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[alsa-devel] ALSA Driver DB Says to Contact You if I have This Hardware... So Here I am.
by Bugs Bane 20 Aug '08

20 Aug '08

Hi there, I have an Edirol 100FX. It has two modes, standard and advanced. The driver DB says it's currently supported for standard mode only, but to get in touch if I have this device so advanced mode can be worked on. I'm wondering what info I can supply to be of help? lsusb reads: Bus 004 Device 002: ID 0582:0046 Roland Corp. Bugsbane

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Re: [alsa-devel] [snd-bt87x]: seems cannot powerup after powerdown...
by Trent Piepho 19 Aug '08

19 Aug '08

On Tue, 19 Aug 2008, Dmitry Butskoy wrote: > Takashi Iwai wrote: > > Added Clemens and alsa-devel to Cc. > > > > At Mon, 18 Aug 2008 19:22:20 +0400, > > Dmitry Butskoy wrote: > > > >> Hi, > >> > >> Recently snd-bt87x driver has started to use CTL_A_PWRDN bit, see > >> http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.26.y.git;a=commi… > >> > >> Unfortunately, it produces a regression, at least in my case. > >> > >> I have "AVerMedia TVPhone 98" card, lspci is: Which PCM device are you using?

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Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45
by S. Aguinaga 19 Aug '08

19 Aug '08

============================================== Trying to figure out where to add this "model=macpro" and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? below are the only files I have. /etc/modprobe.d/ anaconda blacklist-visor i915modeset blacklist floppy-pnp modprobe.conf.dist ============================================== Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), S. Aguinaga wrote: > > S. Aguinaga wrote: > > > > Hello Stan, > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > 2. aplay does not produce sound, period! > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > sudo cat /var/log/messages | grep ALSA > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped (snip) These messages are harmless. It's just information from the driver, enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > I think your problem is a misidentification of your sound chip. The > model being used for your 889A appears to be incompatible, thus all > the errors above. It appears that the macpro ALC885 is being selected > before it gets to the choice for the ALC889A. Perhaps a developer can > look into this. Well, ALC882, ALC885 and ALC889A are almost compatible. In the driver, they are all handled in the same routine. There are a few models for intel macs with ALC88x. Try to specify either model=macpro, model=mbp3 or model=imac24 as the module option of snd-hda-intel driver. Takashi

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[alsa-devel] [PATCH] Remove OSS driver for SGI HAL2 audio device
by Thomas Bogendoerfer 19 Aug '08

