1702. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:59.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:02.000000000 +0800 @@ -1265,6 +1265,13 @@ if (err < 0) return err; spec->multiout.share_spdif = 1; + /* lydia add for second spdif out */ + if (spec->slave_dig_outs[0]) { + err = snd_hda_create_spdif_out_ctls(codec, + spec->slave_dig_outs[0]); + if (err < 0) + return err; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -3127,7 +3134,7 @@ }; static struct hda_pcm_stream vt1708S_pcm_digital_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 2, /* NID is set in via_build_pcms */

2 5

Re: [alsa-devel] [PATCH 18/31] HDA patch_via.c: Add 48K sample rate limit for S/PDIF streams.
by Demian Martin 08 Oct '09

08 Oct '09

Li Bo: I'm not sure how changing the sample rate for SPDIF would have an effect of the SNR if you are only passing digital signals. I have had excellent measured results with the Via chipsets at all sample rates up to 192 KHz at 24 bits on the digital output. Perhaps you can explain how you measured the performance and that may explain your results. Normally if all of the data of the original file is present then any limit on SNR is a function of the DAC. If the data is not the same as the original file then there are more serious issues to be resolved, and it probably would not be useable with Dolby Digital or DTS. Higher sample rates for digital content and digital interfaces are becoming more common and I think it would be a serious mistake to limit the performance of the platform. >Hi, Takashi >According to our HW test, S/PDIF at 48K sample rate has best SNR, so this limit can make >S/PDIF work better. Demian Martin Product Design Service 784 Cary Drive San Leandro, CA 94577 209 613 6990

3 2

[alsa-devel] [PATCH 8/31] HDA patch_via.c: Change VT1708S & VT1702 hp mode controls
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Change VT1708S & VT1702 hp mode controls For VT1708S and VT1702, deactivate "Headphone Playback Volume" and "Headphone Playback Mute" control if "Independent HP" mode is OFF. and rename VT1702 "Independent HP" text. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:18.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:33.000000000 +0800 @@ -566,9 +566,26 @@ unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0x00); + static int hp_ctl_initialized = -1; ucontrol->value.enumerated.item[0] = pinsel; - + /* update HP controls on first time */ + if ((spec->codec_type == VT1708S + || spec->codec_type == VT1702) + && hp_ctl_initialized == -1) { + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + /* deactive hp volume & switch in independent mode */ + list_for_each_entry(kctl, &card->controls, list) + if (spec->hp_independent_mode + && (strcmp(kctl->id.name, + "Headphone Playback Volume") == 0 + || strcmp(kctl->id.name, + "Headphone Playback Switch") == 0)) + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + hp_ctl_initialized = 1; + } return 0; } @@ -3342,11 +3359,13 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { - int err; - + int err, i; + struct hda_input_mux *imux; + static const char *texts[] = { "ON (need reboot)", + "OFF (need reboot)", + NULL}; if (!pin) return 0; - spec->multiout.hp_nid = 0x1D; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -3361,8 +3380,18 @@ if (err < 0) return err; - create_hp_imux(spec); + imux = &spec->private_imux[1]; + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; return 0; }

2 10

[alsa-devel] [PATCH 14/31] HDA patch_via.c: Add Jack detect feature for VT1708.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add Jack detect feature for VT1708. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:22.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:36.000000000 +0800 @@ -257,6 +257,11 @@ unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work hp_jack_work; + int vt1708_jack_detectect; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -966,6 +971,8 @@ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* power down jack detect function */ + {0x1, 0xf81, 0x1}, { } }; @@ -1340,6 +1347,10 @@ return; via_free_kctls(codec); + if (spec->codec_type == VT1708) { + cancel_delayed_work(&spec->hp_jack_work); + flush_scheduled_work(); + } kfree(codec->spec); } @@ -1724,6 +1735,52 @@ return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + snd_hda_codec_write(codec, 0x1, 0, 0xf81, !spec->vt1708_jack_detectect); + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1751,6 +1808,10 @@ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1784,6 +1845,24 @@ return 0; } +static void update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + hp_jack_work.work); + static int present = 1; + + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + present ^= 1; + via_hp_automute(spec->codec); + } + schedule_delayed_work(&spec->hp_jack_work, msecs_to_jiffies(100)); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1860,7 +1939,9 @@ #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->hp_jack_work, update_hp_jack_state); + schedule_delayed_work(&spec->hp_jack_work, msecs_to_jiffies(100)); return 0; }

3 11

Re: [alsa-devel] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200
by Takashi Iwai 08 Oct '09

08 Oct '09

At Wed, 23 Sep 2009 00:39:45 -0500 (CDT), Manoj Iyer wrote: > > > >From c96629e96fecbe59de7124ec913019ed378d0029 Mon Sep 17 00:00:00 2001 > From: Manoj Iyer <manoj.iyer(a)canonical.com> > Date: Tue, 22 Sep 2009 18:33:29 -0500 > Subject: [PATCH] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200 > > Patch was tested on Toshiba NB200 and is found to enable sound. So, it doesn't work without the model quirk? Could you give alsa-info.sh output (run with --no-upload option)? thanks, Takashi > > Signed-off-by: Manoj Iyer <manoj.iyer(a)canonical.com> > Cc: stable(a)kernel.org > --- > 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 30eeb30..713bf69 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -16876,6 +16876,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { > SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), > SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", > ALC662_3ST_6ch_DIG), > + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), > SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), > SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", > ALC662_3ST_6ch_DIG), > -- > 1.6.3.3 > > > --- manjo >

2 13

[alsa-devel] [PATCHv3 1/7] OMAP4: PMIC: Add support for twl6030 codec
by Lopez Cruz, Misael 08 Oct '09

08 Oct '09

In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data may be passed: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected However, codec driver can operate if any or none of them are passed. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- drivers/mfd/twl-core.c | 15 +++++++++++++++ include/linux/i2c/twl.h | 7 +++++++ 2 files changed, 22 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index af4cf47..6432af1 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -108,6 +108,12 @@ #define twl_has_mmc() false #endif +#if defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) +#define twl_has_codec() true +#else +#define twl_has_codec() false +#endif + #if defined(CONFIG_TWL4030_USB) || defined(CONFIG_TWL4030_USB_MODULE) #define twl_has_usb() true #else @@ -190,6 +196,7 @@ #define BCI_SUB_CHIP_ID SUB_CHIP_ID1 #define GPIO_SUB_CHIP_ID 0 /* NOT SUPPORTED IN TWL6030 */ #define KEYPAD_SUB_CHIP_ID 0 /* ADDED FOR COMPILATION ONLY */ +#define CODEC_SUB_CHIP_ID SUB_CHIP_ID3 /* subchip/slave 0 0x48 - POWER */ #define TWL6030_BASEADD_RTC 0x0000 @@ -632,6 +639,14 @@ add_children(struct twl_platform_data *pdata, unsigned long features) if (IS_ERR(child)) return PTR_ERR(child); } + + if (twl_has_codec()) { + child = add_child(CODEC_SUB_CHIP_ID, "twl6030_codec", + pdata->codec, sizeof(*pdata->codec), false, + 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } #endif if (twl_has_usb() && pdata->usb) { diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index b687a8b..e76ca9b 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -472,6 +472,12 @@ struct twl_usb_data { enum twl_usb_mode usb_mode; }; +struct twl_codec_data { + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ +}; + struct twl_platform_data { unsigned irq_base, irq_end; struct twl_bci_platform_data *bci; @@ -479,6 +485,7 @@ struct twl_platform_data { struct twl_madc_platform_data *madc; struct twl_keypad_data *keypad; struct twl_usb_data *usb; + struct twl_codec_data *codec; /* LDO regulators common to TWL4030/TWL6030 */ struct regulator_init_data *vdac; -- 1.5.4.3

5 10

[alsa-devel] Possibility to build alsa driver without having exact kernel source?
by Christian Steffen 07 Oct '09

07 Oct '09

Hello list, I need a new driver for a atom cpu driven device. It needs the "snd_intel_hda" driver to work as it has intels Poulsbo chipset. Currently I have a compiled module, but sound has a constant clicking noise with this one. Therefore I'd like to try an updated alsa driver, which hopefully fixes a problem on Poulsbo chipsets with stepping D0 or newer. My problem is that I only have a compiled running kernel, but no source from it. All I can see (trough uname -a) is: Linux joggler 2.6.30 #4 SMP Tue Jun 23 21:55:08 BST 2009 i686 GNU/Linux So I know, the kernel is a 2.6.30, arch i686. I tried to compile the driver now just with the standard kernel 2.6.30 source from kernel. org. But when trying to load these modules, I get "Invalid module format". Is there any chance to get a working compiled driver without having more information? Thanks! Christian Steffen P Bitte drucken Sie diese Mail nur, wenn dies notwendig ist. please don't print this e-mail unless you really need to

1 0

Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Aha, sorry - I misunderstood what was being asked to test. Two channel audio is working correctly with the development snapshot. Thanks, Gus On Wed 7/10/09 10:16 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 7 Oct 2009 03:12:49 +0000, > wrote: > > > > Hello Takashi Iwai, > > > > I tested your latest snapshot and it did not detect the 6 channel > audio. > This is so per design. The default auto-parser doesn't enable > 6-channel since otherwise you get problems with the automatic HP / > mic > jack detection. > Does it work except for 6-channel things? > thanks, > Takashi > _______________________________________________ > Alsa-devel mailing list > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Message sent via Atmail Open - http://atmail.org/

2 1

[alsa-devel] ascenario: Small updates for new API.
by Stefan Schmidt 07 Oct '09

07 Oct '09

Hello. [PATCH 1/3] ascenario: Rename SND_SCN_INT_QOS to SND_SCN_INT_QOS_POWER As proposed by Jaroslav. Emphasis the power consumption relationship. [PATCH 2/3] ascenario: Fix snd_scenario_get_control_id() declaration to match implementation Just different names in declaration and implementation. I favoured _control_id(). [PATCH 3/3] ascenario: Implement snd_scenario_get_control_id() Replace the four explicit functions with a generic one. I was not able to find another way to get the id from the given numid then getting the full list and iterate through it. If there is something easier let me know. :) regards Stefan Schmidt

1 3

[alsa-devel] [RFC] Add scenario management
by Stefan Schmidt 07 Oct '09

07 Oct '09

Hello. This patchset add support for scenario management, aka use cases, in alsa-lib and example programs in alsa-utils. It allows to adjust the current audio kcontrol settings from a prespective of a changed use case like the switch from listening to music on your phone to an incoming phone call and back. To achieve this it offers control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. This is of course not meant to replace gstreamer, PulseAudio or a sound server, but is meant to work in tandem with such audio software. We think alsa-lib is the best place for this as it will be used on small embedded system which only use alsa from the full stack of audio libs in linux. We see potential for desktop usage as well though. There is also the support of QoS and the submission to salsa on the TODO list, but we think it is best to gather soem feedback and get the core merged before drifting away with to much other features. Please let me know what can be improved. Project page is here :- http://www.slimlogic.co.uk/?p=40 regards Stefan Schmidt

6 48

Re: [alsa-devel] sscape: general code improvements
by krzysztof.h1＠poczta.fm 07 Oct '09

07 Oct '09

"Takashi Iwai" <tiwai(a)suse.de> pisze: > At Tue, 6 Oct 2009 19:34:38 +0200, > Krzysztof Helt wrote: > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > General code and coding style improvements. > > > > Add a missing argument for SET_XXX_MIDI_VOL command. > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > Do you really want to change oss/sscape.c that you wanted to remove? > No, it is a mistake. > > > --- > > > > What is a standard name for switch to select internal or external midi? > It is very easy to > > add such a functionality to the driver. > > What do you mean exactly to switch int/ext midi? > Usually the driver provides the interfaces for both, and let user choose. > If they are exclusive, it's checked in the open. > Or, if preferred, a control element can be used to choose the > interface... > The driver interface (MPU-401 UART) is only one and there is a command to switch it to the on-board synthetizer or external midi (joysitck port on the card). Regards, Krzysztof --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

2 1

Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Hello Takashi Iwai, I tested your latest snapshot and it did not detect the 6 channel audio. I have attached the output of alsa-info for analysis. The original patch that I submitted did not detect the coaxial/optical SPDIF interface on this board. I've been running with the ALC662_3ST_6ch_DIG method instead of ALC662_3ST_6ch which was in the original patch. The _DIG method now shows my spdif interface correctly and my 6 channel analog audio is also working correctly. I do not have a receiver to test the coaxial/optical output, though I suspect it works correctly. I've attached a new patch which uses the _DIG method as I believe its the more proper method to use for this hardware layout. Thanks, Gus Bourg On Mon 28/09/09 9:30 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > Could you check whether the very latest alsa-driver snapshot still > doesn't work without your change? > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna > pshot.tar.gzIf it doesn't work, could you give alsa-info.sh output (run > --no-upload option)? > If your fix is mandatory, I'm going to apply it happily. But, then > please give your sign-off, too. > thanks, > Takashi > > Message sent via Atmail Open - http://atmail.org/

2 1

[alsa-devel] alsa-utils cross compilation error
by Harish Kumar 07 Oct '09

07 Oct '09

Hi, I am trying to cross compile the alsa utils for PPC and as well as arm architecture. the compilation of alsa-libs was successful with their respective toolchains. But configuring the alsa-util is failing. I am using alsa-utils-1.0.11rc2,alsa-lib-1.0.11rc2. I am getting the error checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. I feel i have to provide the path of the include files but not sure which path to be passed and how it needs to be passed to configure. Request your help to resolve the same. I invoked the configure using the following command and the output of the configure is as below. root@localhost alsa-utils-1.0.11rc2]# CC=arm-linux-gcc ./configure --target=arm-linux --host=i686-linux configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... gawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ANSI C... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld checking if the linker (/opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking whether NLS is requested... yes checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ANSI C... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. [2]+ Done gedit configure [root@localhost alsa-utils-1.0.11rc2]# Thanks and Regards Harish Kumar.B

3 3

[alsa-devel] ice1724 SPDIF buffer size?
by Robert Hancock 07 Oct '09

07 Oct '09

I noticed that in the ice1724 driver, we set a default and maximum preallocated buffer size of 256K for the "professional" stream, but only 64K for the SPDIF and independent stereo pair streams, even though the hardware supports 256K for all streams. Is there any reason why we can't use 256K for all of them? I made this change here and it seems to work fine (for SPDIF at least, haven't tested stereo). From a power and CPU usage perspective, it makes sense to allow as big a buffer as possible for when latency isn't critical..

2 1

[alsa-devel] sscape: general code improvements
by Krzysztof Helt 07 Oct '09

07 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> General code and coding style improvements. Add a missing argument for SET_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- What is a standard name for switch to select internal or external midi? It is very easy to add such a functionality to the driver. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be50..4380e85 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -190,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -331,7 +334,8 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) */ static inline void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -352,22 +356,25 @@ static void soundscape_free(struct snd_card *c) */ static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -429,11 +436,12 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) */ static int upload_dma_data(struct soundscape *s, const unsigned char *data, - size_t size) + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +451,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +483,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +500,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -591,7 +603,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +613,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +627,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +651,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +753,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +777,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +793,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +804,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +818,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +881,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +930,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1155,7 +1157,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1186,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1240,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1297,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c index 30c36d1..7e06424 100644 --- a/sound/oss/sscape.c +++ b/sound/oss/sscape.c @@ -160,19 +160,27 @@ static void sleep(unsigned howlong) schedule_timeout(howlong); } +static unsigned char __sscape_read(struct sscape_info *devc, int reg) +{ + unsigned char val; + + outb(reg, PORT(ODIE_ADDR)); + val = inb(PORT(ODIE_DATA)); + return val; +} + static unsigned char sscape_read(struct sscape_info *devc, int reg) { unsigned long flags; unsigned char val; spin_lock_irqsave(&devc->lock,flags); - outb(reg, PORT(ODIE_ADDR)); - val = inb(PORT(ODIE_DATA)); + val = __sscape_read(devc, reg); spin_unlock_irqrestore(&devc->lock,flags); return val; } -static void __sscape_write(int reg, int data) +static void __sscape_write(struct sscape_info *devc, int reg, int data) { outb(reg, PORT(ODIE_ADDR)); outb(data, PORT(ODIE_DATA)); @@ -183,7 +191,7 @@ static void sscape_write(struct sscape_info *devc, int reg, int data) unsigned long flags; spin_lock_irqsave(&devc->lock,flags); - __sscape_write(reg, data); + __sscape_write(devc, reg, data); spin_unlock_irqrestore(&devc->lock,flags); } @@ -342,9 +350,9 @@ static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, in temp |= 0x80; /* 8 bit DMA channel */ temp |= 1; /* Trigger DMA */ - sscape_write(devc, GA_DMAA_REG, temp); + __sscape_write(devc, GA_DMAA_REG, temp); temp &= 0xfe; /* Clear DMA trigger */ - sscape_write(devc, GA_DMAA_REG, temp); + __sscape_write(devc, GA_DMAA_REG, temp); } static int verify_mpu(struct sscape_info *devc) @@ -392,7 +400,7 @@ static void sscape_coproc_close(void *dev_info, int sub_device) spin_lock_irqsave(&devc->lock,flags); if (devc->dma_allocated) { - __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */ + __sscape_write(devc, GA_DMAA_REG, 0x20); /* DMA channel disabled */ devc->dma_allocated = 0; } spin_unlock_irqrestore(&devc->lock,flags); @@ -418,7 +426,7 @@ static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, */ spin_lock_irqsave(&devc->lock,flags); - codec_dma_bits = sscape_read(devc, GA_CDCFG_REG); + codec_dma_bits = __sscape_read(devc, GA_CDCFG_REG); if (devc->dma_allocated == 0) devc->dma_allocated = 1; @@ -500,11 +508,11 @@ static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, x = inb(PORT(HOST_DATA)); if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ { - DDB(printk("Soundscape: Acknowledge = %x\n", x)); + (printk("Soundscape: Acknowledge = %x\n", x)); done = 1; } } - sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); + __sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); spin_unlock_irqrestore(&devc->lock,flags); if (!done) @@ -604,6 +612,16 @@ static coproc_operations sscape_coproc_operations = static struct resource *sscape_ports; static int sscape_is_pnp; +static int sscape_read_host_ctrl(sscape_info* devc) +{ + return host_read(devc); +} + +static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) +{ + host_command2(devc, a, b); +} + static void __init attach_sscape(struct address_info *hw_config) { #ifndef SSCAPE_REGS @@ -669,22 +687,25 @@ static void __init attach_sscape(struct address_info *hw_config) spin_lock_irqsave(&devc->lock,flags); /* Host interrupt enable */ - sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ + __sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ /* DMA A status/trigger register */ - sscape_write(devc, 2, 0x20); /* DMA channel disabled */ + __sscape_write(devc, 2, 0x20); /* DMA channel disabled */ /* DMA B status/trigger register */ - sscape_write(devc, 3, 0x20); /* DMA channel disabled */ + __sscape_write(devc, 3, 0x20); /* DMA channel disabled */ /* Host interrupt config reg */ - sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); + __sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); /* Don't destroy CD-ROM DMA config bits (0xc0) */ - sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0)); + __sscape_write(devc, 5, (regs[5] & 0x3f) | (__sscape_read(devc, 5) & 0xc0)); /* CD-ROM config (WSS codec actually) */ - sscape_write(devc, 6, regs[6]); - sscape_write(devc, 7, regs[7]); - sscape_write(devc, 8, regs[8]); + __sscape_write(devc, 6, regs[6]); + __sscape_write(devc, 7, regs[7]); + __sscape_write(devc, 8, regs[8]); /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */ - sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08); + __sscape_write(devc, 9, (__sscape_read(devc, 9) & 0xf0) | 0x08); spin_unlock_irqrestore(&devc->lock,flags); + sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */ + sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */ + sscape_write_host_ctrl2( devc, 0x8A, 0); } #ifdef SSCAPE_DEBUG2 /* @@ -721,7 +742,7 @@ static int detect_ga(sscape_info * devc) { unsigned char save; - DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base)); + printk(KERN_DEBUG "Entered Soundscape detect_ga(%x)\n", devc->base); /* * First check that the address register of "ODIE" is @@ -771,16 +792,6 @@ static int detect_ga(sscape_info * devc) return 1; } -static int sscape_read_host_ctrl(sscape_info* devc) -{ - return host_read(devc); -} - -static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) -{ - host_command2(devc, a, b); -} - static int sscape_alloc_dma(sscape_info *devc) { char *start_addr, *end_addr; @@ -869,8 +880,8 @@ static void sscape_pnp_start_dma(sscape_info* devc, int arg ) if (arg == 0) reg = 2; else reg = 3; - sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); - sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); + __sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); + __sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); } static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) @@ -885,11 +896,11 @@ static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) sleep ( 1 ); i = 0; do { - d = sscape_read(devc, reg) & 1; + d = __sscape_read(devc, reg) & 1; if ( d == 1) break; i++; } while (i < 500000); - d = sscape_read(devc, reg) & 1; + d = __sscape_read(devc, reg) & 1; return d; } @@ -1242,8 +1253,11 @@ static int __init probe_sscape(struct address_info *hw_config) } if (!detect_ga(devc)) { - if (detect_sscape_pnp(devc)) + printk(KERN_DEBUG "GA detected!\n"); + if (detect_sscape_pnp(devc)) { + printk(KERN_DEBUG "PnP detected!\n"); return 1; + } release_region(devc->base, 2); release_region(devc->base + 2, 6); return 0; ---------------------------------------------------------------------- Zobacz najwiekszy samolot na swiecie! Kliknij >>> http://link.interia.pl/f238f

2 1

[alsa-devel] PCM device volume
by Milan Křápek 07 Oct '09

07 Oct '09

Hi, I am new with alsa. I am developing application that will have audio output on linux systems via alsa. I found many hints when I was trying to play some audio. I used snd_pcm_open(&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); snd_pcm_hw_params_any(pcm_handle, hwparams); to open and set my PCM device for writting audio. But I have got lost when I tried to change my audio volume. I suppose that I need to get some mixer int snd_mixer_open (snd_mixer_t **mixer, int mode) but I could not find how to code that the new mixer will control the PCM device. Pleas is there any tutorial how to do it? Or can somebody show me any code snippet? I have tried to find any similar topic, but without success. Thanks for any advice Milan

1 0

[alsa-devel] Application supporting forward/rewind APIs
by Anuj Aggarwal 07 Oct '09

07 Oct '09

Hi, I want to try the forward/rewind APIs supported by the ALSA library. Can someone point to an ALSA application which exercises these APIs? Best Regards, Anuj Aggarwal

4 8

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 06 Oct '09

06 Oct '09

The following changes since commit 88439ac793934a47f47ad285656b63d09f5937c8: Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The new commits that aren't part of v2.6.32-rc3 (which is being merged up via fix/asoc here) are: Jonathan Cameron (1): ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Mark Brown (4): Merge branch 'for-2.6.32' into for-2.6.33 ASoC: Simplify code for DAPM widget updates ASoC: Push DAPM enumeration register change test out ASoC: Add virtual enumeration support for DAPM muxes Peter Ujfalusi (2): ASoC: Fix SND_SOC_DAPM_LINE handling ASoC: Improve the debugfs hierarchy

2 1

[alsa-devel] ALSA integration CS4281 PMC on PPC platform
by Harish Kumar 06 Oct '09

06 Oct '09

Hi, I am in the process of integrating CS4281 on PPC platform. I have included all the alsa driver and selected the CS4281 in my kernel configuration. The alsa driver is integrated as part of the kernel image. The kernel version is 2.6.15.7. I compiled the alsa-lib and utils and copied to the RFS and tried executing aplay /audio8k16S.wav which flagged the following error. LSA lib pcm.c:2099:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:547: audio open error: No such file or directory Pls find the attachement for the capture of debug messages from proc interface. Thanks and Regards Harish Kumar.B

1 1

[alsa-devel] hda_intel: SPDIF not detected on Asus P7P55D PRO
by Robert Hancock 06 Oct '09

06 Oct '09

I just set up a new system running an Asus P7P55D PRO motherboard. It looks like this uses the VIA VT1828S codec. The analog output seems to work but the SPDIF output isn't recognized. Originally tried 2.6.30.5, but Linus' current git doesn't seem any different. The device only gets detected as HDA Generic, which, from my reading of the code, ignores SPDIF outputs. I tried hacking patch_via.c to add the codec ID to match existing types like VT1708S and VT1702, each of those got the SPDIF output to show up, but no output showed up on the receiver, and also the analog output then stopped being recognized (looks like some node IDs don't match). Is there some extra support that needs to be added for this codec, or anything else I can try to get this working? The SPDIF does work in Windows 7 with just the standard Microsoft HDA driver.. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 The codec data seems to have references to the SPDIF output, but no device is actually getting created. The data is below. Any suggestions? cat /proc/asound/card0/codec#0 Codec: VIA ID 4441 Address: 0 Function Id: 0x1 Vendor Id: 0x11064441 Subsystem Id: 0x104383cf Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x08 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x2a 0x2a] Converter: stream=5, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x09 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0a [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0b [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x26 0x26] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0c [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x2a 0x2a] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0f [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x10 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x14 0x14] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x8b 0x8b] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x08 0x21 Node 0x19 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x09 0x21 Node 0x1a [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x0b 0x21 Node 0x1b [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x34 0x21 Node 0x1c [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Power: setting=D0, actual=D0 Connection: 2 0x35 0x21 Node 0x1d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1e [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x2c* 0x2b 0x2a 0x29 0x28 0x21 Node 0x1f [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x2c 0x2b* 0x2a 0x29 0x28 0x21 Node 0x20 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Power: setting=D0, actual=D0 Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x24 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x19 Node 0x26 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x0a Node 0x27 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1a Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1b Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19037: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x1c Node 0x2a [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x09* 0x0c Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x0a* 0x0c Node 0x2c [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x90331138: [Fixed] CD at Int N/A Conn = ATAPI, Color = Black DefAssociation = 0x3, Sequence = 0x8 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power: setting=D0, actual=D0 Node 0x2d [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000090: OUT HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 1 0x0e Node 0x2e [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074411f0: [Jack] SPDIF Out at Ext Rear Panel Conn = RCA, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x2f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x30 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x31 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x32 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x33 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x34 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 3 0x08* 0x0b 0x0c Node 0x35 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 3 0x08* 0x0b 0x0c