19 Aug '08

With the restructering of the indy button handling the old OSS HAL2 driver got broken. Since there is a new ALSA driver for HAL2, the experimental OSS driver is obsolete and will be removed by this patch. Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- sound/oss/Kconfig | 7 - sound/oss/Makefile | 1 - sound/oss/hal2.c | 1558 ---------------------------------------------------- sound/oss/hal2.h | 248 --------- 4 files changed, 0 insertions(+), 1814 deletions(-) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index d4fafb6..cbee9dc 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -24,13 +24,6 @@ config SOUND_VWSND <file:Documentation/sound/oss/vwsnd> for more info on this driver's capabilities. -config SOUND_HAL2 - tristate "SGI HAL2 sound (EXPERIMENTAL)" - depends on SGI_IP22 && EXPERIMENTAL - help - Say Y or M if you have an SGI Indy or Indigo2 system and want to be able to - use its on-board A2 audio system. - config SOUND_AU1550_AC97 tristate "Au1550/Au1200 AC97 Sound" depends on SOC_AU1550 || SOC_AU1200 diff --git a/sound/oss/Makefile b/sound/oss/Makefile index c611514..e0ae4d4 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -10,7 +10,6 @@ obj-$(CONFIG_SOUND_OSS) += sound.o # Please leave it as is, cause the link order is significant ! obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o -obj-$(CONFIG_SOUND_HAL2) += hal2.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o diff --git a/sound/oss/hal2.c b/sound/oss/hal2.c deleted file mode 100644 index a94b9df..0000000 --- a/sound/oss/hal2.c +++ /dev/null @@ -1,1558 +0,0 @@ -/* - * Driver for A2 audio system used in SGI machines - * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis(a)linux-mips.org> - * - * Based on Ulf Carlsson's code. - * - * 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., 675 Mass Ave, Cambridge, MA 02139, USA. - * - * Supported devices: - * /dev/dsp standard dsp device, (mostly) OSS compatible - * /dev/mixer standard mixer device, (mostly) OSS compatible - * - */ -#include <linux/kernel.h> -#include <linux/module.h> -#include <linux/sched.h> -#include <linux/init.h> -#include <linux/slab.h> -#include <linux/poll.h> -#include <linux/interrupt.h> -#include <linux/dma-mapping.h> -#include <linux/sound.h> -#include <linux/soundcard.h> -#include <linux/mutex.h> - - -#include <asm/io.h> -#include <asm/sgi/hpc3.h> -#include <asm/sgi/ip22.h> - -#include "hal2.h" - -#if 0 -#define DEBUG(args...) printk(args) -#else -#define DEBUG(args...) -#endif - -#if 0 -#define DEBUG_MIX(args...) printk(args) -#else -#define DEBUG_MIX(args...) -#endif - -/* - * Before touching these look how it works. It is a bit unusual I know, - * but it helps to keep things simple. This driver is considered complete - * and I won't add any new features although hardware has many cool - * capabilities. - * (Historical note: HAL2 driver was first written by Ulf Carlsson - ALSA - * 0.3 running with 2.2.x kernel. Then ALSA changed completely and it - * seemed easier to me to write OSS driver from scratch - this one. Now - * when ALSA is official part of 2.6 kernel it's time to write ALSA driver - * using (hopefully) final version of ALSA interface) - */ -#define H2_BLOCK_SIZE 1024 -#define H2_ADC_BUFSIZE 8192 -#define H2_DAC_BUFSIZE 16834 - -struct hal2_pbus { - struct hpc3_pbus_dmacregs *pbus; - int pbusnr; - unsigned int ctrl; /* Current state of pbus->pbdma_ctrl */ -}; - -struct hal2_desc { - struct hpc_dma_desc desc; - u32 cnt; /* don't touch, it is also padding */ -}; - -struct hal2_codec { - unsigned char *buffer; - struct hal2_desc *desc; - int desc_count; - int tail, head; /* tail index, head index */ - struct hal2_pbus pbus; - unsigned int format; /* Audio data format */ - int voices; /* mono/stereo */ - unsigned int sample_rate; - unsigned int master; /* Master frequency */ - unsigned short mod; /* MOD value */ - unsigned short inc; /* INC value */ - - wait_queue_head_t dma_wait; - spinlock_t lock; - struct mutex sem; - - int usecount; /* recording and playback are - * independent */ -}; - -#define H2_MIX_OUTPUT_ATT 0 -#define H2_MIX_INPUT_GAIN 1 -#define H2_MIXERS 2 -struct hal2_mixer { - int modcnt; - unsigned int master; - unsigned int volume[H2_MIXERS]; -}; - -struct hal2_card { - int dev_dsp; /* audio device */ - int dev_mixer; /* mixer device */ - int dev_midi; /* midi device */ - - struct hal2_ctl_regs *ctl_regs; /* HAL2 ctl registers */ - struct hal2_aes_regs *aes_regs; /* HAL2 aes registers */ - struct hal2_vol_regs *vol_regs; /* HAL2 vol registers */ - struct hal2_syn_regs *syn_regs; /* HAL2 syn registers */ - - struct hal2_codec dac; - struct hal2_codec adc; - struct hal2_mixer mixer; -}; - -#define H2_INDIRECT_WAIT(regs) while (regs->isr & H2_ISR_TSTATUS); - -#define H2_READ_ADDR(addr) (addr | (1<<7)) -#define H2_WRITE_ADDR(addr) (addr) - -static char *hal2str = "HAL2"; - -/* - * I doubt anyone has a machine with two HAL2 cards. It's possible to - * have two HPC's, so it is probably possible to have two HAL2 cards. - * Try to deal with it, but note that it is not tested. - */ -#define MAXCARDS 2 -static struct hal2_card* hal2_card[MAXCARDS]; - -static const struct { - unsigned char idx:4, avail:1; -} mixtable[SOUND_MIXER_NRDEVICES] = { - [SOUND_MIXER_PCM] = { H2_MIX_OUTPUT_ATT, 1 }, /* voice */ - [SOUND_MIXER_MIC] = { H2_MIX_INPUT_GAIN, 1 }, /* mic */ -}; - -#define H2_SUPPORTED_FORMATS (AFMT_S16_LE | AFMT_S16_BE) - -static inline void hal2_isr_write(struct hal2_card *hal2, u16 val) -{ - hal2->ctl_regs->isr = val; -} - -static inline u16 hal2_isr_look(struct hal2_card *hal2) -{ - return hal2->ctl_regs->isr; -} - -static inline u16 hal2_rev_look(struct hal2_card *hal2) -{ - return hal2->ctl_regs->rev; -} - -#ifdef HAL2_DUMP_REGS -static u16 hal2_i_look16(struct hal2_card *hal2, u16 addr) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - return regs->idr0; -} -#endif - -static u32 hal2_i_look32(struct hal2_card *hal2, u16 addr) -{ - u32 ret; - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - ret = regs->idr0 & 0xffff; - regs->iar = H2_READ_ADDR(addr | 0x1); - H2_INDIRECT_WAIT(regs); - ret |= (regs->idr0 & 0xffff) << 16; - return ret; -} - -static void hal2_i_write16(struct hal2_card *hal2, u16 addr, u16 val) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->idr0 = val; - regs->idr1 = 0; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_write32(struct hal2_card *hal2, u16 addr, u32 val) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->idr0 = val & 0xffff; - regs->idr1 = val >> 16; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_setbit16(struct hal2_card *hal2, u16 addr, u16 bit) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - regs->idr0 = (regs->idr0 & 0xffff) | bit; - regs->idr1 = 0; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_setbit32(struct hal2_card *hal2, u16 addr, u32 bit) -{ - u32 tmp; - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - tmp = (regs->idr0 & 0xffff) | (regs->idr1 << 16) | bit; - regs->idr0 = tmp & 0xffff; - regs->idr1 = tmp >> 16; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -static void hal2_i_clearbit16(struct hal2_card *hal2, u16 addr, u16 bit) -{ - struct hal2_ctl_regs *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - regs->idr0 = (regs->idr0 & 0xffff) & ~bit; - regs->idr1 = 0; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} - -#if 0 -static void hal2_i_clearbit32(struct hal2_card *hal2, u16 addr, u32 bit) -{ - u32 tmp; - hal2_ctl_regs_t *regs = hal2->ctl_regs; - - regs->iar = H2_READ_ADDR(addr); - H2_INDIRECT_WAIT(regs); - tmp = ((regs->idr0 & 0xffff) | (regs->idr1 << 16)) & ~bit; - regs->idr0 = tmp & 0xffff; - regs->idr1 = tmp >> 16; - regs->idr2 = 0; - regs->idr3 = 0; - regs->iar = H2_WRITE_ADDR(addr); - H2_INDIRECT_WAIT(regs); -} -#endif - -#ifdef HAL2_DUMP_REGS -static void hal2_dump_regs(struct hal2_card *hal2) -{ - DEBUG("isr: %08hx ", hal2_isr_look(hal2)); - DEBUG("rev: %08hx\n", hal2_rev_look(hal2)); - DEBUG("relay: %04hx\n", hal2_i_look16(hal2, H2I_RELAY_C)); - DEBUG("port en: %04hx ", hal2_i_look16(hal2, H2I_DMA_PORT_EN)); - DEBUG("dma end: %04hx ", hal2_i_look16(hal2, H2I_DMA_END)); - DEBUG("dma drv: %04hx\n", hal2_i_look16(hal2, H2I_DMA_DRV)); - DEBUG("syn ctl: %04hx ", hal2_i_look16(hal2, H2I_SYNTH_C)); - DEBUG("aesrx ctl: %04hx ", hal2_i_look16(hal2, H2I_AESRX_C)); - DEBUG("aestx ctl: %04hx ", hal2_i_look16(hal2, H2I_AESTX_C)); - DEBUG("dac ctl1: %04hx ", hal2_i_look16(hal2, H2I_ADC_C1)); - DEBUG("dac ctl2: %08x ", hal2_i_look32(hal2, H2I_ADC_C2)); - DEBUG("adc ctl1: %04hx ", hal2_i_look16(hal2, H2I_DAC_C1)); - DEBUG("adc ctl2: %08x ", hal2_i_look32(hal2, H2I_DAC_C2)); - DEBUG("syn map: %04hx\n", hal2_i_look16(hal2, H2I_SYNTH_MAP_C)); - DEBUG("bres1 ctl1: %04hx ", hal2_i_look16(hal2, H2I_BRES1_C1)); - DEBUG("bres1 ctl2: %04x ", hal2_i_look32(hal2, H2I_BRES1_C2)); - DEBUG("bres2 ctl1: %04hx ", hal2_i_look16(hal2, H2I_BRES2_C1)); - DEBUG("bres2 ctl2: %04x ", hal2_i_look32(hal2, H2I_BRES2_C2)); - DEBUG("bres3 ctl1: %04hx ", hal2_i_look16(hal2, H2I_BRES3_C1)); - DEBUG("bres3 ctl2: %04x\n", hal2_i_look32(hal2, H2I_BRES3_C2)); -} -#endif - -static struct hal2_card* hal2_dsp_find_card(int minor) -{ - int i; - - for (i = 0; i < MAXCARDS; i++) - if (hal2_card[i] != NULL && hal2_card[i]->dev_dsp == minor) - return hal2_card[i]; - return NULL; -} - -static struct hal2_card* hal2_mixer_find_card(int minor) -{ - int i; - - for (i = 0; i < MAXCARDS; i++) - if (hal2_card[i] != NULL && hal2_card[i]->dev_mixer == minor) - return hal2_card[i]; - return NULL; -} - -static void hal2_inc_head(struct hal2_codec *codec) -{ - codec->head++; - if (codec->head == codec->desc_count) - codec->head = 0; -} - -static void hal2_inc_tail(struct hal2_codec *codec) -{ - codec->tail++; - if (codec->tail == codec->desc_count) - codec->tail = 0; -} - -static void hal2_dac_interrupt(struct hal2_codec *dac) -{ - int running; - - spin_lock(&dac->lock); - /* if tail buffer contains zero samples DMA stream was already - * stopped */ - running = dac->desc[dac->tail].cnt; - dac->desc[dac->tail].cnt = 0; - dac->desc[dac->tail].desc.cntinfo = HPCDMA_XIE | HPCDMA_EOX; - /* we just proccessed empty buffer, don't update tail pointer */ - if (running) - hal2_inc_tail(dac); - spin_unlock(&dac->lock); - - wake_up(&dac->dma_wait); -} - -static void hal2_adc_interrupt(struct hal2_codec *adc) -{ - int running; - - spin_lock(&adc->lock); - /* if head buffer contains nonzero samples DMA stream was already - * stopped */ - running = !adc->desc[adc->head].cnt; - adc->desc[adc->head].cnt = H2_BLOCK_SIZE; - adc->desc[adc->head].desc.cntinfo = HPCDMA_XIE | HPCDMA_EOR; - /* we just proccessed empty buffer, don't update head pointer */ - if (running) - hal2_inc_head(adc); - spin_unlock(&adc->lock); - - wake_up(&adc->dma_wait); -} - -static irqreturn_t hal2_interrupt(int irq, void *dev_id) -{ - struct hal2_card *hal2 = dev_id; - irqreturn_t ret = IRQ_NONE; - - /* decide what caused this interrupt */ - if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { - hal2_dac_interrupt(&hal2->dac); - ret = IRQ_HANDLED; - } - if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { - hal2_adc_interrupt(&hal2->adc); - ret = IRQ_HANDLED; - } - return ret; -} - -static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate) -{ - unsigned short mod; - - DEBUG("rate: %d\n", rate); - - if (rate < 4000) rate = 4000; - else if (rate > 48000) rate = 48000; - - if (44100 % rate < 48000 % rate) { - mod = 4 * 44100 / rate; - codec->master = 44100; - } else { - mod = 4 * 48000 / rate; - codec->master = 48000; - } - - codec->inc = 4; - codec->mod = mod; - rate = 4 * codec->master / mod; - - DEBUG("real_rate: %d\n", rate); - - return rate; -} - -static void hal2_set_dac_rate(struct hal2_card *hal2) -{ - unsigned int master = hal2->dac.master; - int inc = hal2->dac.inc; - int mod = hal2->dac.mod; - - DEBUG("master: %d inc: %d mod: %d\n", master, inc, mod); - - hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0); - hal2_i_write32(hal2, H2I_BRES1_C2, ((0xffff & (inc - mod - 1)) << 16) | inc); -} - -static void hal2_set_adc_rate(struct hal2_card *hal2) -{ - unsigned int master = hal2->adc.master; - int inc = hal2->adc.inc; - int mod = hal2->adc.mod; - - DEBUG("master: %d inc: %d mod: %d\n", master, inc, mod); - - hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0); - hal2_i_write32(hal2, H2I_BRES2_C2, ((0xffff & (inc - mod - 1)) << 16) | inc); -} - -static void hal2_setup_dac(struct hal2_card *hal2) -{ - unsigned int fifobeg, fifoend, highwater, sample_size; - struct hal2_pbus *pbus = &hal2->dac.pbus; - - DEBUG("hal2_setup_dac\n"); - - /* Now we set up some PBUS information. The PBUS needs information about - * what portion of the fifo it will use. If it's receiving or - * transmitting, and finally whether the stream is little endian or big - * endian. The information is written later, on the start call. - */ - sample_size = 2 * hal2->dac.voices; - /* Fifo should be set to hold exactly four samples. Highwater mark - * should be set to two samples. */ - highwater = (sample_size * 2) >> 1; /* halfwords */ - fifobeg = 0; /* playback is first */ - fifoend = (sample_size * 4) >> 3; /* doublewords */ - pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD | - (highwater << 8) | (fifobeg << 16) | (fifoend << 24) | - (hal2->dac.format & AFMT_S16_LE ? HPC3_PDMACTRL_SEL : 0); - /* We disable everything before we do anything at all */ - pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; - hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); - /* Setup the HAL2 for playback */ - hal2_set_dac_rate(hal2); - /* Set endianess */ - if (hal2->dac.format & AFMT_S16_LE) - hal2_i_setbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); - else - hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); - /* Set DMA bus */ - hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); - /* We are using 1st Bresenham clock generator for playback */ - hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) - | (1 << H2I_C1_CLKID_SHIFT) - | (hal2->dac.voices << H2I_C1_DATAT_SHIFT)); -} - -static void hal2_setup_adc(struct hal2_card *hal2) -{ - unsigned int fifobeg, fifoend, highwater, sample_size; - struct hal2_pbus *pbus = &hal2->adc.pbus; - - DEBUG("hal2_setup_adc\n"); - - sample_size = 2 * hal2->adc.voices; - highwater = (sample_size * 2) >> 1; /* halfwords */ - fifobeg = (4 * 4) >> 3; /* record is second */ - fifoend = (4 * 4 + sample_size * 4) >> 3; /* doublewords */ - pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD | - (highwater << 8) | (fifobeg << 16) | (fifoend << 24) | - (hal2->adc.format & AFMT_S16_LE ? HPC3_PDMACTRL_SEL : 0); - pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; - hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); - /* Setup the HAL2 for record */ - hal2_set_adc_rate(hal2); - /* Set endianess */ - if (hal2->adc.format & AFMT_S16_LE) - hal2_i_setbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); - else - hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); - /* Set DMA bus */ - hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); - /* We are using 2nd Bresenham clock generator for record */ - hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) - | (2 << H2I_C1_CLKID_SHIFT) - | (hal2->adc.voices << H2I_C1_DATAT_SHIFT)); -} - -static dma_addr_t hal2_desc_addr(struct hal2_codec *codec, int i) -{ - if (--i < 0) - i = codec->desc_count - 1; - return codec->desc[i].desc.pnext; -} - -static void hal2_start_dac(struct hal2_card *hal2) -{ - struct hal2_codec *dac = &hal2->dac; - struct hal2_pbus *pbus = &dac->pbus; - - pbus->pbus->pbdma_dptr = hal2_desc_addr(dac, dac->tail); - pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; - /* enable DAC */ - hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); -} - -static void hal2_start_adc(struct hal2_card *hal2) -{ - struct hal2_codec *adc = &hal2->adc; - struct hal2_pbus *pbus = &adc->pbus; - - pbus->pbus->pbdma_dptr = hal2_desc_addr(adc, adc->head); - pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; - /* enable ADC */ - hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); -} - -static inline void hal2_stop_dac(struct hal2_card *hal2) -{ - hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; - /* The HAL2 itself may remain enabled safely */ -} - -static inline void hal2_stop_adc(struct hal2_card *hal2) -{ - hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; -} - -static int hal2_alloc_dmabuf(struct hal2_codec *codec, int size, - int count, int cntinfo, int dir) -{ - struct hal2_desc *desc, *dma_addr; - int i; - - DEBUG("allocating %dk DMA buffer.\n", size / 1024); - - codec->buffer = (unsigned char *)__get_free_pages(GFP_KERNEL | GFP_DMA, - get_order(size)); - if (!codec->buffer) - return -ENOMEM; - desc = dma_alloc_coherent(NULL, count * sizeof(struct hal2_desc), - (dma_addr_t *)&dma_addr, GFP_KERNEL); - if (!desc) { - free_pages((unsigned long)codec->buffer, get_order(size)); - return -ENOMEM; - } - codec->desc = desc; - for (i = 0; i < count; i++) { - desc->desc.pbuf = dma_map_single(NULL, - (void *)(codec->buffer + i * H2_BLOCK_SIZE), - H2_BLOCK_SIZE, dir); - desc->desc.cntinfo = cntinfo; - desc->desc.pnext = (i == count - 1) ? - (u32)dma_addr : (u32)(dma_addr + i + 1); - desc->cnt = 0; - desc++; - } - codec->desc_count = count; - codec->head = codec->tail = 0; - return 0; -} - -static int hal2_alloc_dac_dmabuf(struct hal2_codec *codec) -{ - return hal2_alloc_dmabuf(codec, H2_DAC_BUFSIZE, - H2_DAC_BUFSIZE / H2_BLOCK_SIZE, - HPCDMA_XIE | HPCDMA_EOX, - DMA_TO_DEVICE); -} - -static int hal2_alloc_adc_dmabuf(struct hal2_codec *codec) -{ - return hal2_alloc_dmabuf(codec, H2_ADC_BUFSIZE, - H2_ADC_BUFSIZE / H2_BLOCK_SIZE, - HPCDMA_XIE | H2_BLOCK_SIZE, - DMA_TO_DEVICE); -} - -static void hal2_free_dmabuf(struct hal2_codec *codec, int size, int dir) -{ - dma_addr_t dma_addr; - int i; - - dma_addr = codec->desc[codec->desc_count - 1].desc.pnext; - for (i = 0; i < codec->desc_count; i++) - dma_unmap_single(NULL, codec->desc[i].desc.pbuf, - H2_BLOCK_SIZE, dir); - dma_free_coherent(NULL, codec->desc_count * sizeof(struct hal2_desc), - (void *)codec->desc, dma_addr); - free_pages((unsigned long)codec->buffer, get_order(size)); -} - -static void hal2_free_dac_dmabuf(struct hal2_codec *codec) -{ - return hal2_free_dmabuf(codec, H2_DAC_BUFSIZE, DMA_TO_DEVICE); -} - -static void hal2_free_adc_dmabuf(struct hal2_codec *codec) -{ - return hal2_free_dmabuf(codec, H2_ADC_BUFSIZE, DMA_FROM_DEVICE); -} - -/* - * Add 'count' bytes to 'buffer' from DMA ring buffers. Return number of - * bytes added or -EFAULT if copy_from_user failed. - */ -static int hal2_get_buffer(struct hal2_card *hal2, char *buffer, int count) -{ - unsigned long flags; - int size, ret = 0; - unsigned char *buf; - struct hal2_desc *tail; - struct hal2_codec *adc = &hal2->adc; - - DEBUG("getting %d bytes ", count); - - spin_lock_irqsave(&adc->lock, flags); - tail = &adc->desc[adc->tail]; - /* enable DMA stream if there are no data */ - if (!tail->cnt && !(adc->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT)) - hal2_start_adc(hal2); - while (tail->cnt > 0 && count > 0) { - size = min((int)tail->cnt, count); - buf = &adc->buffer[(adc->tail + 1) * H2_BLOCK_SIZE - tail->cnt]; - spin_unlock_irqrestore(&adc->lock, flags); - dma_sync_single(NULL, tail->desc.pbuf, size, DMA_FROM_DEVICE); - if (copy_to_user(buffer, buf, size)) { - ret = -EFAULT; - goto out; - } - spin_lock_irqsave(&adc->lock, flags); - tail->cnt -= size; - /* buffer is empty, update tail pointer */ - if (tail->cnt == 0) { - tail->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE; - hal2_inc_tail(adc); - tail = &adc->desc[adc->tail]; - /* enable DMA stream again if needed */ - if (!(adc->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT)) - hal2_start_adc(hal2); - } - buffer += size; - ret += size; - count -= size; - - DEBUG("(%d) ", size); - } - spin_unlock_irqrestore(&adc->lock, flags); -out: - DEBUG("\n"); - - return ret; -} - -/* - * Add 'count' bytes from 'buffer' to DMA ring buffers. Return number of - * bytes added or -EFAULT if copy_from_user failed. - */ -static int hal2_add_buffer(struct hal2_card *hal2, char *buffer, int count) -{ - unsigned long flags; - unsigned char *buf; - int size, ret = 0; - struct hal2_desc *head; - struct hal2_codec *dac = &hal2->dac; - - DEBUG("adding %d bytes ", count); - - spin_lock_irqsave(&dac->lock, flags); - head = &dac->desc[dac->head]; - while (head->cnt == 0 && count > 0) { - size = min((int)H2_BLOCK_SIZE, count); - buf = &dac->buffer[dac->head * H2_BLOCK_SIZE]; - spin_unlock_irqrestore(&dac->lock, flags); - if (copy_from_user(buf, buffer, size)) { - ret = -EFAULT; - goto out; - } - dma_sync_single(NULL, head->desc.pbuf, size, DMA_TO_DEVICE); - spin_lock_irqsave(&dac->lock, flags); - head->desc.cntinfo = size | HPCDMA_XIE; - head->cnt = size; - buffer += size; - ret += size; - count -= size; - hal2_inc_head(dac); - head = &dac->desc[dac->head]; - - DEBUG("(%d) ", size); - } - if (!(dac->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT) && ret > 0) - hal2_start_dac(hal2); - spin_unlock_irqrestore(&dac->lock, flags); -out: - DEBUG("\n"); - - return ret; -} - -#define hal2_reset_dac_pointer(hal2) hal2_reset_pointer(hal2, 1) -#define hal2_reset_adc_pointer(hal2) hal2_reset_pointer(hal2, 0) -static void hal2_reset_pointer(struct hal2_card *hal2, int is_dac) -{ - int i; - struct hal2_codec *codec = (is_dac) ? &hal2->dac : &hal2->adc; - - DEBUG("hal2_reset_pointer\n"); - - for (i = 0; i < codec->desc_count; i++) { - codec->desc[i].cnt = 0; - codec->desc[i].desc.cntinfo = HPCDMA_XIE | (is_dac) ? - HPCDMA_EOX : H2_BLOCK_SIZE; - } - codec->head = codec->tail = 0; -} - -static int hal2_sync_dac(struct hal2_card *hal2) -{ - DECLARE_WAITQUEUE(wait, current); - struct hal2_codec *dac = &hal2->dac; - int ret = 0; - unsigned long flags; - signed long timeout = 1000 * H2_BLOCK_SIZE * 2 * dac->voices * - HZ / dac->sample_rate / 900; - - while (dac->pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_ISACT) { - add_wait_queue(&dac->dma_wait, &wait); - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(timeout); - spin_lock_irqsave(&dac->lock, flags); - if (dac->desc[dac->tail].cnt) - ret = -ETIME; - spin_unlock_irqrestore(&dac->lock, flags); - if (signal_pending(current)) - ret = -ERESTARTSYS; - if (ret) { - hal2_stop_dac(hal2); - hal2_reset_dac_pointer(hal2); - } - remove_wait_queue(&dac->dma_wait, &wait); - } - - return ret; -} - -static int hal2_write_mixer(struct hal2_card *hal2, int index, int vol) -{ - unsigned int l, r, tmp; - - DEBUG_MIX("mixer %d write\n", index); - - if (index >= SOUND_MIXER_NRDEVICES || !mixtable[index].avail) - return -EINVAL; - - r = (vol >> 8) & 0xff; - if (r > 100) - r = 100; - l = vol & 0xff; - if (l > 100) - l = 100; - - hal2->mixer.volume[mixtable[index].idx] = l | (r << 8); - - switch (mixtable[index].idx) { - case H2_MIX_OUTPUT_ATT: - - DEBUG_MIX("output attenuator %d,%d\n", l, r); - - if (r | l) { - tmp = hal2_i_look32(hal2, H2I_DAC_C2); - tmp &= ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); - - /* Attenuator has five bits */ - l = 31 * (100 - l) / 99; - r = 31 * (100 - r) / 99; - - DEBUG_MIX("left: %d, right %d\n", l, r); - - tmp |= (l << H2I_C2_L_ATT_SHIFT) & H2I_C2_L_ATT_M; - tmp |= (r << H2I_C2_R_ATT_SHIFT) & H2I_C2_R_ATT_M; - hal2_i_write32(hal2, H2I_DAC_C2, tmp); - } else - hal2_i_setbit32(hal2, H2I_DAC_C2, H2I_C2_MUTE); - break; - case H2_MIX_INPUT_GAIN: - - DEBUG_MIX("input gain %d,%d\n", l, r); - - tmp = hal2_i_look32(hal2, H2I_ADC_C2); - tmp &= ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M); - - /* Gain control has four bits */ - l = 16 * l / 100; - r = 16 * r / 100; - - DEBUG_MIX("left: %d, right %d\n", l, r); - - tmp |= (l << H2I_C2_L_GAIN_SHIFT) & H2I_C2_L_GAIN_M; - tmp |= (r << H2I_C2_R_GAIN_SHIFT) & H2I_C2_R_GAIN_M; - hal2_i_write32(hal2, H2I_ADC_C2, tmp); - - break; - } - - return 0; -} - -static void hal2_init_mixer(struct hal2_card *hal2) -{ - int i; - - for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) - if (mixtable[i].avail) - hal2->mixer.volume[mixtable[i].idx] = 100 | (100 << 8); - - /* disable attenuator */ - hal2_i_write32(hal2, H2I_DAC_C2, 0); - /* set max input gain */ - hal2_i_write32(hal2, H2I_ADC_C2, H2I_C2_MUTE | - (H2I_C2_L_GAIN_M << H2I_C2_L_GAIN_SHIFT) | - (H2I_C2_R_GAIN_M << H2I_C2_R_GAIN_SHIFT)); - /* set max volume */ - hal2->mixer.master = 0xff; - hal2->vol_regs->left = 0xff; - hal2->vol_regs->right = 0xff; -} - -/* - * XXX: later i'll implement mixer for main volume which will be disabled - * by default. enabling it users will be allowed to have master volume level - * control on panel in their favourite X desktop - */ -static void hal2_volume_control(int direction) -{ - unsigned int master = hal2_card[0]->mixer.master; - struct hal2_vol_regs *vol = hal2_card[0]->vol_regs; - - /* volume up */ - if (direction > 0 && master < 0xff) - master++; - /* volume down */ - else if (direction < 0 && master > 0) - master--; - /* TODO: mute/unmute */ - vol->left = master; - vol->right = master; - hal2_card[0]->mixer.master = master; -} - -static int hal2_mixer_ioctl(struct hal2_card *hal2, unsigned int cmd, - unsigned long arg) -{ - int val; - - if (cmd == SOUND_MIXER_INFO) { - mixer_info info; - - memset(&info, 0, sizeof(info)); - strlcpy(info.id, hal2str, sizeof(info.id)); - strlcpy(info.name, hal2str, sizeof(info.name)); - info.modify_counter = hal2->mixer.modcnt; - if (copy_to_user((void *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - if (cmd == SOUND_OLD_MIXER_INFO) { - _old_mixer_info info; - - memset(&info, 0, sizeof(info)); - strlcpy(info.id, hal2str, sizeof(info.id)); - strlcpy(info.name, hal2str, sizeof(info.name)); - if (copy_to_user((void *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - if (cmd == OSS_GETVERSION) - return put_user(SOUND_VERSION, (int *)arg); - - if (_IOC_TYPE(cmd) != 'M' || _IOC_SIZE(cmd) != sizeof(int)) - return -EINVAL; - - if (_IOC_DIR(cmd) == _IOC_READ) { - switch (_IOC_NR(cmd)) { - /* Give the current record source */ - case SOUND_MIXER_RECSRC: - val = 0; /* FIXME */ - break; - /* Give the supported mixers, all of them support stereo */ - case SOUND_MIXER_DEVMASK: - case SOUND_MIXER_STEREODEVS: { - int i; - - for (val = i = 0; i < SOUND_MIXER_NRDEVICES; i++) - if (mixtable[i].avail) - val |= 1 << i; - break; - } - /* Arg contains a bit for each supported recording source */ - case SOUND_MIXER_RECMASK: - val = 0; - break; - case SOUND_MIXER_CAPS: - val = 0; - break; - /* Read a specific mixer */ - default: { - int i = _IOC_NR(cmd); - - if (i >= SOUND_MIXER_NRDEVICES || !mixtable[i].avail) - return -EINVAL; - val = hal2->mixer.volume[mixtable[i].idx]; - break; - } - } - return put_user(val, (int *)arg); - } - - if (_IOC_DIR(cmd) != (_IOC_WRITE|_IOC_READ)) - return -EINVAL; - - hal2->mixer.modcnt++; - - if (get_user(val, (int *)arg)) - return -EFAULT; - - switch (_IOC_NR(cmd)) { - /* Arg contains a bit for each recording source */ - case SOUND_MIXER_RECSRC: - return 0; /* FIXME */ - default: - return hal2_write_mixer(hal2, _IOC_NR(cmd), val); - } - - return 0; -} - -static int hal2_open_mixdev(struct inode *inode, struct file *file) -{ - struct hal2_card *hal2 = hal2_mixer_find_card(iminor(inode)); - - if (hal2) { - file->private_data = hal2; - return nonseekable_open(inode, file); - } - return -ENODEV; -} - -static int hal2_release_mixdev(struct inode *inode, struct file *file) -{ - return 0; -} - -static int hal2_ioctl_mixdev(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - return hal2_mixer_ioctl((struct hal2_card *)file->private_data, cmd, arg); -} - -static int hal2_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - int val; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *)arg); - - case SNDCTL_DSP_SYNC: - if (file->f_mode & FMODE_WRITE) - return hal2_sync_dac(hal2); - return 0; - - case SNDCTL_DSP_SETDUPLEX: - return 0; - - case SNDCTL_DSP_GETCAPS: - return put_user(DSP_CAP_DUPLEX | DSP_CAP_MULTI, (int *)arg); - - case SNDCTL_DSP_RESET: - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2_reset_adc_pointer(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2_reset_dac_pointer(hal2); - } - return 0; - - case SNDCTL_DSP_SPEED: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - val = hal2_compute_rate(&hal2->adc, val); - hal2->adc.sample_rate = val; - hal2_set_adc_rate(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - val = hal2_compute_rate(&hal2->dac, val); - hal2->dac.sample_rate = val; - hal2_set_dac_rate(hal2); - } - return put_user(val, (int *)arg); - - case SNDCTL_DSP_STEREO: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2->adc.voices = (val) ? 2 : 1; - hal2_setup_adc(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2->dac.voices = (val) ? 2 : 1; - hal2_setup_dac(hal2); - } - return 0; - - case SNDCTL_DSP_CHANNELS: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val != 0) { - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2->adc.voices = (val == 1) ? 1 : 2; - hal2_setup_adc(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2->dac.voices = (val == 1) ? 1 : 2; - hal2_setup_dac(hal2); - } - } - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.voices; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.voices; - return put_user(val, (int *)arg); - - case SNDCTL_DSP_GETFMTS: /* Returns a mask */ - return put_user(H2_SUPPORTED_FORMATS, (int *)arg); - - case SNDCTL_DSP_SETFMT: /* Selects ONE fmt*/ - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val != AFMT_QUERY) { - if (!(val & H2_SUPPORTED_FORMATS)) - return -EINVAL; - if (file->f_mode & FMODE_READ) { - hal2_stop_adc(hal2); - hal2->adc.format = val; - hal2_setup_adc(hal2); - } - if (file->f_mode & FMODE_WRITE) { - hal2_stop_dac(hal2); - hal2->dac.format = val; - hal2_setup_dac(hal2); - } - } else { - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.format; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.format; - } - return put_user(val, (int *)arg); - - case SNDCTL_DSP_POST: - return 0; - - case SNDCTL_DSP_GETOSPACE: { - audio_buf_info info; - int i; - unsigned long flags; - struct hal2_codec *dac = &hal2->dac; - - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - info.fragments = 0; - spin_lock_irqsave(&dac->lock, flags); - for (i = 0; i < dac->desc_count; i++) - if (dac->desc[i].cnt == 0) - info.fragments++; - spin_unlock_irqrestore(&dac->lock, flags); - info.fragstotal = dac->desc_count; - info.fragsize = H2_BLOCK_SIZE; - info.bytes = info.fragsize * info.fragments; - - return copy_to_user((void *)arg, &info, sizeof(info)) ? -EFAULT : 0; - } - - case SNDCTL_DSP_GETISPACE: { - audio_buf_info info; - int i; - unsigned long flags; - struct hal2_codec *adc = &hal2->adc; - - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - info.fragments = 0; - info.bytes = 0; - spin_lock_irqsave(&adc->lock, flags); - for (i = 0; i < adc->desc_count; i++) - if (adc->desc[i].cnt > 0) { - info.fragments++; - info.bytes += adc->desc[i].cnt; - } - spin_unlock_irqrestore(&adc->lock, flags); - info.fragstotal = adc->desc_count; - info.fragsize = H2_BLOCK_SIZE; - - return copy_to_user((void *)arg, &info, sizeof(info)) ? -EFAULT : 0; - } - - case SNDCTL_DSP_NONBLOCK: - file->f_flags |= O_NONBLOCK; - return 0; - - case SNDCTL_DSP_GETBLKSIZE: - return put_user(H2_BLOCK_SIZE, (int *)arg); - - case SNDCTL_DSP_SETFRAGMENT: - return 0; - - case SOUND_PCM_READ_RATE: - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.sample_rate; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.sample_rate; - return put_user(val, (int *)arg); - - case SOUND_PCM_READ_CHANNELS: - val = -EINVAL; - if (file->f_mode & FMODE_READ) - val = hal2->adc.voices; - if (file->f_mode & FMODE_WRITE) - val = hal2->dac.voices; - return put_user(val, (int *)arg); - - case SOUND_PCM_READ_BITS: - return put_user(16, (int *)arg); - } - - return hal2_mixer_ioctl(hal2, cmd, arg); -} - -static ssize_t hal2_read(struct file *file, char *buffer, - size_t count, loff_t *ppos) -{ - ssize_t err; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - struct hal2_codec *adc = &hal2->adc; - - if (!count) - return 0; - if (mutex_lock_interruptible(&adc->sem)) - return -EINTR; - if (file->f_flags & O_NONBLOCK) { - err = hal2_get_buffer(hal2, buffer, count); - err = err == 0 ? -EAGAIN : err; - } else { - do { - /* ~10% longer */ - signed long timeout = 1000 * H2_BLOCK_SIZE * - 2 * adc->voices * HZ / adc->sample_rate / 900; - unsigned long flags; - DECLARE_WAITQUEUE(wait, current); - ssize_t cnt = 0; - - err = hal2_get_buffer(hal2, buffer, count); - if (err > 0) { - count -= err; - cnt += err; - buffer += err; - err = cnt; - } - if (count > 0 && err >= 0) { - add_wait_queue(&adc->dma_wait, &wait); - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(timeout); - spin_lock_irqsave(&adc->lock, flags); - if (!adc->desc[adc->tail].cnt) - err = -EAGAIN; - spin_unlock_irqrestore(&adc->lock, flags); - if (signal_pending(current)) - err = -ERESTARTSYS; - remove_wait_queue(&adc->dma_wait, &wait); - if (err < 0) { - hal2_stop_adc(hal2); - hal2_reset_adc_pointer(hal2); - } - } - } while (count > 0 && err >= 0); - } - mutex_unlock(&adc->sem); - - return err; -} - -static ssize_t hal2_write(struct file *file, const char *buffer, - size_t count, loff_t *ppos) -{ - ssize_t err; - char *buf = (char*) buffer; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - struct hal2_codec *dac = &hal2->dac; - - if (!count) - return 0; - if (mutex_lock_interruptible(&dac->sem)) - return -EINTR; - if (file->f_flags & O_NONBLOCK) { - err = hal2_add_buffer(hal2, buf, count); - err = err == 0 ? -EAGAIN : err; - } else { - do { - /* ~10% longer */ - signed long timeout = 1000 * H2_BLOCK_SIZE * - 2 * dac->voices * HZ / dac->sample_rate / 900; - unsigned long flags; - DECLARE_WAITQUEUE(wait, current); - ssize_t cnt = 0; - - err = hal2_add_buffer(hal2, buf, count); - if (err > 0) { - count -= err; - cnt += err; - buf += err; - err = cnt; - } - if (count > 0 && err >= 0) { - add_wait_queue(&dac->dma_wait, &wait); - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(timeout); - spin_lock_irqsave(&dac->lock, flags); - if (dac->desc[dac->head].cnt) - err = -EAGAIN; - spin_unlock_irqrestore(&dac->lock, flags); - if (signal_pending(current)) - err = -ERESTARTSYS; - remove_wait_queue(&dac->dma_wait, &wait); - if (err < 0) { - hal2_stop_dac(hal2); - hal2_reset_dac_pointer(hal2); - } - } - } while (count > 0 && err >= 0); - } - mutex_unlock(&dac->sem); - - return err; -} - -static unsigned int hal2_poll(struct file *file, struct poll_table_struct *wait) -{ - unsigned long flags; - unsigned int mask = 0; - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - - if (file->f_mode & FMODE_READ) { - struct hal2_codec *adc = &hal2->adc; - - poll_wait(file, &adc->dma_wait, wait); - spin_lock_irqsave(&adc->lock, flags); - if (adc->desc[adc->tail].cnt > 0) - mask |= POLLIN; - spin_unlock_irqrestore(&adc->lock, flags); - } - - if (file->f_mode & FMODE_WRITE) { - struct hal2_codec *dac = &hal2->dac; - - poll_wait(file, &dac->dma_wait, wait); - spin_lock_irqsave(&dac->lock, flags); - if (dac->desc[dac->head].cnt == 0) - mask |= POLLOUT; - spin_unlock_irqrestore(&dac->lock, flags); - } - - return mask; -} - -static int hal2_open(struct inode *inode, struct file *file) -{ - int err; - struct hal2_card *hal2 = hal2_dsp_find_card(iminor(inode)); - - if (!hal2) - return -ENODEV; - file->private_data = hal2; - if (file->f_mode & FMODE_READ) { - struct hal2_codec *adc = &hal2->adc; - - if (adc->usecount) - return -EBUSY; - /* OSS spec wanted us to use 8 bit, 8 kHz mono by default, - * but HAL2 can't do 8bit audio */ - adc->format = AFMT_S16_BE; - adc->voices = 1; - adc->sample_rate = hal2_compute_rate(adc, 8000); - hal2_set_adc_rate(hal2); - err = hal2_alloc_adc_dmabuf(adc); - if (err) - return err; - hal2_setup_adc(hal2); - adc->usecount++; - } - if (file->f_mode & FMODE_WRITE) { - struct hal2_codec *dac = &hal2->dac; - - if (dac->usecount) - return -EBUSY; - dac->format = AFMT_S16_BE; - dac->voices = 1; - dac->sample_rate = hal2_compute_rate(dac, 8000); - hal2_set_dac_rate(hal2); - err = hal2_alloc_dac_dmabuf(dac); - if (err) - return err; - hal2_setup_dac(hal2); - dac->usecount++; - } - - return nonseekable_open(inode, file); -} - -static int hal2_release(struct inode *inode, struct file *file) -{ - struct hal2_card *hal2 = (struct hal2_card *) file->private_data; - - if (file->f_mode & FMODE_READ) { - struct hal2_codec *adc = &hal2->adc; - - mutex_lock(&adc->sem); - hal2_stop_adc(hal2); - hal2_free_adc_dmabuf(adc); - adc->usecount--; - mutex_unlock(&adc->sem); - } - if (file->f_mode & FMODE_WRITE) { - struct hal2_codec *dac = &hal2->dac; - - mutex_lock(&dac->sem); - hal2_sync_dac(hal2); - hal2_free_dac_dmabuf(dac); - dac->usecount--; - mutex_unlock(&dac->sem); - } - - return 0; -} - -static const struct file_operations hal2_audio_fops = { - .owner = THIS_MODULE, - .llseek = no_llseek, - .read = hal2_read, - .write = hal2_write, - .poll = hal2_poll, - .ioctl = hal2_ioctl, - .open = hal2_open, - .release = hal2_release, -}; - -static const struct file_operations hal2_mixer_fops = { - .owner = THIS_MODULE, - .llseek = no_llseek, - .ioctl = hal2_ioctl_mixdev, - .open = hal2_open_mixdev, - .release = hal2_release_mixdev, -}; - -static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3, - int index) -{ - codec->pbus.pbusnr = index; - codec->pbus.pbus = &hpc3->pbdma[index]; - init_waitqueue_head(&codec->dma_wait); - mutex_init(&codec->sem); - spin_lock_init(&codec->lock); -} - -static int hal2_detect(struct hal2_card *hal2) -{ - unsigned short board, major, minor; - unsigned short rev; - - /* reset HAL2 */ - hal2_isr_write(hal2, 0); - /* release reset */ - hal2_isr_write(hal2, H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N); - - hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE); - if ((rev = hal2_rev_look(hal2)) & H2_REV_AUDIO_PRESENT) - return -ENODEV; - - board = (rev & H2_REV_BOARD_M) >> 12; - major = (rev & H2_REV_MAJOR_CHIP_M) >> 4; - minor = (rev & H2_REV_MINOR_CHIP_M); - - printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n", - board, major, minor); - - return 0; -} - -static int hal2_init_card(struct hal2_card **phal2, struct hpc3_regs *hpc3) -{ - int ret = 0; - struct hal2_card *hal2; - - hal2 = kzalloc(sizeof(struct hal2_card), GFP_KERNEL); - if (!hal2) - return -ENOMEM; - - hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0]; - hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1]; - hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2]; - hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3]; - - if (hal2_detect(hal2) < 0) { - ret = -ENODEV; - goto free_card; - } - - hal2_init_codec(&hal2->dac, hpc3, 0); - hal2_init_codec(&hal2->adc, hpc3, 1); - - /* - * All DMA channel interfaces in HAL2 are designed to operate with - * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles - * in D5. HAL2 is a 16-bit device which can accept both big and little - * endian format. It assumes that even address bytes are on high - * portion of PBUS (15:8) and assumes that HPC3 is programmed to - * accept a live (unsynchronized) version of P_DREQ_N from HAL2. - */ -#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \ - (2 << HPC3_DMACFG_D4R_SHIFT) | \ - (2 << HPC3_DMACFG_D5R_SHIFT) | \ - (0 << HPC3_DMACFG_D3W_SHIFT) | \ - (2 << HPC3_DMACFG_D4W_SHIFT) | \ - (2 << HPC3_DMACFG_D5W_SHIFT) | \ - HPC3_DMACFG_DS16 | \ - HPC3_DMACFG_EVENHI | \ - HPC3_DMACFG_RTIME | \ - (8 << HPC3_DMACFG_BURST_SHIFT) | \ - HPC3_DMACFG_DRQLIVE) - /* - * Ignore what's mentioned in the specification and write value which - * works in The Real World (TM) - */ - hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844; - hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844; - - if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED, - hal2str, hal2)) { - printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ); - ret = -EAGAIN; - goto free_card; - } - - hal2->dev_dsp = register_sound_dsp(&hal2_audio_fops, -1); - if (hal2->dev_dsp < 0) { - ret = hal2->dev_dsp; - goto free_irq; - } - - hal2->dev_mixer = register_sound_mixer(&hal2_mixer_fops, -1); - if (hal2->dev_mixer < 0) { - ret = hal2->dev_mixer; - goto unregister_dsp; - } - - hal2_init_mixer(hal2); - - *phal2 = hal2; - return 0; -unregister_dsp: - unregister_sound_dsp(hal2->dev_dsp); -free_irq: - free_irq(SGI_HPCDMA_IRQ, hal2); -free_card: - kfree(hal2); - - return ret; -} - -extern void (*indy_volume_button)(int); - -/* - * Assuming only one HAL2 card. Mail me if you ever meet machine with - * more than one. - */ -static int __init init_hal2(void) -{ - int i, error; - - for (i = 0; i < MAXCARDS; i++) - hal2_card[i] = NULL; - - error = hal2_init_card(&hal2_card[0], hpc3c0); - - /* let Indy's volume buttons work */ - if (!error && !ip22_is_fullhouse()) - indy_volume_button = hal2_volume_control; - - return error; - -} - -static void __exit exit_hal2(void) -{ - int i; - - /* unregister volume butons callback function */ - indy_volume_button = NULL; - - for (i = 0; i < MAXCARDS; i++) - if (hal2_card[i]) { - free_irq(SGI_HPCDMA_IRQ, hal2_card[i]); - unregister_sound_dsp(hal2_card[i]->dev_dsp); - unregister_sound_mixer(hal2_card[i]->dev_mixer); - kfree(hal2_card[i]); - } -} - -module_init(init_hal2); -module_exit(exit_hal2); - -MODULE_DESCRIPTION("OSS compatible driver for SGI HAL2 audio"); -MODULE_AUTHOR("Ladislav Michl"); -MODULE_LICENSE("GPL"); diff --git a/sound/oss/hal2.h b/sound/oss/hal2.h deleted file mode 100644 index 2bd3b52..0000000 --- a/sound/oss/hal2.h +++ /dev/null @@ -1,248 +0,0 @@ -#ifndef __HAL2_H -#define __HAL2_H - -/* - * Driver for HAL2 sound processors - * Copyright (c) 1999 Ulf Carlsson <ulfc(a)bun.falkenberg.se> - * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis(a)linux-mips.org> - * - * 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., 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#include <asm/addrspace.h> -#include <asm/sgi/hpc3.h> -#include <linux/spinlock.h> -#include <linux/types.h> - -/* Indirect status register */ - -#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */ -#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */ -#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */ -#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */ -#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */ - -/* Revision register */ - -#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */ -#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */ -#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */ -#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */ - -/* Indirect address register */ - -/* - * Address of indirect internal register to be accessed. A write to this - * register initiates read or write access to the indirect registers in the - * HAL2. Note that there af four indirect data registers for write access to - * registers larger than 16 byte. - */ - -#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */ - /* block the register resides in */ - /* 1=DMA Port */ - /* 9=Global DMA Control */ - /* 2=Bresenham */ - /* 3=Unix Timer */ -#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */ - /* blockin which the indirect */ - /* register resides */ - /* If IAR_TYPE_M=DMA Port: */ - /* 1=Synth In */ - /* 2=AES In */ - /* 3=AES Out */ - /* 4=DAC Out */ - /* 5=ADC Out */ - /* 6=Synth Control */ - /* If IAR_TYPE_M=Global DMA Control: */ - /* 1=Control */ - /* If IAR_TYPE_M=Bresenham: */ - /* 1=Bresenham Clock Gen 1 */ - /* 2=Bresenham Clock Gen 2 */ - /* 3=Bresenham Clock Gen 3 */ - /* If IAR_TYPE_M=Unix Timer: */ - /* 1=Unix Timer */ -#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */ -#define H2_IAR_PARAM 0x000C /* Parameter Select */ -#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */ - /* 00:word0 */ - /* 01:word1 */ - /* 10:word2 */ - /* 11:word3 */ -/* - * HAL2 internal addressing - * - * The HAL2 has "indirect registers" (idr) which are accessed by writing to the - * Indirect Data registers. Write the address to the Indirect Address register - * to transfer the data. - * - * We define the H2IR_* to the read address and H2IW_* to the write address and - * H2I_* to be fields in whatever register is referred to. - * - * When we write to indirect registers which are larger than one word (16 bit) - * we have to fill more than one indirect register before writing. When we read - * back however we have to read several times, each time with different Read - * Back Indexes (there are defs for doing this easily). - */ - -/* - * Relay Control - */ -#define H2I_RELAY_C 0x9100 -#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */ - -/* DMA port enable */ - -#define H2I_DMA_PORT_EN 0x9104 -#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */ -#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */ -#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */ -#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */ -#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */ - -#define H2I_DMA_END 0x9108 /* global dma endian select */ -#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */ -#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */ -#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */ -#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */ -#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */ - /* 0=b_end 1=l_end */ - -#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */ - -#define H2I_SYNTH_C 0x1104 /* Synth DMA control */ - -#define H2I_AESRX_C 0x1204 /* AES RX dma control */ - -#define H2I_C_TS_EN 0x20 /* Timestamp enable */ -#define H2I_C_TS_FRMT 0x40 /* Timestamp format */ -#define H2I_C_NAUDIO 0x80 /* Sign extend */ - -/* AESRX CTL, 16 bit */ - -#define H2I_AESTX_C 0x1304 /* AES TX DMA control */ -#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ -#define H2I_AESTX_C_CLKID_M 0x18 -#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ -#define H2I_AESTX_C_DATAT_M 0x300 - -/* CODEC registers */ - -#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */ -#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */ -#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */ -#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */ - -/* Bits in CTL1 register */ - -#define H2I_C1_DMA_SHIFT 0 /* DMA channel */ -#define H2I_C1_DMA_M 0x7 -#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ -#define H2I_C1_CLKID_M 0x18 -#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ -#define H2I_C1_DATAT_M 0x300 - -/* Bits in CTL2 register */ - -#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */ -#define H2I_C2_R_GAIN_M 0xf -#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */ -#define H2I_C2_L_GAIN_M 0xf0 -#define H2I_C2_R_SEL 0x100 /* right input select */ -#define H2I_C2_L_SEL 0x200 /* left input select */ -#define H2I_C2_MUTE 0x400 /* mute */ -#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */ -#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */ -#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */ -#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */ -#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */ -#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */ - -#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */ - -/* Clock generator CTL 1, 16 bit */ - -#define H2I_BRES1_C1 0x2104 -#define H2I_BRES2_C1 0x2204 -#define H2I_BRES3_C1 0x2304 - -#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */ -#define H2I_BRES_C1_M 0x03 - -/* Clock generator CTL 2, 32 bit */ - -#define H2I_BRES1_C2 0x2108 -#define H2I_BRES2_C2 0x2208 -#define H2I_BRES3_C2 0x2308 - -#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */ -#define H2I_BRES_C2_INC_M 0xffff -#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */ -#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */ - -/* Unix timer, 64 bit */ - -#define H2I_UTIME 0x3104 -#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */ -#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */ -#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */ -#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */ -#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */ - -struct hal2_ctl_regs { - u32 _unused0[4]; - volatile u32 isr; /* 0x10 Status Register */ - u32 _unused1[3]; - volatile u32 rev; /* 0x20 Revision Register */ - u32 _unused2[3]; - volatile u32 iar; /* 0x30 Indirect Address Register */ - u32 _unused3[3]; - volatile u32 idr0; /* 0x40 Indirect Data Register 0 */ - u32 _unused4[3]; - volatile u32 idr1; /* 0x50 Indirect Data Register 1 */ - u32 _unused5[3]; - volatile u32 idr2; /* 0x60 Indirect Data Register 2 */ - u32 _unused6[3]; - volatile u32 idr3; /* 0x70 Indirect Data Register 3 */ -}; - -struct hal2_aes_regs { - volatile u32 rx_stat[2]; /* Status registers */ - volatile u32 rx_cr[2]; /* Control registers */ - volatile u32 rx_ud[4]; /* User data window */ - volatile u32 rx_st[24]; /* Channel status data */ - - volatile u32 tx_stat[1]; /* Status register */ - volatile u32 tx_cr[3]; /* Control registers */ - volatile u32 tx_ud[4]; /* User data window */ - volatile u32 tx_st[24]; /* Channel status data */ -}; - -struct hal2_vol_regs { - volatile u32 right; /* Right volume */ - volatile u32 left; /* Left volume */ -}; - -struct hal2_syn_regs { - u32 _unused0[2]; - volatile u32 page; /* DOC Page register */ - volatile u32 regsel; /* DOC Register selection */ - volatile u32 dlow; /* DOC Data low */ - volatile u32 dhigh; /* DOC Data high */ - volatile u32 irq; /* IRQ Status */ - volatile u32 dram; /* DRAM Access */ -}; - -#endif /* __HAL2_H */