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[alsa-devel] [PATCH 1/1] ICE1712/24 - Change the Multi Track Peak control (level meters) from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: pavel <pavel(a)insite.cz> * PLEASE NOTE - this change requires the corresponding update of envy24control for ice1712 - kind of an ABI change. * The "Multi Track Peak" control is read-only level meters indicator. * The control is VERY confusing to most users since it is currently displayed in regular mixers. E.g. alsamixer ignores its read-only status and allows changing the levels with keys which makes no sense. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 56d8d67..c7cff6f 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2269,7 +2269,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ice1712_pro_peak_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3896fb9..28ca8a4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2160,7 +2160,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, -- 1.5.4.3

2 1

[alsa-devel] alsa-utils cross compilation error
by Harish Kumar 06 Oct '09

06 Oct '09

Hi, I am trying to cross compile the alsa utils for PPC and as well as arm architecture. the compilation of alsa-libs was successful with their respective toolchains. But configuring the alsa-util is failing. I am using alsa-utils-1.0.11rc2,alsa-lib-1.0.11rc2. I am getting the error checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. I feel i have to provide the path of the include files but not sure which path to be passed and how it needs to be passed to configure. Request your help to resolve the same. I invoked the configure using the following command and the output of the configure is as below. root@localhost alsa-utils-1.0.11rc2]# CC=arm-linux-gcc ./configure --target=arm-linux --host=i686-linux configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... gawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ANSI C... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld checking if the linker (/opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking whether NLS is requested... yes checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ANSI C... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. [2]+ Done gedit configure [root@localhost alsa-utils-1.0.11rc2]# Thanks and Regards Harish Kumar.B

1 0

[alsa-devel] [PATCH 1/1] Changing the Multi Track Peak control from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> * The "Multi Track Peak" control is now of PCM type, to avoid confusing users in other alsa mixers. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/envy24control/levelmeters.c b/envy24control/levelmeters.c index 3bf1a94..fa44979 100644 --- a/envy24control/levelmeters.c +++ b/envy24control/levelmeters.c @@ -261,9 +261,15 @@ void level_meters_reset_peaks(GtkButton *button, gpointer data) void level_meters_init(void) { + int err; + snd_ctl_elem_value_malloc(&peaks); - snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_PCM); snd_ctl_elem_value_set_name(peaks, "Multi Track Peak"); + if ((err = snd_ctl_elem_read(ctl, peaks)) < 0) + /* older ALSA driver, using MIXER type */ + snd_ctl_elem_value_set_interface(peaks, + SND_CTL_ELEM_IFACE_MIXER); } void level_meters_postinit(void) -- 1.5.4.3

1 0

[alsa-devel] [PATCH 1/1] Changing the Multi Track Peak control from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> * The "Multi Track Peak" control is now of PCM type, to avoid confusing users in other alsa mixers. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/envy24control/levelmeters.c b/envy24control/levelmeters.c index 3bf1a94..9bdaace 100644 --- a/envy24control/levelmeters.c +++ b/envy24control/levelmeters.c @@ -262,7 +262,7 @@ void level_meters_reset_peaks(GtkButton *button, gpointer data) void level_meters_init(void) { snd_ctl_elem_value_malloc(&peaks); - snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_PCM); snd_ctl_elem_value_set_name(peaks, "Multi Track Peak"); } -- 1.5.4.3

4 3

[alsa-devel] [PATCH 1/7] ak4114 - fix errors in output selector bits
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: Pavel Hofman <pavel.hofman(a)ivitera.com> * the previous version had a typo - values of AK4114_OPS10-12 were identical with AK4114_OPS00-02 * Since no cards actually use this feature, the bug was not identified earlier Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/ak4114.h b/include/ak4114.h index d293d36..3ce69fd 100644 --- a/include/ak4114.h +++ b/include/ak4114.h @@ -95,13 +95,13 @@ /* AK4114_REG_IO0 */ #define AK4114_TX1E (1<<7) /* TX1 Output Enable (1 = enable) */ -#define AK4114_OPS12 (1<<2) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS11 (1<<1) /* Output Though Data Selector for TX1 pin */ -#define AK4114_OPS10 (1<<0) /* Output Though Data Selector for TX1 pin */ +#define AK4114_OPS12 (1<<6) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS11 (1<<5) /* Output Data Selector for TX1 pin */ +#define AK4114_OPS10 (1<<4) /* Output Data Selector for TX1 pin */ #define AK4114_TX0E (1<<3) /* TX0 Output Enable (1 = enable) */ -#define AK4114_OPS02 (1<<2) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS01 (1<<1) /* Output Though Data Selector for TX0 pin */ -#define AK4114_OPS00 (1<<0) /* Output Though Data Selector for TX0 pin */ +#define AK4114_OPS02 (1<<2) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS01 (1<<1) /* Output Data Selector for TX0 pin */ +#define AK4114_OPS00 (1<<0) /* Output Data Selector for TX0 pin */ /* AK4114_REG_IO1 */ #define AK4114_EFH1 (1<<7) /* Interrupt 0 pin Hold */ -- 1.5.6.3

3 17

[alsa-devel] [PATCH 1/1] ICE1712/24 - Change the Multi Track Peak control (level meters) from MIXER to PCM type
by pavel.hofman＠ivitera.com 06 Oct '09

06 Oct '09

From: pavel <pavel(a)insite.cz> * The "Multi Track Peak" control is a read-only level meters indicator. * The control is VERY confusing to most users since it is currently displayed in regular mixers. E.g. alsamixer ignores its read-only status and allows changing the levels with keys which makes no sense. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 56d8d67..c7cff6f 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2269,7 +2269,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ice1712_pro_peak_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3896fb9..28ca8a4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2160,7 +2160,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, -- 1.5.4.3

2 1

[alsa-devel] Merge of 2.6.32-rc3 into fix/asoc
by Mark Brown 06 Oct '09

06 Oct '09

I've pushed an update of the fix/asoc branch which has 2.6.32-rc3 merged up to my git. This gets underlying platform support from other trees into the branch and will also (once it's merged up there) give topic/asoc access to things like the genirq improvements for I2C and SPI devices which were merged during the merge window. The following changes since commit e655a43544bd3c45a83da93b00a4b115b4fa758e: Jonathan Cameron (1): ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32-merge Changelog omitted - the only changes since your fix/asoc should be those from Linus' tree and the merge commit.

2 3

[alsa-devel] [PATCH] AC97 atmel: add support for AT91(common with AVR32).
by Sedji Gaouaou 06 Oct '09

06 Oct '09

This patch add AC97 support for ATMEL AT91 boards, using the AVR32 code. While AVR is using a DMA, the AT91 chips are using a Peripheral Data Controller. This is linked to the patch: http://www.arm.linux.org.uk/developer/patches/viewpatch.php?id=5640. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- It is based on Takashi git tree(sound-2.6/for-next). sound/atmel/Kconfig | 2 +- sound/atmel/ac97c.c | 354 +++++++++++++++++++++++++++++++++++++-------------- 2 files changed, 262 insertions(+), 94 deletions(-) diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig index 6c228a9..94de43a 100644 --- a/sound/atmel/Kconfig +++ b/sound/atmel/Kconfig @@ -12,7 +12,7 @@ config SND_ATMEL_AC97C tristate "Atmel AC97 Controller (AC97C) driver" select SND_PCM select SND_AC97_CODEC - depends on DW_DMAC && AVR32 + depends on (DW_DMAC && AVR32) || ARCH_AT91 help ALSA sound driver for the Atmel AC97 controller. diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 0c0f877..54d1365 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -13,6 +13,7 @@ #include <linux/device.h> #include <linux/dmaengine.h> #include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/module.h> @@ -31,6 +32,10 @@ #include <linux/dw_dmac.h> +#include <mach/cpu.h> +#include <mach/hardware.h> +#include <mach/gpio.h> + #include "ac97c.h" enum { @@ -63,6 +68,7 @@ struct atmel_ac97c { u64 cur_format; unsigned int cur_rate; unsigned long flags; + int period; /* Serialize access to opened variable */ spinlock_t lock; void __iomem *regs; @@ -241,10 +247,13 @@ static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, params_buffer_bytes(hw_params)); if (retval < 0) return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + + if (cpu_is_at32ap7000()) { + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -263,12 +272,14 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, retval = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) { + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -282,16 +293,20 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } return snd_pcm_lib_free_pages(substream); } static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } return snd_pcm_lib_free_pages(substream); } @@ -299,9 +314,11 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, OCA); int retval; + chip->period = 0; word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); /* assign channels to AC97C channel A */ @@ -324,7 +341,8 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: - word |= AC97C_CMR_CEM_LITTLE; + if (cpu_is_at32ap7000()) + word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ word &= ~(AC97C_CMR_CEM_LITTLE); @@ -363,9 +381,18 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_TX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_TO_DEVICE); + if (cpu_is_at32ap7000()) { + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TNCR); + } return retval; } @@ -374,9 +401,11 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, ICA); int retval; + chip->period = 0; word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); /* assign channels to AC97C channel A */ @@ -415,11 +444,15 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) word |= AC97C_CSR_OVRUN; ac97c_writel(chip, CAMR, word); - /* Enable channel A event interrupt */ word = ac97c_readl(chip, IMR); word |= AC97C_SR_CAEVT; - ac97c_writel(chip, IER, word); + ac97c_writel(chip, IER, /*word*/AC97C_SR_CAEVT); + + /* Enable channel A event interrupt */ + /*word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word);*/ /* set variable rate if needed */ if (runtime->rate != 48000) { @@ -438,9 +471,18 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_RX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_FROM_DEVICE); + if (cpu_is_at32ap7000()) { + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RNCR); + } return retval; } @@ -449,7 +491,7 @@ static int atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -458,15 +500,22 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.tx_chan); - if (retval) - goto out; - camr |= AC97C_CMR_CENA; + if (cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_TXTEN; + } + camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) + dw_dma_cyclic_stop(chip->dma.tx_chan); + else + ptcr = ATMEL_PDC_TXTDIS; if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -476,6 +525,8 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if (!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -484,7 +535,7 @@ static int atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -493,15 +544,22 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.rx_chan); - if (retval) - goto out; + if (cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_RXTEN; + } camr |= AC97C_CMR_CENA; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) + dw_dma_cyclic_stop(chip->dma.rx_chan); + else + ptcr = ATMEL_PDC_RXTDIS; if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -511,6 +569,8 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if (!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -523,7 +583,10 @@ atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + else + bytes = readl(chip->regs + ATMEL_PDC_TPR); bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -540,7 +603,10 @@ atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + else + bytes = readl(chip->regs + ATMEL_PDC_RPR); bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -578,20 +644,67 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) u32 sr = ac97c_readl(chip, SR); u32 casr = ac97c_readl(chip, CASR); u32 cosr = ac97c_readl(chip, COSR); + u32 camr = ac97c_readl(chip, CAMR); if (sr & AC97C_SR_CAEVT) { - dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + struct snd_pcm_runtime *runtime; + int offset, next_period, block_size; + dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", casr & AC97C_CSR_UNRUN ? " UNRUN" : "", casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", casr & AC97C_CSR_TXRDY ? " TXRDY" : "", !casr ? " NONE" : ""); + if (!cpu_is_at32ap7000()) { + if ((casr & camr) & AC97C_CSR_ENDTX) { + runtime = chip->playback_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_TNCR); + + snd_pcm_period_elapsed( + chip->playback_substream); + } + if ((casr & camr) & AC97C_CSR_ENDRX) { + runtime = chip->capture_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_RNCR); + snd_pcm_period_elapsed(chip->capture_substream); + } + } retval = IRQ_HANDLED; } if (sr & AC97C_SR_COEVT) { - dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", + dev_dbg(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", @@ -608,15 +721,50 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) return retval; } +static struct ac97_pcm at91_ac97_pcm_defs[] __devinitdata = { + /* Playback */ + { + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } }, + }, + /* PCM in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } } + }, + /* Mic in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = (1<<AC97_SLOT_MIC), + } } + }, +}; + static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) { struct snd_pcm *pcm; struct snd_pcm_hardware hw = atmel_ac97c_hw; - int capture, playback, retval; + int capture, playback, retval, err; capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + if (!cpu_is_at32ap7000()) { + err = snd_ac97_pcm_assign(chip->ac97_bus, + ARRAY_SIZE(at91_ac97_pcm_defs), + at91_ac97_pcm_defs); + if (err) + return err; + } retval = snd_pcm_new(chip->card, chip->card->shortname, chip->pdev->id, playback, capture, &pcm); if (retval) @@ -775,7 +923,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pclk = clk_get(&pdev->dev, "pclk"); + if (cpu_is_at32ap7000()) { + pclk = clk_get(&pdev->dev, "pclk"); + } else { + pclk = clk_get(&pdev->dev, "ac97_clk"); + } + if (IS_ERR(pclk)) { dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); @@ -844,43 +997,52 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) goto err_ac97_bus; } - if (pdata->rx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->rx_dws; - dma_cap_mask_t mask; + if (cpu_is_at32ap7000()) { + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; - dws->rx_reg = regs->start + AC97C_CARHR + 2; + dws->rx_reg = regs->start + AC97C_CARHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + chip->dma.rx_chan = dma_request_channel(mask, filter, + dws); - dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", dev_name(&chip->dma.rx_chan->dev->device)); - set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - } + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } - if (pdata->tx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->tx_dws; - dma_cap_mask_t mask; + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; - dws->tx_reg = regs->start + AC97C_CATHR + 2; + dws->tx_reg = regs->start + AC97C_CATHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + chip->dma.tx_chan = dma_request_channel(mask, filter, + dws); - dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", dev_name(&chip->dma.tx_chan->dev->device)); - set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - } + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } - if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && - !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { - dev_dbg(&pdev->dev, "DMA not available\n"); - retval = -ENODEV; - goto err_dma; + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + } else { + /* Just pretend that we have DMA channel(for at91 i is actually + * the PDC */ + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); } retval = atmel_ac97c_pcm_new(chip); @@ -897,20 +1059,22 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); - dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", - chip->regs); + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p, irq = %d\n", + chip->regs, irq); return 0; err_dma: - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } err_ac97_bus: snd_card_set_dev(card, NULL); @@ -934,12 +1098,13 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) struct snd_card *card = platform_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + } clk_disable(chip->pclk); - return 0; } @@ -949,11 +1114,12 @@ static int atmel_ac97c_resume(struct platform_device *pdev) struct atmel_ac97c *chip = card->private_data; clk_enable(chip->pclk); - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.tx_chan); - + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + } return 0; } #else @@ -978,14 +1144,16 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) iounmap(chip->regs); free_irq(chip->irq, chip); - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } snd_card_set_dev(card, NULL); snd_card_free(card); -- 1.5.3.7

1 0

[alsa-devel] ice1724 - Changing Level Meters from MIXER to PCM?
by Pavel Hofman 06 Oct '09

06 Oct '09

Hi, Following your discussion on moving low-level controls from MIXER to PCM types, I would like to suggest the following two patches: diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3896fb9..28ca8a4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2160,7 +2160,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, diff --git a/envy24control/levelmeters.c b/envy24control/levelmeters.c index 3bf1a94..9bdaace 100644 --- a/envy24control/levelmeters.c +++ b/envy24control/levelmeters.c @@ -262,7 +262,7 @@ void level_meters_reset_peaks(GtkButton *button, gpointer da void level_meters_init(void) { snd_ctl_elem_value_malloc(&peaks); - snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_value_set_interface(peaks, SND_CTL_ELEM_IFACE_PCM); snd_ctl_elem_value_set_name(peaks, "Multi Track Peak"); } The point is these controls are read-only, display incorrectly in alsamixer, and are very confusing for majority of users who do not read envy24 datasheet and the source code. Thanks for considering this IMO useful change. Regards, Pavel.

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[alsa-devel] [PATCHv3 7/7] ASoC: TWL6030: Detect power-up sequence completion
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. If NAUDINT irq is provided, CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. If AUDPWRON gpio line is provided but NAUDINT irq is not, then CODEC driver enables READYINT and polls on INTID register. If none of them are provided, then CODEC uses manual power sequences and disables all audio interrupts. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 61 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 1 + 2 files changed, 53 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 56fe136..3db7f1d 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -52,6 +52,7 @@ struct twl6030_data { int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; }; /* @@ -372,6 +373,7 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, static irqreturn_t twl6030_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; u8 intid; twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); @@ -391,7 +393,7 @@ static irqreturn_t twl6030_naudint_handler(int irq, void *data) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + complete(&priv->ready); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -626,11 +628,45 @@ static int twl6030_add_widgets(struct snd_soc_codec *codec) return 0; } +static int twl6030_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6030_data *priv = codec->private_data; + int time_left; + u8 intid; + + if (naudint) { + /* wait for ready interrupt with 48 ms timeout */ + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + } else { + /* retry 3 times only */ + for (time_left = 3; time_left > 0; time_left--) { + mdelay(16); + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, + TWL6030_REG_INTID); + if (intid & TWL6030_READYINT) + break; + } + } + + if (!time_left) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + + priv->codec_powered = 1; + + return 0; +} + static int twl6030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct twl6030_data *priv = codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -643,8 +679,10 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for power-up completion */ + ret = twl6030_power_up_completion(codec, naudint); + if (ret) + return ret; /* sync registers updated during power-up sequence */ twl6030_read(codec, TWL6030_REG_NCPCTL); @@ -653,12 +691,11 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -1068,6 +1105,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1090,10 +1128,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) if (ret) goto gpio2_err; } else { - dev_warn(codec->dev, - "no naudint irq, audio interrupts disabled\n"); - twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, - TWL6030_ALLINT_MSK); + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + ~TWL6030_READYMSK & TWL6030_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 786fd80..cdc3a3a 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6030_READYMSK 0x40 #define TWL6030_ALLINT_MSK 0x7B /* NCPCTL (0x05) fields */ -- 1.5.4.3

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[alsa-devel] [PATCHv3 6/7] ASoC: TWL6030: Enable audio interrupt
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 70 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl6030.h | 14 +++++++++ 2 files changed, 79 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 8548442..56fe136 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -46,6 +46,7 @@ struct twl6030_data { struct snd_soc_codec codec; int audpwron; + int naudint; int codec_powered; int pll; int non_lp; @@ -61,7 +62,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x7B, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -367,6 +368,39 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } +/* audio interrupt handler */ +static irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + u8 intid; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + dev_alert(codec->dev, "codec is ready\n"); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + return IRQ_HANDLED; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -993,19 +1027,23 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) struct twl_codec_data *twl_codec = pdev->dev.platform_data; struct snd_soc_codec *codec; struct twl6030_data *priv; - int audpwron; + int audpwron, naudint; int ret = 0; priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (twl_codec) + if (twl_codec) { audpwron = twl_codec->audpwron_gpio; - else + naudint = twl_codec->naudint_irq; + } else { audpwron = -EINVAL; + naudint = 0; + } priv->audpwron = audpwron; + priv->naudint = naudint; codec = &priv->codec; codec->dev = &pdev->dev; @@ -1043,13 +1081,28 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) priv->codec_powered = 0; } + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6030_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6030_codec", codec); + if (ret) + goto gpio2_err; + } else { + dev_warn(codec->dev, + "no naudint irq, audio interrupts disabled\n"); + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } + /* init vio registers */ twl6030_init_vio_regs(codec); /* power on device */ ret = twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret) - goto gpio2_err; + goto irq_err; ret = snd_soc_register_codec(codec); if (ret) @@ -1068,6 +1121,9 @@ dai_err: twl6030_codec = NULL; reg_err: twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); +irq_err: + if (naudint) + free_irq(naudint, codec); gpio2_err: if (gpio_is_valid(audpwron)) gpio_free(audpwron); @@ -1082,10 +1138,14 @@ static int __devexit twl6030_codec_remove(struct platform_device *pdev) { struct twl6030_data *priv = twl6030_codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; if (gpio_is_valid(audpwron)) gpio_free(audpwron); + if (naudint) + free_irq(naudint, twl6030_codec); + snd_soc_unregister_dai(&twl6030_dai); snd_soc_unregister_codec(twl6030_codec); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 87f1e1e..786fd80 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,20 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6030_ALLINT_MSK 0x7B + /* NCPCTL (0x05) fields */ #define TWL6030_NCPENA 0x01 -- 1.5.4.3

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[alsa-devel] [PATCHv3 5/7] ASoC: TWL6030: Add restrictions for low-power playback mode
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. Only in this mode the codec can support 44.1 and 48 kHz, low-power PLL should provide sysclk signal. Currently, handsfree DAC and driver are the only components that can prevent codec to enter to low-power playback mode. Other components like earphone driver, vibrator driver or loopback (which are not yet supported in the driver) can cause the same effect. In order to detect conflicting paths, CODEC driver supervises non-low-power widgets powered by DAPM mechanism, just at the point pcm trigger callback gets called. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 79 +++++++++++++++++++++++++++++++++++++++---- 1 files changed, 71 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 9fc4795..8548442 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -48,6 +48,7 @@ struct twl6030_data { int audpwron; int codec_powered; int pll; + int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; }; @@ -352,6 +353,20 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6030_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -485,10 +500,14 @@ static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { TWL6030_REG_HSLCTL, 0, 0), SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", TWL6030_REG_HSRCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", - TWL6030_REG_HFLCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", - TWL6030_REG_HFRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback switches */ SND_SOC_DAPM_SWITCH("HSDAC Left Playback", @@ -504,10 +523,14 @@ static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), SND_SOC_DAPM_SWITCH("Headset Right Driver", SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HFDAC Left PGA", @@ -668,6 +691,17 @@ static int twl6030_startup(struct snd_pcm_substream *substream, return -EINVAL; } + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && substream->stream) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, priv->sysclk_constraints); @@ -712,6 +746,34 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, return 0; } +static int twl6030_trigger(struct snd_pcm_substream *substream, + int cmd, 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; + struct twl6030_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* + * low-power playback mode is restricted + * for headset path only + */ + if ((priv->sysclk == 17640000) && priv->non_lp) { + dev_err(codec->dev, + "some enabled paths aren't supported at %dHz\n", + priv->sysclk); + return -EPERM; + } + break; + default: + break; + } + + return 0; +} + static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -822,6 +884,7 @@ static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .startup = twl6030_startup, .hw_params = twl6030_hw_params, + .trigger = twl6030_trigger, .set_sysclk = twl6030_set_dai_sysclk, }; -- 1.5.4.3

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[alsa-devel] [PATCHv3 4/7] ASoC: TWL6030: Add support for low-power PLL
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

TWL6030 codec sysclk can be provided by: low-power or high performance PLL. The low-power PLL takes a low-frequency input at 32,768 Hz and generates an approximate of 17.64 or 19.2 MHz. The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. Headset DAC and output driver should be configured according to the selected mode. 17.64 MHz sysclk is recommended only for headset low-power playback mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 232 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 16 +++ 2 files changed, 215 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index f97dec2..9fc4795 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -39,7 +39,7 @@ #include "twl6030.h" -#define TWL6030_RATES (SNDRV_PCM_RATE_48000) +#define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) /* codec private data */ @@ -47,6 +47,9 @@ struct twl6030_data { struct snd_soc_codec codec; int audpwron; int codec_powered; + int pll; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; }; /* @@ -326,6 +329,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) mdelay(10); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -607,55 +633,195 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, return 0; } +/* set of rates for each pll: low-power and high-performance */ + +static unsigned int lp_rates[] = { + 44100, + 48000, +}; + +static struct snd_pcm_hw_constraint_list lp_constraints = { + .count = ARRAY_SIZE(lp_rates), + .list = lp_rates, +}; + +static unsigned int hp_rates[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list hp_constraints = { + .count = ARRAY_SIZE(hp_rates), + .list = hp_rates, +}; + +static int twl6030_startup(struct snd_pcm_substream *substream, + 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; + struct twl6030_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + priv->sysclk_constraints); + + return 0; +} + +static int twl6030_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; + struct twl6030_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6030_HPPLL_ID) + return 0; + + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 44100: + lppllctl |= TWL6030_LPLLFIN; + priv->sysclk = 17640000; + break; + case 48000: + lppllctl &= ~TWL6030_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + return 0; +} + static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_data *priv = codec->private_data; u8 hppllctl, lppllctl; hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppllctl &= ~TWL6030_MCLK_MSK; - - switch (freq) { - case 12000000: - /* mclk input, pll enabled */ - hppllctl |= TWL6030_MCLK_12000KHZ | - TWL6030_HPLLSQRBP | - TWL6030_HPLLENA; - break; - case 19200000: - /* mclk input, pll disabled */ - hppllctl |= TWL6030_MCLK_19200KHZ | - TWL6030_HPLLSQRBP | - TWL6030_HPLLBP; - break; - case 26000000: - /* mclk input, pll enabled */ - hppllctl |= TWL6030_MCLK_26000KHZ | - TWL6030_HPLLSQRBP | - TWL6030_HPLLENA; + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + switch (freq) { + case 32768: + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + /* clk32k input requires low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* lppll divider */ + switch (priv->sysclk) { + case 17640000: + lppllctl |= TWL6030_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6030_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6030_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6030_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; break; - case 38400000: - /* clk slicer, pll disabled */ - hppllctl |= TWL6030_MCLK_38400KHZ | - TWL6030_HPLLSQRENA | - TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6030_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_12000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6030_MCLK_19200KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_26000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6030_MCLK_38400KHZ | + TWL6030_HPLLSQRENA | + TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6030_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; break; default: - dev_err(codec->dev, "unknown sysclk freq %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); - - /* Disable LPPLL and select HPPLL */ - lppllctl = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); - return 0; } static struct snd_soc_dai_ops twl6030_dai_ops = { + .startup = twl6030_startup, + .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, }; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 7375ae8..87f1e1e 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -100,10 +100,26 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + /* ACCCTL (0x2D) fields */ #define TWL6030_RESETSPLIT 0x04 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 + +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