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Re: [alsa-devel] [PATCH] duplicate SNDRV_PCM_FMTBIT_S{16,24}_BE
by Johannes Berg 19 Aug '08

19 Aug '08

roel kluin wrote: > untested, is it correct? not a clue, do you know how long ago that was? :) does the driver check endianness anywhere? > duplicate SNDRV_PCM_FMTBIT_S{16,24}_BE > > Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> > --- > diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c > b/sound/aoa/codecs/snd-aoa-codec-tas.c > index 7a16a33..c922505 100644 > --- a/sound/aoa/codecs/snd-aoa-codec-tas.c > +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c > @@ -654,15 +654,15 @@ static struct snd_kcontrol_new bass_control = { > static struct transfer_info tas_transfers[] = { > { > /* input */ > - .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | > - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | > + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE, > .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | > SNDRV_PCM_RATE_48000, > .transfer_in = 1, > }, > { > /* output */ > - .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | > - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | > + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE, > .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | > SNDRV_PCM_RATE_48000, > .transfer_in = 0, > }, > >

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[alsa-devel] 1.0.18rc1 release. alsa-lib 1.0.17a
by Jaroslav Kysela 19 Aug '08

19 Aug '08

Hi all, I released alsa-driver, alsa-utils, and alsa-plugins packages version 1.0.18rc1. The alsa-lib tree had a few short bugfix commits - so stable alsa-lib 1.0.17a is available for download now. For changes see: http://www.alsa-project.org/main/index.php/Changes_v1.0.17_v1.0.17a http://www.alsa-project.org/main/index.php/Changes_v1.0.17a_v1.0.18rc1 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] alsa git: which version of ac97_codec.c am i supposed to hxr?
by John L. Utz III 18 Aug '08

18 Aug '08

Hello All; I feel confused by the presence of 2 different copies of ac97_codec.c: jutz-gnto ~ # find . -name ac97_codec.c ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c worse yet,they are not in sync: jutz-gnto ~ # diff -w ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c 1d0 < #include "adriver.h" 38c37 < #include "../../alsa-kernel/pci/ac97/ac97_id.h" --- > #include "ac97_id.h" 40c39 < #include "../../alsa-kernel/pci/ac97/ac97_patch.c" --- > #include "ac97_patch.c" 1917d1915 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 0) 1929d1926 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 14) 1931d1927 < #endif 1953,1956d1948 < #else /* <= 2.6.0 */ < #define snd_ac97_dev_register NULL < #define snd_ac97_dev_disconnect NULL < #endif Which of these am i supposed to use? am i supposed to change both of them for some backward compatibility purpose? tnx! johnu

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[alsa-devel] alsactl init - next round
by Jaroslav Kysela 18 Aug '08

18 Aug '08

Hello, The 'alsactl init' command becomes more powerful now. - support for dB - better support for enumerated controls - better "ready to use" udev support, expected use: RUN="/usr/sbin/alsactl -r /var/run/alsactl.init restore $name" - initialization using a guess method - the init/default configuration mostly copied from the alsaconf script, see http://git.alsa-project.org/?p=alsa-utils.git;a=blob_plain;f=alsactl/init/d… The cardinfo database is slowly filling. We have over 120 files there now. Any volunteer for watching of these files and writting a suitable configuration files for 'alsactl init' here? I'm not sure about the initialization procedure. I selected the method - look to database and if not found - use the default "guess" method (without any warranty). But the opposite way is also possible - do a universal initialization and modify only special parameters for specific hardware. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

On Mon, Aug 18, 2008 at 1:20 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 17 Aug 2008 22:07:19 +0300, > > Note that "Mic Playback Volume" controls only the volume of analog > playback. It doesn't change the recording level at all. thanks for reminding, sure. >> the reasoning for the original patch sounds reasonable. The >> mic volume is tunable, only at the very low range. I recall the laptop >> came with Win98, and there was no problem then with the sound -> not a >> HW problem. I have never got this laptop mic working with linux, so >> this is not anything that broke recently. I see there is >> check_volume_resolution function which tries to guess some of the >> register ranges, but maybe it get's it wrong with this HW. > > Maybe, maybe not. You should check rather ac97 registers in > /proc/asound/card0/codec97#0/ac97#0-0+regs. If build with > CONFIG_SND_DEBUG=y, you can write to the register directly, too. > See Documentaiton[/sound/alsa]/Procfile.txt. > I checked this file. It seems it gets the regs right for the mic (range 0x1f). The capture range is 0xf. I don't have the specs to compare it against, so I assume there is nothing to do more here. It seems mic value decreases while going up to full, where as capture increases. Another difference is maybe stereo / mono related, the capture changes both bytes whereas mic only the other one. Also one thing, should the mic level be around 0x40 - 0x5f, not 0x0 - 0x1f. Here's the diff of the registers between mixing the capture and mic from min to max: $ diff /tmp/ac97_regs_capt_{min,full} 15c15 < 0:1c = 0000 --- > 0:1c = 0f0f $ diff /tmp/ac97_regs_mic_{min,full} 8c8 < 0:0e = 005f --- > 0:0e = 0040 I'll recompile it with sound debug just to test it... Thanks, ikke

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Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45
by S. Aguinaga 18 Aug '08

18 Aug '08

Stan, Thank you for spending time to look at this issue. I owe you a drink, when you are in the chicago area let me know and I'll treat you to one. Ok, so I will check with the developers an see what can be done to correct this. Thank you. //Salvador 1. Re: latest Mac Pro (quad core) audio issues (stan) Message: 1 Date: Fri, 15 Aug 2008 15:14:12 -0700 From: stan <ghjeold_i_mwee(a)cox.net> Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues To: "S. Aguinaga" <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Message-ID: <48A5FFB4.10700(a)cox.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed S. Aguinaga wrote: > > Hello Stan, > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > 2. aplay does not produce sound, period! > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > sudo cat /var/log/messages | grep ALSA > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped > > Let me know if I should send something else. > > // Salvador > // Northwestern University > I think your problem is a misidentification of your sound chip. The model being used for your 889A appears to be incompatible, thus all the errors above. It appears that the macpro ALC885 is being selected before it gets to the choice for the ALC889A. Perhaps a developer can look into this. !!HDA-Intel Codec information !!--------------------------- Codec: Realtek ALC889A !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC885 Analog [ALC885 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 One shows ALC889A and the other shows ALC885. In the driver code there is a comment to the effect that the 889A is best handled as an ALC888, yet the macpro is listed under the ALC885. I am not a developer so can't go beyond this, but it seems that there is a problem. There should be a macpro under the ALC888 as well as the 885. Here are the model options for the two candidates ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 targa Targa T8, MSI-1049 T8 asus-a7j ASUS A7J asus-a7m ASUS A7M macpro MacPro support mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) ALC883/888 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O 3stack-6ch 3-jack 6-channel 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) acer-aspire Acer Aspire 9810 medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targs/MSI with 2-channel laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) _From the code, here is the comment indicating it should be treated as an ALC888: if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { /* Pick up systems that don't supply PCI SSID */ switch (codec->subsystem_id) { case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: /* ALC889A is handled better as ALC888-compatible */ if (codec->revision_id == 0x100103) { alc_free(codec); return patch_alc883(codec); } printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); board_config = ALC882_AUTO; } } ------------------------------ Message: 2 Date: Fri, 15 Aug 2008 16:15:45 -0700 From: "John L. Utz III" <john.utz(a)dmx.com> Subject: [alsa-devel] alsa git: which version of ac97_codec.c am i supposed to hxr? To: "ALSA Developers" <alsa-devel(a)alsa-project.org> Message-ID: <op.ufyawjke647oa9@jutz-gnto64> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes Hello All; I feel confused by the presence of 2 different copies of ac97_codec.c: jutz-gnto ~ # find . -name ac97_codec.c ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c worse yet,they are not in sync: jutz-gnto ~ # diff -w ./alsa-driver/pci/ac97/ac97_codec.c ./alsa-kmirror/pci/ac97/ac97_codec.c 1d0 < #include "adriver.h" 38c37 < #include "../../alsa-kernel/pci/ac97/ac97_id.h" --- > #include "ac97_id.h" 40c39 < #include "../../alsa-kernel/pci/ac97/ac97_patch.c" --- > #include "ac97_patch.c" 1917d1915 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 0) 1929d1926 < #if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 6, 14) 1931d1927 < #endif 1953,1956d1948 < #else /* <= 2.6.0 */ < #define snd_ac97_dev_register NULL < #define snd_ac97_dev_disconnect NULL < #endif Which of these am i supposed to use? am i supposed to change both of them for some backward compatibility purpose? tnx! johnu ------------------------------ Message: 3 Date: Sat, 16 Aug 2008 06:27:02 +0000 (UTC) From: Simon Tremblay <bidou(a)videotron.qc.ca> Subject: [alsa-devel] Configure and use dmix with a new ASoC driver To: alsa-devel(a)alsa-project.org Message-ID: <loom.20080816T061550-625(a)post.gmane.org> Content-Type: text/plain; charset=us-ascii Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done? ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 18, Issue 45 ******************************************

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Re: [alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by krzysztof.h1＠poczta.fm 18 Aug '08

18 Aug '08

> At Mon, 18 Aug 2008 10:50:15 +0200, > Rene Herman wrote: > > > > On 17-08-08 22:38, Krzysztof Helt wrote: > > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > > > Mute sound by setting mute bit without > > > setting volume to 0. It makes both source code > > > and binary shorter. > > > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Grmmbbled-over-by: Rene Herman <rene.herman(a)gmail.com> > > > > This should be safe and the current function indeed generates crap code > > > but in the sense of always expecting the worst broken hardware variants > > > to be out there I must say I'd rather have seen that GCC were beter. But > > > yes, sure. > > Yeah, the slightest concern is the intention of using 0x80 there. > If this patch was already tested on some real hardwares (and I guess > it should work), I'm willing to apply it. Krzystof, was this tested > on your machines? > I have tested it only on cs4236 (inside laptop) and opti931 I have in the old PC at the moment. If you want I can retest on more hardware. The second check I noticed that for some registers the volume is defined as gain (so 0 is the least audible) and for some (CS4231_MONO_CTRL, CS4231_LEFT/RIGHT_OUTPUT) is defined as attentuation (the 0 is the most audible). In two cases, the old code set volume to the loudest and mute it with mute bit, probably the opposite what was intended. It is not a crucial patch. You may drop it. Regards, Krzysztof ---------------------------------------------------------------------- Wygraj laptopa lub iPoda! Wez udzial w igrzyskach! Kliknij >>> http://link.interia.pl/f1eda

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[alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by Krzysztof Helt 18 Aug '08

18 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Mute sound by setting mute bit without setting volume to 0. It makes both source code and binary shorter. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp alsa-ref/sound/isa/wss/wss_lib.c alsa-driver/sound/isa/wss/wss_lib.c --- alsa-ref/sound/isa/wss/wss_lib.c 2008-08-14 00:05:30.000000000 +0200 +++ alsa-driver/sound/isa/wss/wss_lib.c 2008-08-17 22:33:24.620085406 +0200 @@ -574,7 +574,7 @@ static void snd_wss_calibrate_mute(struc { unsigned long flags; - mute = mute ? 1 : 0; + mute = mute ? 0x80 : 0; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->calibrate_mute == mute) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -589,34 +589,34 @@ static void snd_wss_calibrate_mute(struc chip->image[CS4231_LOOPBACK]); } snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); + mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX1_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); + mute | chip->image[CS4231_AUX2_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX2_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); + mute | chip->image[CS4231_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + mute | chip->image[CS4231_RIGHT_OUTPUT]); if (!(chip->hardware & WSS_HW_AD1848_MASK)) { snd_wss_dout(chip, CS4231_LEFT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + mute | chip->image[CS4231_LEFT_LINE_IN]); snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + mute | chip->image[CS4231_RIGHT_LINE_IN]); snd_wss_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); } if (chip->hardware == WSS_HW_INTERWAVE) { snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); + mute | chip->image[CS4231_LEFT_MIC_INPUT]); snd_wss_dout(chip, CS4231_RIGHT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); + mute | chip->image[CS4231_RIGHT_MIC_INPUT]); snd_wss_dout(chip, CS4231_LINE_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); + mute | chip->image[CS4231_LINE_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_LINE_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); + mute | chip->image[CS4231_LINE_RIGHT_OUTPUT]); } chip->calibrate_mute = mute; spin_unlock_irqrestore(&chip->reg_lock, flags); ---------------------------------------------------------------------- Igrzyska na Dzikim Zachodzie! Sprawdz >> http://link.interia.pl/f1edc

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Re: [alsa-devel] developing audio with ALSA
by Henry Ho 18 Aug '08

18 Aug '08

Hi After reading the cross-compilation instruction in INSTALL file, I already successfully configured the alsa-lib-2.0.17. I also successfully made the library. Adding --prefix=xxx during configuration can set the output directory to xxx. I also modify my Makefile for application, including the search path of libasound.so.2.***. Now I can successfully make it. However, when I run my application on the target, I got the following error message: ****************************************************************************************** ./pcm: error while loading shared libraries: libasound.so.2: cannot open shared object file: No such file or directory ****************************************************************************************** I have put libasound.so.2.0.0 in /tmp of the target. I also modified /etc/ld.so.conf, adding /tmp in ld.so.conf file. However, I cannot find ldconfig on the target. Is anyone can tell me how to solve it? Should I rebuild a busybox with this command? Henry ===================================================================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of RichWave is strictly prohibited; and any information in this email irrelevant to the official business of RichWave shall be deemed as neither given nor endorsed by RichWave.

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[alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

Hi, I'm having mic problems with snd_maestro3 and es1988 allegro hw. The laptop is HP OmniBook Xe3. The mic is way too silent (unusable) even with 20db boost option on. I found this seems to have been really common problem, but only one came with a fix [1] to the problem, enabling one more bit to volume control. Could someone please help me apply that fix to current alsa? I tried compiling 2.6.26.2 kernel and made the first change of the patch from mail [1] attachement [2]. No change. The second part of patch I did not find anymore from the code. There are similar parts in ac97_patch.c. Any hint what should be added there? The original patched code is somewhat outdated compared to modern one... But the reasoning for the original patch sounds reasonable. The mic volume is tunable, only at the very low range. I recall the laptop came with Win98, and there was no problem then with the sound -> not a HW problem. I have never got this laptop mic working with linux, so this is not anything that broke recently. I see there is check_volume_resolution function which tries to guess some of the register ranges, but maybe it get's it wrong with this HW. I tried with external headset, the behavior is just the same. The headset is working fine with IBM T41, so not due the mic itself. Any help would be preciated, thanks, ikke [1] http://thread.gmane.org/gmane.linux.alsa.devel/2025 [2] http://cache.gmane.org//gmane/linux/alsa/devel/2025-001.bin Other posts related, with no solutions: http://article.gmane.org/gmane.linux.alsa.user/9802 http://article.gmane.org/gmane.linux.alsa.user/15260 http://article.gmane.org/gmane.linux.alsa.user/9663 http://article.gmane.org/gmane.linux.alsa.devel/27195 HW is HP Omnibook Xe3, 00:08.0 Multimedia audio controller: ESS Technology ES1988 Allegro-1 (rev 12) Ubuntu up to date hardy, with manually compiled linux-2.6.26.2. the patch: ----------------------------------- --- ac97_codec.c Fri Jun 21 16:09:15 2002 +++ /usr/src/linux/sound/pci/ac97/ac97_codec.c Sun Jun 16 21:56:00 2002 @@ -579,7 +579,7 @@ static const snd_kcontrol_new_t snd_ac97_controls_mic[3] = { AC97_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), -AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 15, 1), +AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0) }; @@ -1137,7 +1137,7 @@ for (idx = 0; idx < 3; idx++) if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic[idx], ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_MIC, 0x801f); + snd_ac97_write_cache(ac97, AC97_MIC, 0x805f); /* build Line controls */ for (idx = 0; idx < 2; idx++) ----------------------------------- amixer settings: ----------------------------------- $ amixer Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 21 [68%] [-15.00dB] [on] Front Right: Playback 21 [68%] [-15.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 15 [100%] [0.00dB] [on] Simple mixer control 'Aux',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 15 [100%] [22.50dB] [on] Front Right: Capture 15 [100%] [22.50dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] ----------------------------------- ---> looking at the mic range it seems to be 0-31 already, capture is only 0-15, I wonder if that's right...

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[alsa-devel] Configure and use dmix with a new ASoC driver
by Simon Tremblay 18 Aug '08

18 Aug '08

Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done?

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[alsa-devel] [PATCH] opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
by Krzysztof Helt 17 Aug '08

17 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove card pointer from the snd_opl3sa2 structure and break circular reference snd_card->snd_opl3sa2->snd_card. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A very small opl3sa2 clean up. diff -urp linux-ref/sound/isa/opl3sa2.c linux-2.6/sound/isa/opl3sa2.c --- linux-ref/sound/isa/opl3sa2.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/opl3sa2.c 2008-08-15 12:30:58.000000000 +0200 @@ -124,7 +124,6 @@ static int pnpc_registered; #define OPL3SA2_PM_D3 (OPL3SA2_PM_ADOWN|OPL3SA2_PM_PSV|OPL3SA2_PM_PDN|OPL3SA2_PM_PDX) struct snd_opl3sa2 { - struct snd_card *card; int version; /* 2 or 3 */ unsigned long port; /* control port */ struct resource *res_port; /* control port resource */ @@ -222,14 +221,13 @@ static void snd_opl3sa2_write(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_detect(struct snd_card *card) { - struct snd_card *card; + struct snd_opl3sa2 *chip = card->private_data; unsigned long port; unsigned char tmp, tmp1; char str[2]; - card = chip->card; port = chip->port; if ((chip->res_port = request_region(port, 2, "OPL3-SA control")) == NULL) { snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); @@ -298,12 +296,14 @@ static int __devinit snd_opl3sa2_detect( static irqreturn_t snd_opl3sa2_interrupt(int irq, void *dev_id) { unsigned short status; - struct snd_opl3sa2 *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_opl3sa2 *chip; int handled = 0; - if (chip == NULL || chip->card == NULL) + if (card == NULL || card->private_data == NULL) return IRQ_NONE; + chip = card->private_data; status = snd_opl3sa2_read(chip, OPL3SA2_IRQ_STATUS); if (status & 0x20) { @@ -327,8 +327,10 @@ static irqreturn_t snd_opl3sa2_interrupt snd_opl3sa2_read(chip, OPL3SA2_MASTER_RIGHT); snd_opl3sa2_read(chip, OPL3SA2_MASTER_LEFT); if (chip->master_switch && chip->master_volume) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } } return IRQ_RETVAL(handled); @@ -512,9 +514,9 @@ static void snd_opl3sa2_master_free(stru chip->master_volume = NULL; } -static int __devinit snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; + struct snd_opl3sa2 *chip = card->private_data; struct snd_ctl_elem_id id1, id2; struct snd_kcontrol *kctl; unsigned int idx; @@ -650,7 +652,6 @@ static struct snd_card *snd_opl3sa2_card chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; - chip->card = card; card->private_free = snd_opl3sa2_free; return card; } @@ -672,9 +673,12 @@ static int __devinit snd_opl3sa2_probe(s xdma2 = dma2[dev]; if (xdma2 < 0) chip->single_dma = 1; - if ((err = snd_opl3sa2_detect(chip)) < 0) + err = snd_opl3sa2_detect(card); + if (err < 0) return err; - if (request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, "OPL3-SA2", chip)) { + err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + "OPL3-SA2", card); + if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); return -ENODEV; } @@ -694,7 +698,8 @@ static int __devinit snd_opl3sa2_probe(s err = snd_wss_mixer(wss); if (err < 0) return err; - if ((err = snd_opl3sa2_mixer(chip)) < 0) + err = snd_opl3sa2_mixer(card); + if (err < 0) return err; err = snd_wss_timer(wss, 0, NULL); if (err < 0) ---------------------------------------------------------------------- Prosty kredyt na wszystkie potrzeby. Od 1000 zl do 120 000 zl. Bez poreczycieli i zabezpieczen >>> http://link.interia.pl/f1edf

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[alsa-devel] Problem with latest alsa-info.sh patch
by Travis Place 17 Aug '08

17 Aug '08

The latest patch to allow the uploading to alsa-project.org seems to work fine, for the most part, unless you want to use the --pastebin option for the legacy upload to pastebin.ca . It seems that the uploaded data gets truncated. Not sure if this is the script, or the pastebin having problems. Ill look into it a bit more when i have time (its 5:30am here, and i need sleep). Just thought id point this out. I realise the 'default' behaviour is to use alsa-project.org for uploading, but i feel the other pastebin.ca mode should still function(I personally get people to upload there first, and then 'submit' card details to alsa-project.org/db/ afterwards. Thanks, Travis Place (wishie)

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[alsa-devel] [SOUND/SOC] removed unused #include <version.h>
by Huang Weiyi 17 Aug '08

17 Aug '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/wm8753.c sound/soc/codecs/wm9712.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809..dc7b18f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -34,7 +34,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fb7f9a..2f1c91b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h>

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Re: [alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 16 Aug '08

16 Aug '08

Hello Stan, This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" 2. aplay does not produce sound, period! 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: sudo cat /var/log/messages | grep ALSA Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped Let me know if I should send something else. // Salvador // Northwestern University ----- Original Message ---- From: stan <ghjeold_i_mwee(a)cox.net> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Friday, August 15, 2008 11:54:12 AM Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues S. Aguinaga wrote: > Hello, > > I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. > > // Salvador > Fedora uses pulseaudio by default. How are you determining that there is no output? i.e. what program are you using, and how Try aplay -vv some.wav Try aplay -vv -Dhw:0,0 some.wav Try aplay -vv -Dplughw:0,0 some.wav Are there error messages or does it look like it is playing without sound? Are there messages in /var/log/messages regarding sound at boot or while trying to play. I don't use pulseaudio and don't know it very well, but there is more information here. http://www.pulseaudio.org Could you run a script which provides more information about your sound setup and post the link back here? The script is found at http://www.alsa-project.org/alsa-info.sh You can run the script so the results occur locally, by specifying the ---no-upload option. alsa-info.sh --no-upload You would then paste the output into your email instead of providing a link. This information will enable me to determine if your setup is OK, and look for other problems.

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[alsa-devel] [PATCH 1/2] wm8990: Fix routing of left DAC to speaker mixer
by Mark Brown 15 Aug '08

15 Aug '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) This ought to go into 2.6.27 - it's a fairly straightforward bug fix. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 3ecce51..9505a18 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, - {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"}, /* LONMIX */ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, -- 1.5.6.3

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[alsa-devel] WM8750 utilization
by Harald Radke 15 Aug '08

15 Aug '08

Hi there! Sorry, if this is the wrong list for this, just tell me if so (and maybe point out where to go instead *g*). I am trying to improve usage of the wm8750 codec on my Loox720 PDA. Its a PXA272 device. So far WM8750 driver works quite ok (via I2S), however there are some things that are kinda strange. The Loox720 has a headphone jack, a earpiece and a speaker. When driving LOUT1/ROUT1, sound goes via headphones, LOUT2/ROUT2 drives the earpiece. When setting OUT3 on ROUT1, the speaker plays fine. Strange: - when disabling either LOUT2 or ROUT2 (while the other is enabled), the speaker plays the sound (not sure if the earpiece still is active as speaker output is much louder, but I think so) - when disabling ROUT1/LOUT1 with LOUT2/ROUT2 active, the earpiece plays fine, however there is still some output on the headphone (really really silently) - Trying to utilize headphone switch via the WM8750 registers doesn't really work ): depending on the polarity bit either one of the OUTs is either not muted at all or activated at all. Any ideas or hints? Thanks a lot! Harald Radke

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[alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 15 Aug '08

15 Aug '08

Hello, I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. // Salvador

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[alsa-devel] SPDIF audio / non-audio bit
by Ben Stanley 15 Aug '08

15 Aug '08

Dear alsa-devel, I recently advised this list of some problems I experienced with the SPDIF audio / non-audio bit being set inappropriately while using the ca0106 driver [1]. Seeing as I got no replies, I am trying again. When playing AC3 encoded streams (Dolby Digital, DTS), the SPDIF non-audio bit is set. Otherwise, for PCM streams, the SPDIF is set to audio. Could someone explain to me what part of the software is responsible for correctly setting the SPDIF audio / non-audio bit? Is it * alsa-driver * alsa-lib * application ? It seems to me that currently some applications (MythTV, xine) set it correctly, but most other applications ignore it. So if it is set to non-audio (because for example MythTV crashed before re-setting it to audio), then all applications that are not aware of the setting have their audio broken. Could someone set me straight here so that I can try to produce a permanent fix for this? Thanks, Ben Stanley. [1] http://thread.gmane.org/gmane.linux.alsa.devel/54398/focus=55188

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.27-rc3-74-g027ee62
by Takashi Iwai 15 Aug '08

15 Aug '08

At Fri, 15 Aug 2008 12:39:19 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > @@ -362,10 +363,8 @@ static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream) > } > if (substream->pcm->device & 2) > runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID); > - if ((err = add_playback_constraints(runtime)) < 0) { > - kfree(dpcm); > + if ((err = add_playback_constraints(runtime)) < 0) > return err; > - } Run checkpatch.pl please... Takashi

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[alsa-devel] [Fwd: [Alsa-user] found bug, fixed bug, cannot tell anyone]
by stan 15 Aug '08