1 0

[alsa-devel] [PATCHv3 3/7] ASoC: TWL6030: Manual power-up/down sequences
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

TWL6030 codec device can be powered-up/down through a specific register writes sequence. These sequences can be used when no gpio line is provided for AUDPWRON. When the codec is powered-up in this way, automatic power-down sequence (triggered by thermal shutdown) is not possible. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 112 ++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/twl6030.h | 16 ++++++ 2 files changed, 115 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 3c6c540..f97dec2 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -244,6 +244,88 @@ static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) } } +/* twl6030 codec manual power-up sequence */ +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6030_NCPENA | TWL6030_NCPOPEN; + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); +} + +/* twl6030 codec manual power-down sequence */ +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6030_NCPENA | TWL6030_NCPOPEN); + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -480,12 +562,15 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-up sequence latency */ mdelay(16); - } - /* sync registers updated during power-up sequence */ - twl6030_read(codec, TWL6030_REG_NCPCTL); - twl6030_read(codec, TWL6030_REG_LDOCTL); - twl6030_read(codec, TWL6030_REG_LPPLLCTL); + /* sync registers updated during power-up sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6030_power_up(codec); + } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); @@ -502,12 +587,16 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-down sequence latency */ udelay(500); - } - /* sync registers updated during power-down sequence */ - twl6030_read(codec, TWL6030_REG_NCPCTL); - twl6030_read(codec, TWL6030_REG_LDOCTL); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + /* sync registers updated during power-down sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6030_power_down(codec); + } priv->codec_powered = 0; break; @@ -723,9 +812,6 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) goto gpio2_err; priv->codec_powered = 0; - } else { - /* if no gpio is provided, then assume its always on */ - priv->codec_powered = 1; } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index d99d469..7375ae8 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,18 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* NCPCTL (0x05) fields */ + +#define TWL6030_NCPENA 0x01 +#define TWL6030_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6030_LSLDOENA 0x01 +#define TWL6030_HSLDOENA 0x04 +#define TWL6030_REFENA 0x40 +#define TWL6030_OSCENA 0x80 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -88,6 +100,10 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* ACCCTL (0x2D) fields */ + +#define TWL6030_RESETSPLIT 0x04 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

1 0

[alsa-devel] [PATCHv3 2/7] ASoC: TWL6030: Add twl6030 codec driver
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary PDM-based digital audio interface. Audio paths supported are: - Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right - Output: Headset Left/Right, Handsfree Left/Right Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 824 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 94 +++++ 4 files changed, 924 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..0effb52 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL6030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -114,6 +115,9 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TWL4030 tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..b70c8a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -13,6 +13,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -50,6 +51,7 @@ 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_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..3c6c540 --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,824 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_48000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6030_data { + struct snd_soc_codec codec; + int audpwron; + int codec_powered; +}; + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x7B, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * twl6030 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6030_vio_reg[TWL6030_VIOREGNUM] = { + TWL6030_REG_ASICID, + TWL6030_REG_ASICREV, + TWL6030_REG_INTID, + TWL6030_REG_INTMR, + TWL6030_REG_NCPCTL, + TWL6030_REG_LDOCTL, + TWL6030_REG_AMICBCTL, + TWL6030_REG_DMICBCTL, + TWL6030_REG_HKCTL1, + TWL6030_REG_HKCTL2, + TWL6030_REG_GPOCTL, + TWL6030_REG_TRIM1, + TWL6030_REG_TRIM2, + TWL6030_REG_TRIM3, + TWL6030_REG_HSOTRIM, + TWL6030_REG_HFOTRIM, + TWL6030_REG_ACCCTL, + TWL6030_REG_STATUS, +}; + +/* + * twl6030 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6030_vdd_reg[TWL6030_VDDREGNUM] = { + TWL6030_REG_HPPLLCTL, + TWL6030_REG_LPPLLCTL, + TWL6030_REG_LPPLLDIV, + TWL6030_REG_MICLCTL, + TWL6030_REG_MICRCTL, + TWL6030_REG_MICGAIN, + TWL6030_REG_LINEGAIN, + TWL6030_REG_HSLCTL, + TWL6030_REG_HSRCTL, + TWL6030_REG_HSGAIN, + TWL6030_REG_EARCTL, + TWL6030_REG_HFLCTL, + TWL6030_REG_HFLGAIN, + TWL6030_REG_HFRCTL, + TWL6030_REG_HFRGAIN, + TWL6030_REG_VIBCTLL, + TWL6030_REG_VIBDATL, + TWL6030_REG_VIBCTLR, + TWL6030_REG_VIBDATR, + TWL6030_REG_ALB, + TWL6030_REG_DLB, +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL6030_MODULE_AUDIO, value, reg); +} + +static void twl6030_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + for (i = 0; i < TWL6030_VIOREGNUM; i++) { + reg = twl6030_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + if ((reg == TWL6030_REG_ASICID) || + (reg == TWL6030_REG_ASICREV) || + (reg == TWL6030_REG_STATUS)) + continue; + twl6030_write(codec, reg, cache[reg]); + } +} + +static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6030_VDDREGNUM; i++) { + reg = twl6030_vdd_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6030_data *priv = codec->private_data; + int audpwron = priv->audpwron; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* power-up sequence latency */ + mdelay(16); + } + + /* sync registers updated during power-up sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + + /* initialize vdd/vss registers with reg_cache */ + twl6030_init_vdd_regs(codec); + + priv->codec_powered = 1; + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + } + + /* sync registers updated during power-down sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 hppllctl, lppllctl; + + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppllctl &= ~TWL6030_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_12000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6030_MCLK_19200KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_26000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6030_MCLK_38400KHZ | + TWL6030_HPLLSQRENA | + TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk freq %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + + /* Disable LPPLL and select HPPLL */ + lppllctl = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .set_sysclk = twl6030_set_dai_sysclk, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6030_codec; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6030_codec); + + codec = twl6030_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __devinit twl6030_codec_probe(struct platform_device *pdev) +{ + struct twl_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6030_data *priv; + int audpwron; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) + audpwron = twl_codec->audpwron_gpio; + else + audpwron = -EINVAL; + + priv->audpwron = audpwron; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6030_dai.dev = &pdev->dev; + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } else { + /* if no gpio is provided, then assume its always on */ + priv->codec_powered = 1; + } + + /* init vio registers */ + twl6030_init_vio_regs(codec); + + /* power on device */ + ret = twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto gpio2_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6030_codec = codec; + + ret = snd_soc_register_dai(&twl6030_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6030_codec = NULL; +reg_err: + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6030_codec_remove(struct platform_device *pdev) +{ + struct twl6030_data *priv = twl6030_codec->private_data; + int audpwron = priv->audpwron; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + snd_soc_unregister_dai(&twl6030_dai); + snd_soc_unregister_codec(twl6030_codec); + + kfree(twl6030_codec); + twl6030_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int twl6030_codec_suspend(struct platform_device *pdev, + pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int twl6030_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define twl6030_codec_suspend NULL +#define twl6030_codec_resume NULL +#endif + +static struct platform_driver twl6030_codec_driver = { + .driver = { + .name = "twl6030_codec", + .owner = THIS_MODULE, + }, + .probe = twl6030_codec_probe, + .remove = __devexit_p(twl6030_codec_remove), + .suspend = twl6030_codec_suspend, + .resume = twl6030_codec_resume, +}; + +static int __init twl6030_codec_init(void) +{ + return platform_driver_register(&twl6030_codec_driver); +} +module_init(twl6030_codec_init); + +static void __exit twl6030_codec_exit(void) +{ + platform_driver_unregister(&twl6030_codec_driver); +} +module_exit(twl6030_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..d99d469 --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,94 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +#define TWL6030_VIOREGNUM 18 +#define TWL6030_VDDREGNUM 21 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_LPLLFIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +#endif /* End of __TWL6030_H__ */ -- 1.5.4.3

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[alsa-devel] [PATCHv3 0/7] TWL6030 audio codec initial support
by Lopez Cruz, Misael 06 Oct '09

06 Oct '09

Following patch series adds initial support for TWL6030 codec driver. Changes from v2: - CODEC now handles scenarios where no platform data is passed - Removed GPIOLIB dependency of codec driver - Removed magic values, register reads instead - Added runtime constraints depending on sysclk - Added more constraints for switching to low-power mode - Use request_threaded_irq instead (oneshot support required) - wait_for_completion with timeout - Changelog now gives a better explanation of features and corresponding changes Thanks, -Misa --- Misael Lopez Cruz (7): OMAP4: PMIC: Add support for twl6030 codec ASoC: TWL6030: Add twl6030 codec driver ASoC: TWL6030: Manual power-up/down sequences ASoC: TWL6030: Add support for low-power PLL ASoC: TWL6030: Add restrictions for low-power playback mode ASoC: TWL6030: Enable audio interrupt ASoC: TWL6030: Detect power-up sequence completion drivers/mfd/twl-core.c | 15 + include/linux/i2c/twl.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 1242 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 141 +++++ 6 files changed, 1411 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h

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[alsa-devel] [PATCH 18/31] HDA patch_via.c: Add 48K sample rate limit for S/PDIF streams.
by Li Bo 06 Oct '09

06 Oct '09

[ALSA] HDA VIA: Add 48K sample rate limit for S/PDIF streams. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:11:02.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:04.000000000 +0800 @@ -1228,6 +1228,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -1241,6 +1242,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, }; static int via_build_controls(struct hda_codec *codec) @@ -2042,6 +2044,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -2657,6 +2660,7 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -3137,6 +3141,7 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -3581,6 +3586,7 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open,

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[alsa-devel] [PATCH 0/31] HDA patch_via.c update
by Li Bo 06 Oct '09

06 Oct '09

Hi! Following are a series of patches for patch_via.c, started from git clone of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 recently. Please have a review and give comments, thank you! Outlook Access Web cannot send plain-text mail, so my gmail is used instead. You can also contact with me with LoganLi(a)viatech.com.cn, or mail to HaraldWelte(a)viatech.com and lydiawang(a)viatech.com.cn.

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[alsa-devel] Bug report, multichannel LPCM over nvidia HDMI not working
by Chris Howells 05 Oct '09

05 Oct '09

Hi, I am trying to get multi-channel LPCM working over HDMI, from an Acer Revo (Nvidia Ion) to a Sony STR-DH800 amp. 2 channel LPCM is working perfectly, as is bitstreaming AC3 and DTS (with xbmc). The latest I've tried is with kernel 2.6.31.1. I am testing with speaker-test. I have found that: 'speaker-test -D hdmi -c2' works perfect. The amp lights up show 2 channel LPCM. Sound is from the front left and front right. http://chrishowells.co.uk/speakertest2ch.jpg 'speaker-test -D hdmi -c6' or (-c8) does not work. The amp realises that there is some PCM sounds, but the amp does not seem to recognise the stream properly, *however*, sound does come out of the left and right. http://chrishowells.co.uk/speakertest6ch.jpg The version of speaker-test is from Alsa 1.0.20. Here's the results of running alsa-info.sh http://www.alsa-project.org/db/?f=d90ca1ce07bf20af3241114b11fb31f79ed63fec The 'Misc = NO_PRESENCE' struck me as being a bit odd. The Revo has been upgraded to the A2 bios (didn't work with A1 either), and the Nvidia drivers are 190.32. I've spent quite a few days trying to get this work, including alsa 1.0.20 and 1.0.21 on ubuntu 9.04 with and without the 8ch patch, Ubuntu karmic beta standard, as well as patching alsa 1.0.20 it with the 8ch patch. Most grateful for any help.