15 Aug '08

Forwarded as per request -------- Original Message -------- From: - Thu Aug 14 15:28:40 2008 Message-ID: <001d01c8fe5e$3e4eed40$0402a8c0@sj4> From: Wesley Johnson <johnson2412(a)usgo.net> To: <alsa-user(a)lists.sourceforge.net> Date: Thu, 14 Aug 2008 15:36:55 -0500 (CDT) MIME-Version: 1.0 How do you notify ALSA dev team about a bug that has been in 1.0.16 and 1.0.17. ??? Bugtracker will not let me login, nor email an account confirmation. Getting an ALSA home page account seems to have no purpose at all, it does not let you do anything. BUG: When you try to compile driver for ens1371 by itself the compile will fail. The driver for ens1371 has an include of the file for ens1370, but the makefule does not have this dependency so the ens1370 file is missing. The dev team is probably always compiling for all. Diff file that fixes this problem follows. *** pci/Makefile.orig Mon Jul 14 02:53:57 2008 --- pci/Makefile Thu Jul 17 21:25:42 2008 *************** *** 27,34 **** --- 27,35 ---- atiixp_modem.c: atiixp_modem.patch $(SND_TOPDIR)/alsa-kernel/pci/atiixp_modem.c bt87x.c: bt87x.patch $(SND_TOPDIR)/alsa-kernel/pci/bt87x.c cmipci.c: cmipci.patch $(SND_TOPDIR)/alsa-kernel/pci/cmipci.c ens1370.c: ens1370.patch $(SND_TOPDIR)/alsa-kernel/pci/ens1370.c + ens1371.c: ens1370.c fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c I have been trying for since July 17 to try to send this to ALSA dev team, and have been blocked at every attempt It is now Aug 13. If someone knows how to do this, please post a reply. Wesley D. Johnson --- Get FREE High Speed Internet from USFamily.Net! -- http://www.usfamily.net/mkt-freepromo.html --- ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

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Re: [alsa-devel] [Alsa-user] found bug, fixed bug, cannot tell anyone
by Rene Herman 15 Aug '08

15 Aug '08

On 14-08-08 22:36, Wesley Johnson wrote: > How do you notify ALSA dev team about a bug that has been in 1.0.16 > and 1.0.17. ??? Here will do, although the alsa-devel mailing list would've been better still. Added to CC. > Bugtracker will not let me login, nor email an account confirmation. > Getting an ALSA home page account seems to have no purpose at all, it > does not let you do anything. The homepage is a wiki so it surely does. > BUG: > When you try to compile driver for ens1371 by itself the compile will fail. > The driver for ens1371 has an include of the file for ens1370, but the > makefule does not have this dependency > so the ens1370 file is missing. The dev team is probably always compiling > for all. > > Diff file that fixes this problem follows. > > *** pci/Makefile.orig Mon Jul 14 02:53:57 2008 > --- pci/Makefile Thu Jul 17 21:25:42 2008 > *************** > *** 27,34 **** > --- 27,35 ---- > atiixp_modem.c: atiixp_modem.patch > $(SND_TOPDIR)/alsa-kernel/pci/atiixp_modem.c > bt87x.c: bt87x.patch $(SND_TOPDIR)/alsa-kernel/pci/bt87x.c > cmipci.c: cmipci.patch $(SND_TOPDIR)/alsa-kernel/pci/cmipci.c > ens1370.c: ens1370.patch $(SND_TOPDIR)/alsa-kernel/pci/ens1370.c > + ens1371.c: ens1370.c I expect this should be a straight copy of the ens1370.c line but never use the standalone alsa-driver package. Now that we're on alsa-devel, people who do will reply though. > fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c > intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c > maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c > via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c > > > I have been trying for since July 17 to try to send this to ALSA dev team, > and have been blocked at every attempt > It is now Aug 13. If someone knows how to do this, please post a reply. Well, not to unduly annoy you, and thank you very much for the report and the persistence, but basically all Linux development happens on mailinglists and the mailinglist information is right there on the homepage so, well... The alsa-devel mailinglist is moderated for non-subscribers; if you did email alsa-devel before and just weren't let through the moderation queue then please say so because in that case we definitely have a problem to fix. Rene.

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Re: [alsa-devel] sound-problems with ALC262
by Takashi Iwai 15 Aug '08

15 Aug '08

At Thu, 14 Aug 2008 07:09:41 +0200, Tobias Kaefer wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > > Takashi Iwai schrieb: > | At Wed, 13 Aug 2008 08:07:37 +0200, > | Tobias Kaefer wrote: > |> ... > |> "stan" told me to use this script > |> http://www.alsa-project.org/alsa-info.sh to create more information > |> about my system. Here is the pastebin link of the output of that script: > |> http://pastebin.ca/1169153 > | > | At a quick glance, I see no problem in the configuration. > | > | Does the problem exist even if you play via PCM "hw" device, such as > | > | % aplay -Dhw foo.wav > | > | ?? > | If your WAV file isn't with the supported format/rate, use plughw: > | > | % aplay -Dplughw foo.wav > > Yes. The problem still exists when using the PCM "hw" or "plughw" device. What happens when you load the driver with bdl_pos_adj=0 option? Takashi

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Re: [alsa-devel] [PATCH] hda: dynamically create capture mux controls
by Matthew Ranostay 15 Aug '08

15 Aug '08

Dynamically create capture mux volume controls when a output amp is detected. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..eae6df1 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -794,7 +794,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), { } /* end */ }; @@ -908,12 +907,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - /* analog pc-beep replaced with digital beep support */ /* HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), @@ -931,11 +927,9 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -943,7 +937,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; @@ -953,12 +946,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -967,11 +957,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -982,15 +970,12 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2351,7 +2336,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) +static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, + int idx, const char *name, unsigned long val) { struct snd_kcontrol_new *knew; @@ -2371,6 +2357,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = stac92xx_control_templates[type]; + knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; @@ -2379,6 +2366,14 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char return 0; } + +/* add dynamic controls */ +static int stac92xx_add_control(struct sigmatel_spec *spec, int type, + const char *name, unsigned long val) +{ + return stac92xx_add_control_idx(spec, type, 0, name, val); +} + /* flag inputs as additional dynamic lineouts */ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) { @@ -2780,6 +2775,26 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, return 0; } +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + err = stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3078,6 +3093,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2)

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[alsa-devel] [PATCH] hda: 92HD75xx fixes
by Matthew Ranostay 15 Aug '08

15 Aug '08

Fixed several noise issues with DACs and ADCs on some 92HD75xxx based codecs with certain revision id's. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..9ec0610 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -145,6 +145,9 @@ struct sigmatel_spec { unsigned int gpio_data; unsigned int gpio_mute; + /* stream */ + unsigned int stream_delay; + /* analog loopback */ unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -190,6 +193,7 @@ struct sigmatel_spec { unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; + unsigned int powerdown_adcs; /* i/o switches */ unsigned int io_switch[2]; @@ -2010,6 +2014,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + if (spec->stream_delay) + msleep(spec->stream_delay); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -2073,9 +2079,14 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); + if (spec->powerdown_adcs) { + msleep(40); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } @@ -2084,8 +2095,12 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + snd_hda_codec_cleanup_stream(codec, nid); + if (spec->powerdown_adcs) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); return 0; } @@ -3275,6 +3290,12 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init); + /* power down adcs initially */ + if (spec->powerdown_adcs) + for (i = 0; i < spec->num_adcs; i++) + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ @@ -3909,6 +3930,47 @@ again: return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_POWER_STATE, pwr); + + msleep(1); + for (i = 0; i < spec->num_adcs; i++) { + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, pwr); + } +}; + +static int stac92hd71xx_resume(struct hda_codec *codec) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D0); + return stac92xx_resume(codec); +} + +static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + return 0; +}; + +#endif + +static struct hda_codec_ops stac92hd71bxx_patch_ops = { + .build_controls = stac92xx_build_controls, + .build_pcms = stac92xx_build_pcms, + .init = stac92xx_init, + .free = stac92xx_free, + .unsol_event = stac92xx_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .resume = stac92hd71xx_resume, + .suspend = stac92hd71xx_suspend, +#endif +}; static int patch_stac92hd71bxx(struct hda_codec *codec) { @@ -3920,6 +3982,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = stac92xx_patch_ops; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; @@ -3951,6 +4014,14 @@ again: spec->init = stac92hd71bxx_core_init; break; case 0x111d7608: /* 5 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 0 || + (codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; @@ -3960,6 +4031,13 @@ again: stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; /* fallthru */ @@ -3976,6 +4054,7 @@ again: spec->gpio_dir = 0x01; spec->gpio_data = 0x01; + spec->powerdown_adcs = 1; spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -4008,8 +4087,6 @@ again: return err; } - codec->patch_ops = stac92xx_patch_ops; - return 0; };

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[alsa-devel] is anybody currently building alsa git on gentoo?
by John L. Utz III 15 Aug '08

15 Aug '08

aclocal revision management is not working, the automake-wrapper thingy is installed, but i get warnings about the lack of an aclocal-1.6 and then patching eventually fails. tnx for any help you can provide. johnu

1 1

[alsa-devel] [PATCH] hda: dynamically create capture mux contols
by Matthew Ranostay 14 Aug '08

14 Aug '08

Dynamically create capture mux volume controls when a output amp is detected. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..621cc48 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -794,7 +794,9 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), + */ { } /* end */ }; @@ -908,14 +910,15 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + */ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - /* analog pc-beep replaced with digital beep support */ /* + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ @@ -931,11 +934,15 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + */ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + */ { } /* end */ }; @@ -943,7 +950,9 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), + */ { } /* end */ }; @@ -957,8 +966,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - + */ { } /* end */ }; @@ -971,7 +981,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), + /* HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), + */ { } /* end */ }; @@ -990,7 +1002,9 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), + */ { } /* end */ }; @@ -2780,6 +2794,35 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, return 0; } +/* labels for mux inputs */ +static const char *stac92xx_mux_labels[4] = { + "Capture Mux 1", "Capture Mux 2", "Capture Mux 3", + "Capture Mux 4" +}; + +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + char name[32]; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + sprintf(name, "%s Capture Volume", + stac92xx_mux_labels[i]); + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3078,6 +3121,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2)

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Re: [alsa-devel] Swissonic TS22PCI / Teratec Phase 22 / driver problems with ice1724
by Takashi Iwai 14 Aug '08

14 Aug '08

At Thu, 14 Aug 2008 15:59:06 +0200, Manuel Schneider wrote: > > Takashi Iwai schrieb, Am 14.08.2008 15:47: > > > Pass model=phase22 module option to snd-ice1724 driver. Then it > > handles the card as if it's phase22. > > Thanks for this hin. Unfortunately it changes nothing - still kernel > error message, still no usable sound device. Are you sure? With the proper model name, at least you won't get EEPROM version error any more. > > I also tried the following kernel patch, which gives me an accessible > sound device and usable mixer (and no kernel error), but recording does > not work (just noise). Likely an incompatible hardware design for the recording... Takashi > The patch is quite simple - it just adds the subvendor ID of the > Swissonic card and adds all phase22 functionality to it. > > Manuel > > --- > -:[/usr/src/linux/sound/pci/ice1712]#> diff -uN phase.h.bak phase.h > --- phase.h.bak 2008-08-12 10:13:49.000000000 +0200 > +++ phase.h 2008-08-12 09:49:23.000000000 +0200 > @@ -25,9 +25,11 @@ > */ > > #define PHASE_DEVICE_DESC "{Terratec,Phase 22},"\ > + "{Swissonic,TS22PCI},"\ > "{Terratec,Phase 28}," > > #define VT1724_SUBDEVICE_PHASE22 0x3b155011 > +#define VT1724_SUBDEVICE_TS22PCI 0x3b157b11 > #define VT1724_SUBDEVICE_PHASE28 0x3b154911 > > /* entry point */ > --- > > --- > -:[/usr/src/linux/sound/pci/ice1712]#> diff -uN phase.c.bak phase.c > --- phase.c.bak 2008-08-12 09:51:28.000000000 +0200 > +++ phase.c 2008-08-12 09:51:28.000000000 +0200 > @@ -124,6 +124,11 @@ > ice->num_total_adcs = 2; > ice->vt1720 = 1; // Envy24HT-S have 16 bit wide GPIO > break; > + case VT1724_SUBDEVICE_TS22PCI: > + ice->num_total_dacs = 2; > + ice->num_total_adcs = 2; > + ice->vt1720 = 1; // Envy24HT-S have 16 bit wide GPIO > + break; > default: > snd_BUG(); > return -EINVAL; > @@ -139,6 +144,10 @@ > if ((err = snd_ice1712_akm4xxx_init(ak, &akm_phase22, > &akm_phase22_priv, ice)) < 0) > return err; > break; > + case VT1724_SUBDEVICE_TS22PCI: > + if ((err = snd_ice1712_akm4xxx_init(ak, &akm_phase22, > &akm_phase22_priv, ice)) < 0) > + return err; > + break; > } > > return 0; > @@ -153,6 +162,10 @@ > err = snd_ice1712_akm4xxx_build_controls(ice); > if (err < 0) > return err; > + case VT1724_SUBDEVICE_TS22PCI: > + err = snd_ice1712_akm4xxx_build_controls(ice); > + if (err < 0) > + return err; > } > return 0; > } > @@ -896,6 +909,15 @@ > .eeprom_data = phase22_eeprom, > }, > { > + .subvendor = VT1724_SUBDEVICE_TS22PCI, > + .name = "Swissonic TS22PCI", > + .model = "phase22", > + .chip_init = phase22_init, > + .build_controls = phase22_add_controls, > + .eeprom_size = sizeof(phase22_eeprom), > + .eeprom_data = phase22_eeprom, > + }, > + { > .subvendor = VT1724_SUBDEVICE_PHASE28, > .name = "Terratec PHASE 28", > .model = "phase28", > --- > > > > -- > Kanal Ratte 104,5 MHz > ...das Freie Radio im Wiesental > > Hauptstrasse 82 > 79650 Schopfheim > > Tel: +49 7622 6692-53 > Fax: +49 7622 6692-54 >

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[alsa-devel] Swissonic TS22PCI / Teratec Phase 22 / driver problems with ice1724
by Manuel Schneider 14 Aug '08

14 Aug '08

Hi, we had a Teratec Phase 22 on our transmitter line which works great under Linux, using the ice1724 driver from alsa. As the original sound card broke due to electrical surge on the transmitter line, we had to replace it. The Phase 22 is not available anymore, so we bought a Swissonic TS22PCI which is based on the Phase 22: http://www.thomann.de/gb/swissonic_ts22pci.htm The PCB looks exactly the same as the Phase 22, but the driver does not initialize the card correctly: --- dmesg: ice1724: No matching model found for ID 0x3b157b11 ice1724: Invalid EEPROM version 0 AC'97 0 does not respond - RESET AC'97 0 access is not valid [0xffffffff], removing mixer. ice1712: cannot initialize pro ac97, skipped --- I already updated to Linux 2.6.25 and tried the current alsa-drivers package (1.0.17), but I get always the same errors in the kernel log. I also deleted /etc/asound.* and /var/lib/alsa/asound.state, ran alsa-conf - no luck. Do you have any ideas how I can get this card to work? Thanks a lot for your suggestions, more information see attached. Manuel --- -:[~]#> lspci -vv 00:0a.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: TERRATEC Electronic GmbH Unknown device 117b Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 32 Interrupt: pin A routed to IRQ 18 Region 0: I/O ports at c000 [size=32] Region 1: I/O ports at bc00 [size=128] Capabilities: [80] Power Management version 1 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- --- -:[~]#> cat /etc/modules.d/alsa alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # --- BEGIN: Generated by ALSACONF, do not edit. --- # --- ALSACONF version 1.0.16 --- alias snd-card-0 snd-ice1724 alias sound-slot-0 snd-ice1724 # --- END: Generated by ALSACONF, do not edit. --- --- -:[~]#> lsmod Module Size Used by snd_pcm_oss 24224 0 snd_mixer_oss 14720 1 snd_pcm_oss snd_ice1724 92328 0 snd_ice17xx_ak4xxx 3648 1 snd_ice1724 snd_ac97_codec 105704 1 snd_ice1724 snd_ak4114 8448 1 snd_ice1724 snd_pcm 59660 4 snd_pcm_oss,snd_ice1724,snd_ac97_codec,snd_ak4114 snd_timer 19272 1 snd_pcm snd_page_alloc 8272 1 snd_pcm snd_pt2258 4032 1 snd_ice1724 snd_i2c 5312 2 snd_ice1724,snd_pt2258 snd_ak4xxx_adda 8512 2 snd_ice1724,snd_ice17xx_ak4xxx snd_mpu401_uart 6912 1 snd_ice1724 snd_rawmidi 19808 1 snd_mpu401_uart snd 49288 12 snd_pcm_oss,snd_mixer_oss,snd_ice1724,snd_ac97_codec,snd_ak4114,snd_pcm,snd_timer,snd_pt2258,snd_i2c,snd_ak4xxx_adda,snd_mpu401_uart,snd_rawmidi capi 14272 0 capifs 5072 2 capi kernelcapi 31936 1 capi nfs 118264 3 lockd 62064 2 nfs sunrpc 167432 17 nfs,lockd rtc 11384 0 via_rhine 19976 0 mii 4928 1 via_rhine ac97_bus 1984 1 snd_ac97_codec soundcore 6624 1 snd --- -:[~]#> modinfo snd_ice1724 filename: /lib/modules/2.6.25-gentoo-r7/kernel/sound/pci/ice1712/snd-ice1724.ko license: GPL description: VIA ICEnsemble ICE1724/1720 (Envy24HT/PT) author: Jaroslav Kysela <perex(a)perex.cz> alias: pci:v00001412d00001724sv*sd*bc*sc*i* depends: snd,snd-ac97-codec,snd-pcm,snd-i2c,snd-mpu401-uart,snd-ice17xx-ak4xxx,snd-pt2258,snd-ak4xxx-adda,snd-ak4114 vermagic: 2.6.25-gentoo-r7 mod_unload parm: index:Index value for ICE1724 soundcard. (array of int) parm: id:ID string for ICE1724 soundcard. (array of charp) parm: enable:Enable ICE1724 soundcard. (array of bool) parm: model:Use the given board model. (array of charp) --- -:[~]#> cat /proc/asound/cards 0 [ICE1724 ]: ICE1724 - ICEnsemble ICE1724 ICEnsemble ICE1724 at 0xc000, irq 18 --- -:[~]#> cat /proc/asound/ICE1724/ice1724 ICEnsemble ICE1724 at 0xc000, irq 18 EEPROM: Subvendor : 0x3b157b11 Size : 32 bytes Version : 0 System Config : 0x0 ACLink : 0x0 I2S : 0x0 S/PDIF : 0x0 GPIO direction : 0xffff00 GPIO mask : 0xffffff GPIO state : 0xffffff Extra #18 : 0xff Extra #19 : 0xff Extra #20 : 0xff Extra #21 : 0xff Extra #22 : 0xff Extra #23 : 0xff Extra #24 : 0xff Extra #25 : 0xff Extra #26 : 0x0 Extra #27 : 0x0 Extra #28 : 0x0 Extra #29 : 0x0 Extra #30 : 0x0 Extra #31 : 0x0 ... --- -- Kanal Ratte 104,5 MHz ...das Freie Radio im Wiesental Hauptstrasse 82 79650 Schopfheim Tel: +49 7622 6692-53 Fax: +49 7622 6692-54

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[alsa-devel] Alsa-driver compile fail on Ubuntu powerpc
by nguyen tritai 14 Aug '08

14 Aug '08

Hi all, I'm trying to bring up audio on powerpc AMCC 440EPX board with the soundcard Sound Blaster Audigy CA0106. The board is runing Ubuntu 7.04. I configured: ./configure --with-cards=ca0106 --with-sequencer=yes --with-kernel=/usr/src/linux Then I make and get issue: make[2]: Leaving directory `/root/alsa-driver-1.0.17/usb' make[2]: Entering directory `/root/alsa-driver-1.0.17/misc' make[2]: Leaving directory `/root/alsa-driver-1.0.17/misc' make[1]: Leaving directory `/root/alsa-driver-1.0.17' make -C /usr/src/linux SUBDIRS=/root/alsa-driver-1.0.17 CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux' CC [M] /root/alsa-driver-1.0.17/acore/memory_wrapper.o make[1]: Leaving directory `/usr/src/linux' root@sequoia:~/alsa-driver-1.0.17# vim log In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h:278:1: warning: "GFP_DMA32" redefined In file included from include/linux/slab.h:14, from include/linux/percpu.h:5, from include/linux/rcupdate.h:41, from include/linux/pid.h:4, from include/linux/sched.h:76, from /root/alsa-driver-1.0.17/include/asm/elf.h:5, from include/linux/elf.h:7, from include/linux/module.h:14, from /root/alsa-driver-1.0.17/include/adriver.h:26, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/linux/gfp.h:105:1: warning: this is the location of the previous definition In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h:828: error: static declaration of âjiffies_to_msecsâ follows non-static declaration include/linux/jiffies.h:270: error: previous declaration of âjiffies_to_msecsâ was here /root/alsa-driver-1.0.17/include/adriver.h:847: error: static declaration of âmsecs_to_jiffiesâ follows non-static declaration include/linux/jiffies.h:272: error: previous declaration of âmsecs_to_jiffiesâ was here In file included from /root/alsa-driver-1.0.17/include/adriver.h:944, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/linux/pci.h:594: error: expected identifier or â(â before numeric constant In file included from /root/alsa-driver-1.0.17/include/asm/pci.h:8, from include/linux/pci.h:945, from /root/alsa-driver-1.0.17/include/adriver.h:944, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/linux/mm.h:235: error: conflicting types for âsnd_compat_vmalloc_to_pageâ /root/alsa-driver-1.0.17/include/adriver.h:748: error: previous declaration of âsnd_compat_vmalloc_to_pageâ was here In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h: In function âsnd_pci_orig_save_stateâ: /root/alsa-driver-1.0.17/include/adriver.h:1186: error: too many arguments to function âpci_save_stateâ /root/alsa-driver-1.0.17/include/adriver.h: In function âsnd_pci_orig_restore_stateâ: /root/alsa-driver-1.0.17/include/adriver.h:1190: error: too many arguments to function âpci_restore_stateâ In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h:1552:1: warning: "page_to_pfn" redefined In file included from /root/alsa-driver-1.0.17/include/asm/page.h:150, from /root/alsa-driver-1.0.17/include/asm/thread_info.h:30, from include/linux/thread_info.h:35, from include/linux/preempt.h:9, from include/linux/spinlock.h:49, from include/linux/seqlock.h:29, from include/linux/time.h:8, from include/linux/stat.h:60, from include/linux/module.h:10, from /root/alsa-driver-1.0.17/include/adriver.h:26, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/asm-generic/memory_model.h:79:1: warning: this is the location of the previous definition /root/alsa-driver-1.0.17/acore/memory_wrapper.c: In function âsnd_compat_vmalloc_to_pageâ: /root/alsa-driver-1.0.17/acore/memory_wrapper.c:44: error: implicit declaration of function âVMALLOC_VMADDRâ /root/alsa-driver-1.0.17/acore/memory_wrapper.c:50: error: implicit declaration of function âpte_offsetâ /root/alsa-driver-1.0.17/acore/memory_wrapper.c:50: warning: assignment makes pointer from integer without a cast make[3]: *** [/root/alsa-driver-1.0.17/acore/memory_wrapper.o] Error 1 make[2]: *** [/root/alsa-driver-1.0.17/acore] Error 2 make[1]: *** [_module_/root/alsa-driver-1.0.17] Error 2 The issue looks like an error in alsa-driver source code. I searched the error information and It seems there're lots of people having the same issue but no helpful answers. Do anyone knows how to pass the issue? Thanks, TaiNguyen

2 1

[alsa-devel] ASoC machine driver: use more than one at91 ssc
by Simon Tremblay 14 Aug '08

14 Aug '08

Hi, I want to use more than one SSC (2 to be more accurate) to be able to do what is indicated at Figure 4 of page 10/13 of this document http://www.atmel.com/dyn/resources/prod_documents/doc6020.pdf (how to setup SSC with a I2S codec). I saw in sound/soc/at91/eti_b1_wm8731.c use of 1 SSC. Does I need more than one snd_soc_dai_link in my snd_soc_machine? But for what I understand for now, I need one SSC for sound output and 2 SSC for sound input. I don't know if there is some example about how to do that. Simon

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[alsa-devel] BUG_ON warning
by Matthew Ranostay 14 Aug '08

14 Aug '08

Hello Takashi, I'm getting this warning with the latest snapshot from the git repo. Any ideas on what is causing this? -Matt Ranostay WARNING: at /home/mranostay/git/alsa-driver/acore/timer.c:311 snd_timer_close() BUG? (!timer) Pid: 2461, comm: aplay Not tainted 2.6.25.9 #0 [<e08f5ddf>] snd_timer_close+0x30/0x37 [snd_timer] [<e08f5e12>] snd_timer_user_release+0x2c/0x4a [snd_timer] [<c047e0d2>] __fput+0xad/0x15b [<c047e359>] fput+0x17/0x19 [<c047ba44>] filp_close+0x50/0x5a [<c047cb86>] sys_close+0x69/0x9f [<c0405a76>] syscall_call+0x7/0xb [<c0620000>] ? schedule+0x61b/0x668 =======================

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[alsa-devel] alsactl init - implementation and more..
by Jaroslav Kysela 14 Aug '08

14 Aug '08

Hi, some developers probably already noted, I commited to alsactl utility (alsa-utils package) initial implementation of soundcard initialization stuff aka 'alsactl init'. The basic idea was to "reuse" nice udev configuration style and modify it to ALSA purposes. I hope that I ended up with a syntax which is easy understandable and configuration can be created with minimal effort. Please, look to 'man 7 alsactl_init' for more details and to alsa-utils/alsactl/init directory for example files. Note that the runtime evaluation of commands can be modified using environment variables like udev does. Just try 'alsactl -E CMD=help init' or 'CMD=help alsactl init'. Hardware can be detected using sysfs. For older kernel, I'm plannig to add some compatibility layer to read at least ATTR{subsystem_vendor} and ATTR{subsystem_device} information from procfs (codec files). Also, a way to gather working configurations for specific hardware is required. A nice script alsa-info.sh is already in the alsa-driver package, so I just created a simple persistent "file" dabatase on www.alsa-project.org: http://www.alsa-project.org/cardinfo-db/ and modified alsa-info.sh to use this database for the contents upload by default. Developers can download whole database in a tar package to extract necessary information using command line text tools and use this database also for other purposes (HDA codec graphing and so). More ideas, corrections, opinions are welcome. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] S24_LE Mode for USB Driver
by Charles Eidsness 13 Aug '08

13 Aug '08

Hi All, It looks like the parse_audio_format_i_type function at line 2369 of usbaudio.c (linux v2.6.23) doesn't support the case where bSubframeSize (fmt[5]) = 4 bytes and bBitResolution (fmt[6]) = 24 bits, which should correspond to a S24_LE (or BE) type. I'm developing a new USD DAC which supports this mode (24-bit sample in a 4 byte package). Right now Linux sends me S32_LE formated data, which is wrong. Would it be possible to expand this function to support S24_LE/BE? It would only require adding a few lines of code. Thanks! Charles

3 6

[alsa-devel] snd_assert() removal
by Rene Herman 13 Aug '08

13 Aug '08

Hi Takashi. Just see the snd_assert() removal. Much better as far as I'm concerned but just as a remark -- there's many combined ones of the form: diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 33e9cf1..6a85fdc 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -967,7 +967,8 @@ int snd_cs4236_mixer(struct snd_wss *chip) int err; struct snd_kcontrol_new *kcontrol; - snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); + if (snd_BUG_ON(!chip || !chip->card)) + return -EINVAL; card = chip->card; strcpy(card->mixername, snd_wss_chip_id(chip)); in there. Just thought I'd point that Andrew Morton has been pushing back on multiple conditions per regular BUG_ON()/WARN_ON() lately as they end up giving less useful information then split ones ("so what triggered?"). Just a remark as said but if you agree with him, maybe now is the best time to split them directly as well. Rene.

2 1

[alsa-devel] [PATCH] ALSA-Configuration.txt: supported cards update
by Alexander Beregalov 13 Aug '08

13 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> ALSA-Configuration.txt: supported cards update Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- Documentation/sound/alsa/ALSA-Configuration.txt | 28 ++++++++++++++++------ 1 files changed, 20 insertions(+), 8 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 6f6d117..59069d7 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -746,8 +746,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-hda-intel -------------------- - Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8), - ATI SB450, SB600, RS600, + Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, + PCH, SCH), + ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, + RV630, RV635, RV670, RV770, VIA VT8251/VT8237A, SIS966, ULI M5461 @@ -1168,6 +1170,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Event Electronics, EZ8 * Digigram VX442 * Lionstracs, Mediastaton + * Terrasoniq TS 88 model - Use the given board model, one of the following: delta1010, dio2496, delta66, delta44, audiophile, delta410, @@ -1202,7 +1205,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * TerraTec Phase 22 * TerraTec Phase 28 * AudioTrak Prodigy 7.1 - * AudioTrak Prodigy 7.1LT + * AudioTrak Prodigy 7.1 LT + * AudioTrak Prodigy 7.1 XT + * AudioTrak Prodigy 7.1 HIFI + * AudioTrak Prodigy 7.1 HD2 * AudioTrak Prodigy 192 * Pontis MS300 * Albatron K8X800 Pro II @@ -1213,12 +1219,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Shuttle SN25P * Onkyo SE-90PCI * Onkyo SE-200PCI + * ESI Juli@ + * Hercules Fortissimo IV + * EGO-SYS WaveTerminal 192M model - Use the given board model, one of the following: revo51, revo71, amp2000, prodigy71, prodigy71lt, - prodigy192, aureon51, aureon71, universe, ap192, - k8x800, phase22, phase28, ms300, av710, se200pci, - se90pci + prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, + juli, aureon51, aureon71, universe, ap192, k8x800, + phase22, phase28, ms300, av710, se200pci, se90pci, + fortissimo4, sn25p, wt192m This module supports multiple cards and autoprobe. @@ -1257,7 +1267,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for AC'97 motherboards from Intel and compatibles. * Intel i810/810E, i815, i820, i830, i84x, MX440 - ICH5, ICH6, ICH7, ESB2 + ICH5, ICH6, ICH7, 6300ESB, ESB2 * SiS 7012 (SiS 735) * NVidia NForce, NForce2, NForce3, MCP04, CK804 CK8, CK8S, MCP501 @@ -1297,7 +1307,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ICH5, ICH6, ICH7 * SiS 7013 (SiS 735) * NVidia NForce, NForce2, NForce2s, NForce3 - * AMD AMD8111 + * AMD AMD768, AMD8111 * ALi m5455 ac97_clock - AC'97 codec clock base (0 = auto-detect) @@ -1955,6 +1965,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * CHIC True Sound 4Dwave * Shark Predator4D-PCI * Jaton SonicWave 4D + * SiS SI7018 PCI Audio + * Hoontech SoundTrack Digital 4DWave NX pcm_channels - max channels (voices) reserved for PCM wavetable_size - max wavetable size in kB (4-?kb)

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[alsa-devel] [PATCH] sound/pci: supported cards update
by Alexander Beregalov 13 Aug '08

13 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> sound/pci: supported devices update Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/delta.h | 1 + sound/pci/ice1712/wtm.c | 2 +- sound/pci/intel8x0.c | 6 ++++++ sound/pci/intel8x0m.c | 1 + 4 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index ea7116c..f7f14df 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -31,6 +31,7 @@ "{MidiMan M Audio,Delta DiO 2496},"\ "{MidiMan M Audio,Delta 66},"\ "{MidiMan M Audio,Delta 44},"\ + "{MidiMan M Audio,Delta 410},"\ "{MidiMan M Audio,Audiophile 24/96},"\ "{Digigram,VX442},"\ "{Lionstracs,Mediastation}," diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index a08d17c..784dfd4 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -508,7 +508,7 @@ struct snd_ice1712_card_info snd_vt1724_wtm_cards[] __devinitdata = { { .subvendor = VT1724_SUBDEVICE_WTM, .name = "ESI Waveterminal 192M", - .model = "WT192M", + .model = "wt192m", .chip_init = wtm_init, .build_controls = wtm_add_controls, .eeprom_size = sizeof(wtm_eeprom), diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 048d99e..a0c8492 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -59,6 +59,12 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," "{SiS,SI7012}," "{NVidia,nForce Audio}," "{NVidia,nForce2 Audio}," + "{NVidia,nForce3 Audio}," + "{NVidia,MCP04}," + "{NVidia,MCP501}," + "{NVidia,CK804}," + "{NVidia,CK8}," + "{NVidia,CK8S}," "{AMD,AMD768}," "{AMD,AMD8111}," "{ALI,M5455}}"); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index faf674e..cdb55c6 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -54,6 +54,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," "{NVidia,NForce2 Modem}," "{NVidia,NForce2s Modem}," "{NVidia,NForce3 Modem}," + "{AMD,AMD8111}," "{AMD,AMD768}}"); static int index = -2; /* Exclude the first card */

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[alsa-devel] [PATCH v2] sound/pci: supported cards update
by Alexander Beregalov 13 Aug '08

13 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> sound/pci: supported cards update Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/delta.h | 1 + sound/pci/intel8x0.c | 6 ++++++ sound/pci/intel8x0m.c | 1 + 3 files changed, 8 insertions(+) diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index ea7116c..f7f14df 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -31,6 +31,7 @@ "{MidiMan M Audio,Delta DiO 2496},"\ "{MidiMan M Audio,Delta 66},"\ "{MidiMan M Audio,Delta 44},"\ + "{MidiMan M Audio,Delta 410},"\ "{MidiMan M Audio,Audiophile 24/96},"\ "{Digigram,VX442},"\ "{Lionstracs,Mediastation}," diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 048d99e..a0c8492 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -59,6 +59,12 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," "{SiS,SI7012}," "{NVidia,nForce Audio}," "{NVidia,nForce2 Audio}," + "{NVidia,nForce3 Audio}," + "{NVidia,MCP04}," + "{NVidia,MCP501}," + "{NVidia,CK804}," + "{NVidia,CK8}," + "{NVidia,CK8S}," "{AMD,AMD768}," "{AMD,AMD8111}," "{ALI,M5455}}"); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index faf674e..cdb55c6 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -54,6 +54,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," "{NVidia,NForce2 Modem}," "{NVidia,NForce2s Modem}," "{NVidia,NForce3 Modem}," + "{AMD,AMD8111}," "{AMD,AMD768}}"); static int index = -2; /* Exclude the first card */

1 0

[alsa-devel] developing audio with ALSA
by Henry Ho 13 Aug '08

13 Aug '08

Hi I'm developing audio application using ALSA library. My target platform is ARM platform baseboard. I use GNU toolchain cross compiler as you can see from the following message: arm-none-linux-gnueabi-gcc -lasound -g -O2 -I./alsa-lib-1.0.17rc2/include -o pcm pcm.c /usr/bin/../lib/gcc/arm-none-linux-gnueabi/4.2.0/../../../../arm-none-linux-gnueabi/bin/ld: cannot find -lasound collect2: ld returned 1 exit status make: *** [versatile] Error 1 I know that is because the linker cannot find the asound library. Should I cross compiling ALSA driver/lib/utils first? Henry ===================================================================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of RichWave is strictly prohibited; and any information in this email irrelevant to the official business of RichWave shall be deemed as neither given nor endorsed by RichWave.