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[alsa-devel] [PATCH] sscape: force AD1848 codec mode on old Soundscape
by Krzysztof Helt 05 Oct '09

05 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The second version: do not set joystick variable to zeros (false). BTW. The OSS sscape from the 2.6 kernel cannot handle my Media FX card. It seems that only Soundscape PnP may work (after fixing existing bugs). I prefer removing it completely. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index cf98525..6de56d1 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1639,6 +1639,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - MPU-401 IRQ # (PnP setup) dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index c739374..e348f8b 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -54,6 +54,7 @@ static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS] __devinitdata; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -79,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -145,7 +149,6 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; unsigned char midi_vol; }; @@ -815,7 +818,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -836,12 +838,30 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; + + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (sscape->type == SSCAPE_VIVO) + case SSCAPE_VIVO: port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -927,6 +947,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; + int val; const char *name; /* @@ -1026,6 +1047,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT -- 1.6.0.3 ---------------------------------------------------------------------- Nie boisz sie wampirow! A wi�c Graj >>> http://link.interia.pl/f2390

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[alsa-devel] step-by-step usb-snd-audio instructions?
by Anna Keenan 05 Oct '09

05 Oct '09

Hi there ALSA people, I'm very new to Linux, running an eee pc 1000H withLinux Easypeasy 1.5 and the new skype beta. I love my USB headset - I'm frequently on international skype conference calls and on other operating systems I've found that nothing else gives me a clear sound. But I am apparently missing the usb-snd-audio driver, as it doesn't come packaged with Easypeasy, so my beloved headset won't function. Skype audio is generally also really crackly when i'm using the normal microphone and speakers, but i think that's a different problem. Does anyone out there have step-by-step instructions on how to add the snd-usb-audio driver? I've found no user-friendly 'how to' online, despite hours trawling (and posting on) forums... I would *so* love some help or even a response telling where else I should be looking for help... Anna K -- Anna Keenan Youth Climate Advocate +31 6 45043 155 (Netherlands) +61 (0)4 1979 2263 (Global) Skype ID: anna.c.keenan "If you don't change course, you'll end up where you're heading." www.itsgettinghotinhere.org http://climatechangeperspectives.blogspot.com

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 05 Oct '09

05 Oct '09

The following changes since commit 834eb6c599a8efa1fe9b77d469562e0c78c876e1: Mark Brown (1): Merge remote branch 'takashi/fix/asoc' into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Jonathan Cameron (1): ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Peter Ujfalusi (1): ASoC: Fix SND_SOC_DAPM_LINE handling sound/soc/codecs/wm8940.c | 2 +- sound/soc/soc-dapm.c | 5 +++-- 2 files changed, 4 insertions(+), 3 deletions(-)

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Re: [alsa-devel] Asterisk and ALSA/Jack
by Pavel Hofman 05 Oct '09

05 Oct '09

Fabien COMTE wrote: > Hi, > > I want to use Asterisk with ALSA / Jack audio. > > I tried the next configurations : > > 1 ) app_jack.so -> does not work. > > 2 ) chan_alsa.so without jackd started > my .asoundrc : > pcm.!default > { > type hw > card 1 > } > pcm.jack1 { > type jack > playback_ports { > 0 alsa_pcm:playback_1 > 1 alsa_pcm:playback_2 > } > capture_ports { > 0 alsa_pcm:capture_1 > 1 alsa_pcm:capture_2 > } > } > > pcm.jack1b { > type plug > device jack1b > server linux > > > slave { > pcm "jack1" > rate 8000 > format FLOAT > channels 2 > } > } > > ctl.mixer0 { > type hw > card 1 > } > > in alsa.conf : > input_device=default > output_device=default > > That works fine. > > 3 ) chan_alsa.so with jackd started > Jackd launch command : > jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 > --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 > --outchannels 2 --dither triangular & > > my .asoundrc : > > pcm.!default > { > type plug > slave { > pcm "jack1" > rate 8000 > format FLOAT > channels 2 > } > } > pcm.jack1 { > type jack > playback_ports { > 0 alsa_pcm:playback_1 > 1 alsa_pcm:playback_2 > } > capture_ports { > 0 alsa_pcm:capture_1 > 1 alsa_pcm:capture_2 > } > } > > pcm.jack1b { > type plug > device jack1b > server linux > > > slave { > pcm "jack1" > rate 8000 > format FLOAT > channels 2 > } > } > > > ctl.mixer0 { > type hw > card 1 > } > > in alsa.conf : > input_device=default > output_device=default > > It doesn't work. Asterisk outputs are : > [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open > failed: No such file or directory > [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open > failed: No such file or directory > [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:272 soundcard_init: Problem > opening ALSA I/O devices > > But aplay my_test_file.wav works fine and arecord too... > > Does anyone have an idea ? > Does asterisk run under the same user as the .asoundrc is defined for? Since your card is nb. 1, perhaps asterisk uses a default configuration for card0 which may not be present on your system. Pavel.

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[alsa-devel] [PATCH] sscape: force AD1848 codec mode on old Soundscape
by Krzysztof Helt 05 Oct '09

05 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Even more OSS code moved into the alsa. I have tested it on early SPEA Media FX card. The only problem is that MIDI playback is not played timely. However, I observe this problem on cards with OPL4 chipset and Dream chipset. I assume it is more general problem with mpu401 and MIDI playback. If Theodore Allen test this on Ensoniq Soundscape card the OSS sscape driver can be removed. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index cf98525..6de56d1 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1639,6 +1639,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - MPU-401 IRQ # (PnP setup) dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index c739374..e348f8b 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -54,6 +54,7 @@ static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = false }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -79,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -145,7 +149,6 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; unsigned char midi_vol; }; @@ -815,7 +818,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -836,12 +838,30 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; + + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (sscape->type == SSCAPE_VIVO) + case SSCAPE_VIVO: port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -927,6 +947,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; + int val; const char *name; /* @@ -1026,6 +1047,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT -- 1.6.0.3 --------------------------------------------------------------- Przygotuj si� na przysz�o��! Kliknij >>> http://link.interia.pl/f238a

2 1

[alsa-devel] [PATCH] sound: Don't assume i2c device probing always succeeds
by Jean Delvare 05 Oct '09

05 Oct '09

If i2c device probing fails, then there is no driver to dereference after calling i2c_new_device(). Stop assuming that probing will always succeed, to avoid NULL pointer dereferences. We have an easier access to the driver anyway. Reported-by: Tim Shepard <shep(a)alum.mit.edu> Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> --- The code is similar to the one in therm_adt746x, for which Tim reported a real-world oops, so it should be fixed ASAP. sound/aoa/codecs/onyx.c | 4 +++- sound/aoa/codecs/tas.c | 4 +++- sound/ppc/keywest.c | 4 +++- 3 files changed, 9 insertions(+), 3 deletions(-) --- linux-2.6.32-rc1.orig/sound/aoa/codecs/onyx.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/aoa/codecs/onyx.c 2009-09-30 15:13:58.000000000 +0200 @@ -996,6 +996,8 @@ static void onyx_exit_codec(struct aoa_c onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } +static struct i2c_driver onyx_driver; + static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) @@ -1027,7 +1029,7 @@ static int onyx_create(struct i2c_adapte * Let i2c-core delete that device on driver removal. * This is safe because i2c-core holds the core_lock mutex for us. */ - list_add_tail(&client->detected, &client->driver->clients); + list_add_tail(&client->detected, &onyx_driver.clients); return 0; } --- linux-2.6.32-rc1.orig/sound/aoa/codecs/tas.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/aoa/codecs/tas.c 2009-09-30 15:13:58.000000000 +0200 @@ -882,6 +882,8 @@ static void tas_exit_codec(struct aoa_co } +static struct i2c_driver tas_driver; + static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) @@ -902,7 +904,7 @@ static int tas_create(struct i2c_adapter * Let i2c-core delete that device on driver removal. * This is safe because i2c-core holds the core_lock mutex for us. */ - list_add_tail(&client->detected, &client->driver->clients); + list_add_tail(&client->detected, &tas_driver.clients); return 0; } --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-09-30 15:13:12.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-09-30 15:13:58.000000000 +0200 @@ -40,6 +40,8 @@ static int keywest_probe(struct i2c_clie return 0; } +struct i2c_driver keywest_driver; + /* * This is kind of a hack, best would be to turn powermac to fixed i2c * bus numbers and declare the sound device as part of platform @@ -65,7 +67,7 @@ static int keywest_attach_adapter(struct * This is safe because i2c-core holds the core_lock mutex for us. */ list_add_tail(&keywest_ctx->client->detected, - &keywest_ctx->client->driver->clients); + &keywest_driver.clients); return 0; } -- Jean Delvare

3 9

[alsa-devel] hda-intel via codec on optical sound [PARTIAL SUCCESS]
by AL13N 05 Oct '09

05 Oct '09

forget about all of it, i'm a stupid doofus ------------------------------------------------------- First of all, I HAVE optical sound output. secondly, the hda analyzer still reports the connector as not_present thirdly, i probably have found a bug: * it happens when playing passthrough DD and DTS, as well as when using a52 * what happens is that the front right is playing on the LFE (muffled, but that's probably normal) * so when speakertesting -t wav -c 6 to a52 which points to hw0,1 , my optical surround set recognizes it as dolby sound, no problem. * the sound "front right" as well as the LFE comes from the subwoofer. * keep in mind this also happens in passthrough. What i've done, is i'm using alsa-driver-1.0.21 (after seeing some digital out changes between 19 and 21.) It is weird that it still says not_connected in codec#0, but the "-D iec958:CARD=SB" seems to work fine. can anyone help me find the bug? or tell me this is not a bug? Sincerely, AL13N ----------------------------------------------------------------- I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] hda-intel via codec on optical sound [PARTIAL SUCCESS]
by AL13N 05 Oct '09

05 Oct '09

First of all, I HAVE optical sound output. secondly, the hda analyzer still reports the connector as not_present thirdly, i probably have found a bug: * it happens when playing passthrough DD and DTS, as well as when using a52 * what happens is that the front right is playing on the LFE (muffled, but that's probably normal) * so when speakertesting -t wav -c 6 to a52 which points to hw0,1 , my optical surround set recognizes it as dolby sound, no problem. * the sound "front right" as well as the LFE comes from the subwoofer. * keep in mind this also happens in passthrough. What i've done, is i'm using alsa-driver-1.0.21 (after seeing some digital out changes between 19 and 21.) It is weird that it still says not_connected in codec#0, but the "-D iec958:CARD=SB" seems to work fine. can anyone help me find the bug? or tell me this is not a bug? Sincerely, AL13N ----------------------------------------------------------------- I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] hda-intel via codec on optical sound
by AL13N 04 Oct '09

04 Oct '09

I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] hda-intel via codec on optical sound
by AL13N 04 Oct '09

04 Oct '09

I have posted this, but no reaction on alsa-user... (i'm not on the alsa-devel list) Maybe this is a bug somewhere, that is what i wish to know. I have a onboard hda_intel card. it has 2 codes and is split in 2 parts: * via * atihdmi I want to have sound in the optical output SPDIF/TOSLINK whatever you want to call it. hdanalyzer shows 2 digital pins: 0x20 ext SPDIF ATAPI 0x21 ext SPDIF Optical but both are marked as 'not present'. the "2d card has only one pin and one audio out" on the motherboard there is an internal SPDIF connector, which may be the ATAPI one. I believe that the SPDIF Optical is the correct one. Why is it not present? is that something software related? or is the audio chip broken? that is what i really want to know at this point. is it broken or can it be fixed? mail sent to alsa-user: -------------------------------------------------------------------------------------------------- i have only an optical link to surround set, (which has worked before) to test out. the toslink looks red on the surround endpoint. the surround system always says: "no data" (analog stereo or DD or DTS causes it to show a diff text) when i play something, (be it stereo, DD or DTS sample) at iec958 , the /proc/asound/card0/pcm1p/sub0/status shows effectively that the sound IS running there. I've set all soundlevels to 75% and unmuted everything. i've looked at the codec#0 file and that may give something: node 0x21 (digital with optical connector) shows as not present. it looks like there are on seperate models for the VIA codec. how would i go solve this? Sincerely, AL13N aplay -l ----------- **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 codec#0: ------------ Codec: VIA VT1708S Address: 0 Vendor Id: 0x11060397 Subsystem Id: 0x1043836c Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=1, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x13 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x17 Node 0x14 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x1e Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x16 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x97 0x97] [0x97 0x97] Power: setting=D0, actual=D0 Connection: 7 0x10 0x1f 0x1a 0x1b 0x1e 0x1d 0x25 Node 0x17 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D0 Connection: 6 0x1f 0x1a* 0x1b 0x1e 0x1d 0x16 Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x11 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1a [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x0181303e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x1c [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x16 Node 0x1d [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1e [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19038: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x8 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x16* 0x25 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x503701f7: [N/A] CD at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x7 Misc = NO_PRESENCE Pin-ctls: 0x00: Power: setting=D0, actual=D0 Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x074521f0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power: setting=D0, actual=D0 Connection: 1 0x15 Node 0x22 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x26 Node 0x23 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x27 Node 0x24 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x26 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x27 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x25 -- AL13N is my name and head-biting is my game.

1 0

[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 04 Oct '09

04 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version of this patch with improved issues pointed by Takashi. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..c739374 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -147,12 +147,6 @@ struct soundscape { struct snd_wss *chip; struct snd_mpu401 *mpu; - /* - * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL - */ - spinlock_t fwlock; - unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +158,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +168,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR "sscape: Failed to allocate " + "%lu bytes for DMA\n", + size); return NULL; } } @@ -482,7 +473,8 @@ static int upload_dma_data(struct soundscape *s, */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR "sscape: DMA upload has timed out\n"); + snd_printk(KERN_ERR + "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } @@ -504,10 +496,12 @@ static int upload_dma_data(struct soundscape *s, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); + snd_printk(KERN_ERR "sscape: No response " + "from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); + snd_printk(KERN_ERR + "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -536,7 +530,7 @@ static int sscape_upload_bootblock(struct snd_card *card) ret = request_firmware(&init_fw, "scope.cod", card->dev); if (ret < 0) { - snd_printk(KERN_ERR "Error loading scope.cod"); + snd_printk(KERN_ERR "sscape: Error loading scope.cod"); return ret; } ret = upload_dma_data(sscape, init_fw->data, init_fw->size); @@ -554,7 +548,8 @@ static int sscape_upload_bootblock(struct snd_card *card) data &= 0xf; if (ret == 0 && data > 7) { - snd_printk(KERN_ERR "timeout reading firmware version\n"); + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); ret = -EAGAIN; } @@ -575,12 +570,13 @@ static int sscape_upload_microcode(struct snd_card *card, int version) err = request_firmware(&init_fw, name, card->dev); if (err < 0) { - snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", + version); return err; } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", init_fw->size >> 10); release_firmware(init_fw); @@ -750,7 +746,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); if ((inb(wss_io) & 0x80) != 0) goto _done; @@ -774,7 +769,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); /* * SoundScape successfully detected! @@ -794,38 +788,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ static int mpu401_open(struct snd_mpu401 * mpu) { - int err; - if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); - err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); + snd_printk(KERN_ERR "sscape: MIDI disabled, " + "please load firmware\n"); + return -ENODEV; } - return err; -} - -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); + return 0; } /* @@ -845,8 +814,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -993,13 +960,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1036,7 +1003,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,8 +1040,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + snd_printk(KERN_ERR + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); goto _release_dma; } strcpy(card->driver, "SoundScape"); @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "sscape: device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

2 1

Re: [alsa-devel] Asterisk and ALSA/Jack
by Fabien COMTE 03 Oct '09

03 Oct '09

Hi, I want to use Asterisk with ALSA / Jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } pcm.jack1b { type plug device jack1b server linux slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } ctl.mixer0 { type hw card 1 } in alsa.conf : input_device=default output_device=default That works fine. 3 ) chan_alsa.so with jackd started Jackd launch command : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & my .asoundrc : pcm.!default { type plug slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } pcm.jack1b { type plug device jack1b server linux slave { pcm "jack1" rate 8000 format FLOAT channels 2 } } ctl.mixer0 { type hw card 1 } in alsa.conf : input_device=default output_device=default It doesn't work. Asterisk outputs are : [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open failed: No such file or directory [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:175 alsa_card_init: snd_pcm_open failed: No such file or directory [Oct 3 10:34:50] ERROR[1364]: chan_alsa.c:272 soundcard_init: Problem opening ALSA I/O devices But aplay my_test_file.wav works fine and arecord too... Does anyone have an idea ? Thanks, Fabien

1 0

Re: [alsa-devel] My Codec: IDT 92HD75B3X5 issues
by Munzir Taha (منذر طه) 02 Oct '09

02 Oct '09

On Yau al-Thulatha 09 Shawwal 1430 5:37:04 pm you wrote: > At Tue, 29 Sep 2009 16:24:29 +0300, > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > On Yau al-Thulatha 09 Shawwal 1430 9:24:19 am Takashi Iwai wrote: > > > At Tue, 29 Sep 2009 07:40:56 +0300, > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: > > > > > At Mon, 28 Sep 2009 06:58:46 +0300, > > > > > > > > > > Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > > > > > > > > =?utf-8?q?_=D8=B7=D9=87?=) wrote: > > > > > > Hi sirs, > > > > > > These are the issues I have with my Laptop HP Pavilion dv6 1225ee > > > > > > regarding audio support. > > > > > > > > > > > > 1. No sense for headphone jack (jack_present?), when headphone is > > > > > > plugged (present_int_hp?) > > > > > > > > > > > > 2. There headphone mic jack is not working. The one next to the > > > > > > webcam is always recording even when I change the "Mic Jack" > > > > > > control from "Mic In" to "Line In" > > > > > > > > > > Make sure that you use alsa-driver-snapshot tarball instead of > > > > > 1.0.21 released version. There are many fixes since 1.0.21 > > > > > release. > > > > > > > > > > > > > > > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-dr > > > > >iver -sn apshot.tar.gz > > > > > > > > Thanks a lot Takashi, things are better now. However, when I tested > > > > the internal mic it's very noisy. I played a bit with the capture and > > > > digital controls but still compared with Windows, it's lame. > > > > > > OK, could you give alsa-info.sh output at this state? > > > Please run with --no-upload option, and attach the generated file. > > > > Attached. > > Try to run: > > % amixer -c0 set Capture cap 12dB > % amixer -c0 set "Digital Mic" 10dB > % amixer -c0 set Digital 0dB > > Any difference? No, still there is a clear shshshshshsh sound in the background. Please check the attachment for a short arecord -f cd without me speaking on the mic and still you can hear this noise very clear. Also attached is the alsa-info report at the time of recording -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

2 13

[alsa-devel] OFFTOPIC atkbd.c: Use 'setkeycodes... (Was: Re: My Codec: IDT 92HD75B3X5 issues)
by Simos Xenitellis 02 Oct '09

02 Oct '09

On Tue, Sep 29, 2009 at 7:40 AM, Munzir Taha (منذر طه) <munzirtaha(a)gmail.com> wrote: > On Yaum al-Ithnain 08 Shawwal 1430 11:41:52 am Takashi Iwai wrote: >> At Mon, 28 Sep 2009 06:58:46 +0300, >> >> Munzir Taha (=?utf-8?q?=D9=85=D9=86=D8=B0=D8=B1?= > =?utf-8?q?_=D8=B7=D9=87?=) wrote: >> > Hi sirs, >> > These are the issues I have with my Laptop HP Pavilion dv6 1225ee >> > regarding audio support. >> > >> > 1. No sense for headphone jack (jack_present?), when headphone is plugged >> > (present_int_hp?) >> > >> > 2. There headphone mic jack is not working. The one next to the webcam is >> > always recording even when I change the "Mic Jack" control from "Mic In" >> > to "Line In" >> >> Make sure that you use alsa-driver-snapshot tarball instead of 1.0.21 >> released version. There are many fixes since 1.0.21 release. > > >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sn >> apshot.tar.gz > > Thanks a lot Takashi, things are better now. However, when I tested the > internal mic it's very noisy. I played a bit with the capture and digital > controls but still compared with Windows, it's lame. > >> > 3. The mute/unmute LED doesn't change color to RED when its muted like in >> > Windows. >> >> This isn't supported yet as we have no hardware information. >> Likely one of GPIO pins corresponds to the LED on/off. You can try >> hda-verb to change the GPIO data bit. > > Ok, I will try it and report back. > >> > Touching that LED shows this in /var/log/messages: >> > >> > Sep 28 06:53:47 dv6 kernel: [ 9197.986789] atkbd.c: Unknown key released >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9197.986791] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> > Sep 28 06:53:47 dv6 kernel: [ 9198.353568] atkbd.c: Unknown key pressed >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9198.353572] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> > Sep 28 06:53:47 dv6 kernel: [ 9198.359205] atkbd.c: Unknown key released >> > (translated set 2, code 0x94 on isa0060/serio0). >> > Sep 28 06:53:47 dv6 kernel: [ 9198.359208] atkbd.c: Use 'setkeycodes e014 >> > <keycode>' to make it known. >> >> It's not about the sound driver but the input layer, as you can find. >> You can defien the scan code 0xe014 by yourself. > > You are right. What's the best component to report this bug against? kernel? Those keys are a 'HAL' thing. Until recently there has been a website by the HAL developer with detailed instructions. Search for 'HAL Quirk Site' on Google to view the site in Google cache. The website was not updated for some time (http://www.mail-archive.com/pm-utils@lists.freedesktop.org/msg01594.html) and being obsolete, it was probably taken down. Your changes would need to make it eventually to the hal-info repository, http://cgit.freedesktop.org/hal-info/ Use the Freedesktop Bugzilla, product 'hal' and component 'hal-info' when reporting, https://bugs.freedesktop.org/enter_bug.cgi?product=hal Simos

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[alsa-devel] [PATCH] Soc: wm8940: Fix check on error code form snd_soc_codec_set_cache_io
by Jonathan Cameron 02 Oct '09

02 Oct '09

Soc: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Fix for typo in commit 8d50e447d19fec64adebeef55f2b60d695435412 ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs Signed-off-by: Jonathan Cameron <jic23(a)cam.ac.uk> --- sound/soc/codecs/wm8940.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae..1ef2454 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940, codec->reg_cache = &wm8940->reg_cache; ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); - if (ret == 0) { + if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } -- 1.6.3.3

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[alsa-devel] big latency and buffering question
by Achim Herrmann 02 Oct '09

02 Oct '09

Hi all, I'm completely new to sound and ALSA development and I have to develop an application which is capturing periodically a small amount of audio data (e.g. 64 frames), doing some processing and doing playback of the result. I started with ALSA example "pcm.c" to see if ALSA can meet these requirements. During investigation i found out, that up to a choosen buffer size of 140ms, generated sine sound is fine. Below 140ms, sound is stuttering, independent of call method I'm using. For buffer size of 140ms I found out, that target buffer size is 6174 frames (at 44.1kHz sample rate), period size is 3087 frames and loop inside write_and_poll function is calculated every 70 ms and 3087 frames are written. Think this is doublebuffering then, right? But if I now choose a buffersize of 139 ms or lower, target buffer size and period size are always the same. So no double buffering anymore and sound is stuttering. So, currently I'm far away from goal. I'd like to have a buffer size of e.g. 128 frames (2ms). Thus, period size should be 64 frames and loop should be calculated every 1ms. (If necessary I would use Linux-RT) If I choose a buffersize of 2ms, sound is far away from being sine :-) - Can I use ALSA for a application described in first sentence above? - What do I have to change in pcm.c to be able to create a sine signal using a buffersize of e.g. 128 frames and a period size of 64 frames? Any hints are appreciated Thanks in advance Achim

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[alsa-devel] [PATCH] ASoC: Improve the debugfs hierarchy
by Peter Ujfalusi 02 Oct '09

02 Oct '09

Change the way the debugfs entries are created: If the codec->dev is valid, than use: debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/ if the codec->dev is NULL: debugfs/asoc/{codec->name}/ as root for the debugfs entries. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The patch applies on top of sound-2.6 topic/asoc commit 88439ac793934a47f47ad285656b63d09f5937c8 ASoC: add support for multiple cards/codecs in debugfs sound/soc/soc-core.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e4ab36d..1dec9d2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,8 +1256,12 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { char codec_root[128]; - snprintf(codec_root, sizeof(codec_root), - "%s-%s", dev_name(codec->socdev->dev), codec->name); + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); codec->debugfs_codec_root = debugfs_create_dir(codec_root, debugfs_root); -- 1.6.5.rc2

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 02 Oct '09

02 Oct '09

The following changes since commit c877c25170e2655d519b29e91d6c91d5d1a72a6f: Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Mark Brown (1): Merge remote branch 'takashi/fix/asoc' into for-2.6.32 sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++++++---------------------- 1 files changed, 49 insertions(+), 49 deletions(-)

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[alsa-devel] [PATCH 06/34] don't use __devexit_p to wrap hal2_remove
by Uwe Kleine-König 02 Oct '09

02 Oct '09

The function hal2_remove is defined using __exit, so don't use __devexit_p but __exit_p to wrap it. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/mips/hal2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index c52691c..72523a7 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -926,7 +926,7 @@ static int __exit hal2_remove(struct platform_device *pdev) static struct platform_driver hal2_driver = { .probe = hal2_probe, - .remove = __devexit_p(hal2_remove), + .remove = __exit_p(hal2_remove), .driver = { .name = "sgihal2", .owner = THIS_MODULE, -- 1.6.4.3

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[alsa-devel] [PATCH] Don't check if period size is equal to buffer size
by Lubomir Rintel 02 Oct '09

02 Oct '09

At least some drivers seem to allow a single period and those which don't would correct us anyway. Moreover, Pulseaudio with tsched sets this frequently. --- aplay/aplay.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c7c82a1..4fa36cb 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1044,11 +1044,6 @@ static void set_params(void) } snd_pcm_hw_params_get_period_size(params, &chunk_size, 0); snd_pcm_hw_params_get_buffer_size(params, &buffer_size); - if (chunk_size == buffer_size) { - error(_("Can't use period equal to buffer size (%lu == %lu)"), - chunk_size, buffer_size); - exit(EXIT_FAILURE); - } snd_pcm_sw_params_current(handle, swparams); if (avail_min < 0) n = chunk_size; -- 1.6.2.5

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[alsa-devel] [PATCH] sound: Make keywest_driver static
by Jean Delvare 02 Oct '09

02 Oct '09

From: Jean Delvare <khali(a)linux-fr.org> Subject: sound: Make keywest_driver static I can't see any reason for struct i2c_driver keywest_driver to not be static. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/keywest.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-10-01 09:45:18.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-10-01 18:05:06.000000000 +0200 @@ -93,7 +93,7 @@ static const struct i2c_device_id keywes { } }; -struct i2c_driver keywest_driver = { +static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, -- Jean Delvare

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[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 02 Oct '09

02 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, remove redundant "sscape:" prefix in messages printed by snd_printk and convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sscape.c | 85 +++++++++++++--------------------------------------- 1 files changed, 21 insertions(+), 64 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..ece2bb9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -149,9 +149,8 @@ struct soundscape { /* * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL + * its firmware. */ - spinlock_t fwlock; unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +163,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +173,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR + "Failed to allocate %lu bytes for DMA\n", + size); return NULL; } } @@ -797,37 +793,13 @@ static int mpu401_open(struct snd_mpu401 * mpu) int err; if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); + snd_printk(KERN_ERR "MIDI disabled, please load firmware\n"); err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); } return err; } -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); -} - /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ @@ -845,8 +817,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -908,21 +878,20 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) { - snd_printk(KERN_ERR "sscape: No PCM device " - "for AD1845 chip\n"); + snd_printk(KERN_ERR "No PCM device for AD1845 chip\n"); goto _error; } err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR "sscape: No mixer device " + snd_printk(KERN_ERR "No mixer device " "for AD1845 chip\n"); goto _error; } if (chip->hardware != WSS_HW_AD1848) { err = snd_wss_timer(chip, 0, NULL); if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " + snd_printk(KERN_ERR "No timer device " "for AD1845 chip\n"); goto _error; } @@ -932,7 +901,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip)); if (err < 0) { - snd_printk(KERN_ERR "sscape: Could not create " + snd_printk(KERN_ERR "Could not create " "MIDI mixer control\n"); goto _error; } @@ -968,15 +937,14 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR - "sscape: can't grab port 0x%lx\n", port[dev]); + snd_printk(KERN_ERR "can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; if (sscape->type == SSCAPE_VIVO) { wss_res = request_region(wss_port[dev], 4, "SoundScape"); if (!wss_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n", + snd_printk(KERN_ERR "can't grab port 0x%lx\n", wss_port[dev]); err = -EBUSY; goto _release_region; @@ -988,18 +956,18 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ err = request_dma(dma[dev], "SoundScape"); if (err < 0) { - snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]); + snd_printk(KERN_ERR "can't grab DMA %d\n", dma[dev]); goto _release_region; } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1030,13 +998,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", irq[dev]); return -ENXIO; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,7 +1041,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + snd_printk(KERN_ERR "No AD1845 device at 0x%lx, IRQ %d\n", wss_port[dev], irq[dev]); goto _release_dma; } @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- 2012 rok wcale nie musi by� tragiczny! Kliknij >>> http://link.interia.pl/f238b

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[alsa-devel] Patches for alsactl init db
by Dan Chen 02 Oct '09

02 Oct '09

Hi, Attached are two patches against alsa-utils.git HEAD that create an intel8x0 init db file for ICH4. This is necessary from bug reports in Debian and Ubuntu that state that External Amplifier must be muted to have audible (analog) playback by default on the Sony VAIO VGN-T350P. Thanks, Dan

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[alsa-devel] Help! Playing multiple audio stream using alsa libraries
by Arjun Sambamoorthy 02 Oct '09

02 Oct '09

Hello all, I am trying to play two audio streams using alsa libraries at the same time, but I am not able to do so. I could play a single audio stream though. I get bi-directional G711 MU Law RTP streams and I am using the sample playback capture code from the tutorial section for playing it. I am using the Non blocking and interleaved mode for playing the streams. The alsa library plays both the audio streams for a few seconds but it is incomprehensible and after few seconds I get the error "write to audio interface failed (broken pipe)". I also tried opening two pcm handles for each stream to the same device ("default"), but the second handle is unusable gives the same error "write to audio interface failed (broken pipe)". I have the same problem playing two audio streams to the /dev/dsp device. I am pretty sure I am not doing it the right way. Any help is highly appreciated or pointer to links that explains how to play multiple streams using alsa. Thanks Arjunen

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[alsa-devel] [PATCH 1/3] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 01 Oct '09

01 Oct '09

Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> -- This applies to Kevin's temp/asoc branch --- arch/arm/mach-davinci/include/mach/asp.h | 6 ++ sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++-------- 2 files changed, 61 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..a3d2aa1 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -51,6 +51,12 @@ struct snd_platform_data { u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned disable_channel_combine:1; /* McASP specific fields */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..081b2d4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { .name = "I2S PCM Stereo out", }; @@ -113,6 +130,27 @@ struct davinci_mcbsp_dev { u32 pcr; struct clk *clk; struct davinci_pcm_dma_params *dma_params[2]; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned disable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -359,6 +397,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -388,27 +428,22 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - - dma_params->acnt = dma_params->data_type; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && !dev->disable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -524,7 +559,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->disable_channel_combine = pdata->disable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

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[alsa-devel] A few fix patches for ASoC
by Takashi Iwai 01 Oct '09

01 Oct '09

Mark, I applied a few fix patches to fix/asoc and topic/asoc branches for build errors I found with cross-build tests in this morning. Check below and sync to your tree. commit bb26276744a80d066681836f4d49c70010b129d6 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 07:39:45 2009 +0200 ASoC: Fix build errors of wm8711.c with SPI commit c877c25170e2655d519b29e91d6c91d5d1a72a6f Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:33:47 2009 +0200 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 commit 140318aaa924ce9664ff59366993228cf1547f1d Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 1 08:40:32 2009 +0200 ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c

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[alsa-devel] LEFT or RIGHT Channel in alsa mixer
by Navink Singh 01 Oct '09

01 Oct '09

Hi Developers, I need to configure my audio device such as to use only LEFT or RIGHT Channel. I have carried out following steps to proceed with 1) Opened PCM playback Device 2) Configured its hardware and Software parameters 3) Open mixer 4) Attached Mixer to playback device 5) Registered Mixer 6) Loaded the mixer 7) Get the first element of mixer A) Now when i use following API, snd_mixer_selem_is_playback_mono(aElem); i found mixer is using MONO channel. But i am configuring my PCM Device for 2 channels I am using snd_mixer_selem_set_playback_switch() API to set right or Left channel to MUTE {Does not work} If i set all channels to Mute {Works fine} snd_mixer_selem_set_playback_switch_all (aElem, 0) Also i am able to set SND_MIXER_SCHN_MONO to mute {works fine} snd_mixer_selem_set_playback_switch(aElem, SND_MIXER_SCHN_MONO, 0) B) There is also an API in my MIXER driver to set the right and left balance, but i did not find any API in Application to do so {Is there any such API in Application}. Thanks in Advance, Navin Kumar Singh Tata Consultancy Services DLF Akruti Info Park, PN 28, Rajiv Gandhi Infotech Park Hinjewadi, PHASE II Pune - 411057,Maharashtra India Ph:- +91 20 66752458 Cell:- +919860227357 Mailto: navink.singh(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 01 Oct '09

01 Oct '09

The following changes since commit 140318aaa924ce9664ff59366993228cf1547f1d: Takashi Iwai (1): ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Chaithrika U S (1): ASoC: DaVinci: McASP FIFO related updates Lopez Cruz, Misael (1): ASoC: Add PDM DAI format definition Mark Brown (6): ASoC: Clean up WM8974 PLL configuration Merge branch 'upstream/wm8974' into for-2.6.33 ASoC: Factor out analogue platform data from WM8993 Merge remote branch 'takashi/fix/asoc' into for-2.6.32 Merge remote branch 'takashi/topic/asoc' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 include/sound/soc-dai.h | 1 + include/sound/soc.h | 1 + sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++------------------ sound/soc/codecs/wm8974.c | 23 +++++---- sound/soc/codecs/wm8993.c | 36 +++---------- sound/soc/codecs/wm_hubs.c | 35 +++++++++++++ sound/soc/codecs/wm_hubs.h | 5 ++ sound/soc/davinci/davinci-i2s.c | 2 + sound/soc/davinci/davinci-mcasp.c | 17 +++---- sound/soc/davinci/davinci-pcm.c | 21 +++++++- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/pxa/Kconfig | 2 +- sound/soc/soc-core.c | 26 +++++++--- 13 files changed, 161 insertions(+), 107 deletions(-)

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[alsa-devel] Tascam
by humbert.olivier.1＠free.fr 01 Oct '09

01 Oct '09

Hi all in first thanks for your work, it permit us to make music with GNU Linux :) I'm Olivier from France. I work on linuxMAO.org website. It's a french website for music assisted by computer. In this website, i make an article for make work the tascam us-224 on GNU Linux. Sometime I was angry because all is'nt recognize on my sound card. (Like 4 faders for 4 tracks). So I make a letter in what i said something like that : "Hey Tascam, it's cool for you ? For me it's cool, because i buy your product and it's a good one, and now, i play music with my friend and children and many people. But, all doesn't work with my OS, GNU/Linux it's damage because you don't see that many people on linux and other OS, have to go to your concurent (like M-audio or RME) because they're more open than you. blablabla blablabla ..." They don't answer. ... :/ Another boy make another one few month ago. ... They don't answer too. ... :/ ... Last week, a new boy email me that, he was very happy because my website page for the TASCAM us 224 help him to make work on linux. (it was great, i'm happy) So i speak with him and said to him "hey, you can make an email to Tascam (in fact sennheiser in france, the official distributor for tascam products) and say to them "hello boys, can you give us the source code or specifications blablabla" " ... and this one, he answered positively ! The sennheiser man said to us that he speak with japanese guys who said "what they needs?". So, i'm not developper I search for the (one or many) people who made the tascam us 224 (and 122 and 448 i think) driver for ALSA system. because, we've got an open way to make it better. Or another one motived ! The ball is here now. Thanks for read and for your work. Olivier

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[alsa-devel] [PATCH 0/1 - try2] ASoC: debugfs support improvement
by Peter Ujfalusi 01 Oct '09

01 Oct '09

Hello, I have noticed that the support for multiple cards/codecs via the debugfs is kind of broken. The second codec, which tries to create the debugfs files would emit the following messages: ASoC: Failed to create codec register debugfs file Failed to create pop time debugfs file ... The following patch fixes this by adding a directory per codec to host the files. The additional directory is named as: {dev_name(socdev->dev)}-{codec->name} for example: debugfs/asoc/soc-audio.0-twl4030/... So it is easier to find the correct codec we are looking for. Note: The patch accidentally fixes one trailing space in the soc_pcm_apply_symmetry function. --- Peter Ujfalusi (1): ASoC: add support for multiple cards/codecs in debugfs include/sound/soc.h | 1 + sound/soc/soc-core.c | 28 ++++++++++++++++++++-------- 2 files changed, 21 insertions(+), 8 deletions(-)

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[alsa-devel] [PATCH] ASoC: Use codec mutex in dapm_set_pga()
by ext-eero.nurkkala＠nokia.com 01 Oct '09

01 Oct '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> codec->mutex seems required if widget->* values are altered. Moreover, snd_soc_dapm_put_volsw() may alter widget->saved_value. dapm_set_pga() uses widget->saved_value in a for loop which now has a distant chance of getting out of control. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> --- sound/soc/soc-dapm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f79711b..d6fb6c4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -349,6 +349,7 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + mutex_lock(&widget->codec->mutex); if (power) { int i; /* power up has happended, increase volume to last level */ @@ -373,6 +374,7 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) } widget->muted = 1; } + mutex_unlock(&widget->codec->mutex); } return 0; } -- 1.6.0.4

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[alsa-devel] adding hrtimer based delay to ioplug ALSA plugin on ARM9 (AT91) platform ?
by Stefan Schoenleitner 01 Oct '09

01 Oct '09

Hi, I'm currently developing an ALSA ioplug plugin which should transfer audio samples over UART to a DSP (I already wrote that in a previous post). The DSP wants to receive 156 - 164 audio samples with a sampling rate of 8 kHz and a format of S16_LE each 20 msec (which is more or less the number of audio samples multiplied by the period length of the sampling rate). My idea is to add a high precision delay to the transfer callback of the ALSA plugin. Unfortunately, I found out that the hrtimer resolution on the system is as low as 10ms: ------------------------------------------------------------------- kernel: hrtimer resolution: 0 sec, 10000000 nsec, loops_per_jiffy: 448512 ------------------------------------------------------------------- * Does this mean that if I use nanosleep for example each delay has to be an integral multiple of the hrtimer precision and for example a delay of 15ms would not be supported (since it is no integral multiple of 10ms) ? * It is possible to increase the precision (e.g. by changing the kernel configuration) ? * Do you know a better solution for the task, where the task is to send and receive audio samples to/from the DSP over UART and the timing is critical ? cheers, Stefan

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