2 1

[alsa-devel] Problems with alsa on ARM
by Alan Horstmann 12 Aug '08

12 Aug '08

Hi, I am trying to assist a collegue, we're working on ARM (AT91), and having probs with alsa stuff. Can anyone help interpret this strace from aplay -v -Dhw:0 ourfile.wav? I am comparing it to the same operation on a desktop PC. After all sorts of pre-amble, it gets down to accessing snd devices: open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbecd36ac) = 0 ioctl(3, 0x40045532, 0xbecd36b4) = 0 open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK) = 4 close(3) = 0 ioctl(4, AGPIOC_ACQUIRE or APM_IOC_STANDBY, 0xbecd34f8) = 0 fcntl(4, F_GETFL) = 0x802 (flags O_RDWR|O_NONBLOCK) ioctl(4, AGPIOC_INFO, 0xbecd36a4) = 0 ioctl(4, AGPIOC_RELEASE or APM_IOC_SUSPEND, 0xbecd369c) = 0 mmap2(NULL, 4096, PROT_READ, MAP_SHARED, 4, 0x80000) = -1 ENXIO (No such device or address) ioctl(4, 0xc0844123, 0xbecd3618) = 0 fcntl(4, F_GETFL) = 0x802 (flags O_RDWR|O_NONBLOCK) fcntl(4, F_SETFL, O_RDWR) = 0 ioctl(4, AGPIOC_ACQUIRE or APM_IOC_STANDBY, 0xbecd398c) = 0 The first failure is from mmap2(..). Exactly the same line (and values) is successful on PC, but I don't know what it is doing or why it might fail. There are no other negative returns at all, but after opening the wav file, the playback info is written out, and set_params begins: ioctl(4, 0xc25c4110, 0xbecd3628) = 0 ioctl(4, 0xc25c4110, 0xbecd3628) = 0 ioctl(4, 0xc25c4110, 0xbecd3628) = 0 ioctl(4, 0xc25c4110, 0xbecd3628) = 0 ioctl(4, 0xc25c4111, 0xbecd3628) = 0 ioctl(4, 0xc0844123, 0x1c210) = 0 ioctl(4, 0xc0684113, 0xbecd3528) = 0 ioctl(4, 0x4140, 0) = 0 ioctl(4, 0xc0844123, 0x1c210) = 0 write(2, "aplay", 5aplay) = 5 write(2, ": ", 2: ) = 2 write(2, "set_params", 10set_params) = 10 write(2, ":", 1:) = 1 write(2, "1012", 41012) = 4 write(2, ": ", 2: ) = 2 write(2, "unable to install sw params:", 28unable to install sw params:) = 28 All seems to go OK with hw_params, but sw_params must return error (not seen) as the error check fails. The values printed out show xfer_align = 3201120484 which looks suspicious to me. Any clues on either of these would be greatly appreciated. Alan

3 3

[alsa-devel] Questions about /dev/snd and /proc/asound (new ASoC driver)
by Simon Tremblay 12 Aug '08

12 Aug '08

Hi, I have some issues with /dev/snd. First I should explain what I try to do. I am developing a driver for TLV320AIC23B codec and want to use it on a AT91RM9200-based board. I read documentation in sound/soc and created a codec driver (tlv320aic23b) and a machine driver (for my board). I think it's all I have to do since AT91 is supported and tlv320aic23b works with i2c and I2S. When I load my module I obtain that output: # modprobe snd-soc-armus-tlv320aic23b ASoC version 0.13.1 AIC23B Audio Codec 0.1 <6>asoc: aic23b <-> at91-ssc1 mapping ok <6>armus_tlv320aic23b: MCLK rate 12000000Hz armus_tlv320aic23b: Codec in Master Mode (I activated debug in soc-core and in my machine driver (based mainly on eti_b1_wm8731 machine driver)). I think there is no error here for now... But there is no /dev/snd created. I don't know for now why... Maybe someone has an idea? I think all I have in /proc/asound is correct too: In /proc/asound I have this listing: lrwxrwxrwx 1 root root 5 Jan 1 00:02 ARMUSTLV320AIC2 -> card0 dr-xr-xr-x 4 root root 0 Jan 1 00:02 card0 -r--r--r-- 1 root root 0 Jan 1 00:02 cards -r--r--r-- 1 root root 0 Jan 1 00:02 devices -r--r--r-- 1 root root 0 Jan 1 00:02 modules -r--r--r-- 1 root root 0 Jan 1 00:02 pcm -r--r--r-- 1 root root 0 Jan 1 00:02 timers -r--r--r-- 1 root root 0 Jan 1 00:02 version cat cards: 0 [ARMUSTLV320AIC2]: aic23b - ARMUS_TLV320AIC23B ARMUS_TLV320AIC23B (aic23b) cat devices: 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer cat modules: 0 snd_soc_tlv320aic23b cat pcm: 00-00: TLV320AIC23B PCM aic23b-PCM-0 : : playback 1 : capture 1 cat timers: G0: system timer : 7812.500us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE cat version (of alsa): Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Nov 20 19:16:42 2007 UTC). My Linux Kernel version (uname -a): Linux uclibc 2.6.24 #32 PREEMPT Thu Aug 7 16:20:06 EDT 2008 armv4tl unknown. When I try to use aplay it doesn't work. I have this error ouput from aplay: ALSA lib confmisc.c:769:(parse_card) cannot find card '' ALSA lib conf.c:3510:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:3510:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:3510:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:3982:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:545: audio open error: No such device I think it's because I have nothing in /dev/snd. I'm right? Thank you, Simon Tremblay

3 3

[alsa-devel] Ideal fragment size and number range
by Rahul Iyer 12 Aug '08

12 Aug '08

Hi All For a reasonably responsive application, what would be the ideal fragment size and maximum fragment number range when set using SNDCTL_DSP_SETFRAGMENT (0xMMMMSSSS)? I understand the answer would correspond to the audio buffer size, but I am not certain about the formula involved here. Also for an hda-intel sound card what's the size of ring buffer (for both recording and playback)? Thanks ../rahul

1 0

[alsa-devel] ASoC suspend/resume
by Dmitry Baryshkov 12 Aug '08

12 Aug '08

Hi, I'm trying to clean up spitz ASoC driver: I need to put some GPIO pins to low during suspend and restore them during resume. What is the most logical place to do that? I can't find any suitable suspend hook. -- With best wishes Dmitry

2 1

[alsa-devel] [PATCH] spitz: MONO -> MONO1
by Dmitry Baryshkov 12 Aug '08

12 Aug '08

Correct route name to be MONO1 instead of MONO to follow recent fix in wm8750. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/spitz.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index eefc25b..37cb768 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -297,7 +297,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "LINPUT3"); snd_soc_dapm_disable_pin(codec, "RINPUT3"); snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO"); + snd_soc_dapm_disable_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { -- 1.5.6.3

3 3

[alsa-devel] HDA-Intel - Patch to support new AMD HDMI Audio
by Yang, Libin 12 Aug '08

12 Aug '08

Hi Takashi, This patch is to support new HDMI audio device. Would you please review it? Thanks a lot! --------------------------------------------------------------------- Signed-off-by: Libin Yang <libin.yang(a)amd.com> --- alsa-driver-1.0.17/alsa-kernel/pci/hda/hda_intel.c.orig 2008-08-12 15:33:37.000000000 +0800 +++ alsa-driver-1.0.17/alsa-kernel/pci/hda/hda_intel.c 2008-08-12 15:34:30.000000000 +0800 @@ -2268,6 +2268,7 @@ { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI }, Thanks & Regards, Libin

2 1

[alsa-devel] How to report error in _pointer function?
by Timur Tabi 11 Aug '08

11 Aug '08

In my driver, I can detect in the _pointer callback function whether or not the DMA engine has actually started. Sometimes, when I have a programming error, the DMA will not start, so my _pointer function calculates a crazy value for the current position. The number it returns causes ALSA to go haywire (see my post titled, "underrun!!! (at least 1786051083.613 ms long)". What is the best way to handle this? Is there a way I can tell ALSA, "hey, this is really screwed up, just abort playback and return an error to the app"? -- Timur Tabi Linux kernel developer at Freescale

2 9

[alsa-devel] [PATCH 1/2] spitz: MONO -> MONO1
by Dmitry Baryshkov 11 Aug '08

11 Aug '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/spitz.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 8f5c2ac..dff0f0e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -303,7 +303,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "LINPUT3"); snd_soc_dapm_disable_pin(codec, "RINPUT3"); snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO"); + snd_soc_dapm_disable_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { -- 1.5.6.3

4 13

[alsa-devel] clicking/noise when upsampling with ESI juli@
by Amaury De Ganseman 11 Aug '08

11 Aug '08

Hi, There's a problem, when I start to play a file the sound is bad during 1sec or less (clicking,cracking,...) or when I switch between tracks I hear clicks. Any idea ? Note: It works very well with OSS driver (last mercurial) Thanks for help. Here my asoundrc: defaults.pcm.rate_converter "speexrate_best" pcm.!default { type plug slave.pcm { type dmix ipc_key 1478 slave { pcm "hw:0,1" format S32_LE #period_time 0 period_size 2048 buffer_size 8192 rate 48000 } } #rate_converter "speexrate_best" #rate_converter "samplerate_best" }

1 0

[alsa-devel] Kill snd_assert()
by Takashi Iwai 11 Aug '08

11 Aug '08

Hi, as some people already mentioned, we have a pretty ugly macro, snd_assert(). Actually, this macro is sometimes useful, but the implementation including the code flow in arguments is really bad. I tried to kill this, either the following way: - replace snd_assert(cond, flow...); with if (snd_BUG_ON(!cond)) flow...; - use WARN_ON() if the check is needed essentially regardless of CONFIG_SND_DBEBUG value - remove snd_assert() for obviously unneeded code-paths The new snd_BUG_ON() macro takes the condition, and works like WARN_ON() if CONFIG_SND_DEBUG=y: prints the warning with stack trace, and returns the evaluated value. When CONFIG_SND_DEBUG=n, the macro returns always zero, so that the compiler will optimize out the unreached flows. Since snd_BUG_ON() takes the condition of a bug, it's a negative condition against snd_assert(). It follows BUG_ON() and WARN_ON(), and easier to understand, IMO. The patches are found on topic/remove-assert branch of my git tree: git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git For gitweb, http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog;h=… I haven't merged this topic yet. If you have a big objection or a better implementation idea, please let me know. Takashi

2 3

[alsa-devel] [PATCH] wm8750: it's MONO1, not MONO
by Dmitry Baryshkov 11 Aug '08

11 Aug '08

Since first commit wm8750 contained output named MONO, but all routes mentioned MONO1. Correct MONO to be MONO1. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8750.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e23cb09..2e71394 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -348,7 +348,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), SND_SOC_DAPM_OUTPUT("ROUT2"), - SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), SND_SOC_DAPM_INPUT("LINPUT1"), -- 1.5.6.3

3 3

[alsa-devel] [PATCH] sound/pci/Kconfig: update for ice1712/24
by Alexander Beregalov 11 Aug '08

11 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> sound/pci/Kconfig: update for ice1712/24 Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/Kconfig | 14 +++++++++----- 1 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index f7d95b2..e14dc54 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -649,8 +649,9 @@ config SND_ICE1712 Currently supported hardware is: M-Audio Delta 1010(LT), DiO 2496, 66, 44, 410, Audiophile 24/96; Digigram VX442; - TerraTec EWX 24/96, EWS 88MT, 88D, DMX 6Fire, Phase 88; - Hoontech SoundTrack DSP 24/Value/Media7.1; Event EZ8. + TerraTec EWX 24/96, EWS 88MT/D, DMX 6Fire, Phase 88; + Hoontech SoundTrack DSP 24/Value/Media7.1; Event EZ8; + Lionstracs Mediastation, Terrasoniq TS 88. To compile this driver as a module, choose M here: the module will be called snd-ice1712. @@ -665,9 +666,12 @@ config SND_ICE1724 ICE/VT1724/1720 (Envy24HT/PT) chips. Currently supported hardware is: AMP AUDIO2000; M-Audio - Revolution 7.1; TerraTec Aureon 5.1 Sky, 7.1 Space/Universe; - AudioTrak Prodigy 7.1; Pontis MS300; Albatron K8X800 Pro II; - Chaintech ZNF3-150/250. + Revolution 5.1, 7.1, Audiophile 192; TerraTec Aureon 5.1 Sky, + 7.1 Space/Universe, Phase 22/28; Onkyo SE-90PCI, SE-200PCI; + AudioTrak Prodigy 192, 7.1 (HIFI/LT/XT), HD2; Hercules + Fortissimo IV; ESI Juli@; Pontis MS300; EGO-SYS WaveTerminal + 192M; Albatron K8X800 Pro II; Chaintech ZNF3-150/250, 9CJS, + AV-710; Shuttle SN25P. To compile this driver as a module, choose M here: the module will be called snd-ice1724.

2 1

[alsa-devel] [PATCH] ice1724/revo: simple clean up
by Alexander Beregalov 11 Aug '08

11 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> ice1724/revo: simple clean up Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/revo.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 4d26314..5e1b156 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -524,16 +524,20 @@ static int __devinit revo_init(struct snd_ice1712 *ice) ak = ice->akm = kcalloc(2, sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ak) return -ENOMEM; - ice->akm_codecs = 2; switch (ice->eeprom.subvendor) { case VT1724_SUBDEVICE_REVOLUTION71: ice->akm_codecs = 2; - if ((err = snd_ice1712_akm4xxx_init(ak, &akm_revo_front, &akm_revo_front_priv, ice)) < 0) + err = snd_ice1712_akm4xxx_init(ak, &akm_revo_front, + &akm_revo_front_priv, ice); + if (err < 0) return err; - if ((err = snd_ice1712_akm4xxx_init(ak + 1, &akm_revo_surround, &akm_revo_surround_priv, ice)) < 0) + err = snd_ice1712_akm4xxx_init(ak+1, &akm_revo_surround, + &akm_revo_surround_priv, ice); + if (err < 0) return err; /* unmute all codecs */ - snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE); + snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, + VT1724_REVO_MUTE); break; case VT1724_SUBDEVICE_REVOLUTION51: ice->akm_codecs = 2;

2 1

Re: [alsa-devel] [PATCH 2.6.26.2] hda_intel: ALSA HD Audio patch for Intel Ibex Peak DeviceIDs
by Takashi Iwai 11 Aug '08

11 Aug '08

At Fri, 8 Aug 2008 15:56:39 -0700, Seth Heasley wrote: > > This patch adds the Intel Ibex Peak (PCH) HD Audio Controller DeviceIDs. > > Signed-off by: Seth Heasley <seth.heasley(a)intel.com> Your embedded patch couldn't be applied properly, maybe because of broken spaces. Check your MUA setting. Anyway, I applied it manually. Thanks! Takashi > > --- linux-2.6.26.2/sound/pci/hda/hda_intel.c.orig 2008-08-08 11:25:05.000000000 -0700 > +++ linux-2.6.26.2/sound/pci/hda/hda_intel.c 2008-08-08 11:27:24.000000000 -0700 > @@ -98,6 +98,7 @@ > "{Intel, ICH8}," > "{Intel, ICH9}," > "{Intel, ICH10}," > + "{Intel, PCH}," > "{Intel, SCH}," > "{ATI, SB450}," > "{ATI, SB600}," > @@ -2181,6 +2182,8 @@ > { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, > { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, > { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, > + /* PCH */ > + { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* ATI SB 450/600 */ > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

1 0

[alsa-devel] [PATCH] powerpc/drivers: use linux/of_device.h instead of asm/of_device.h
by Stephen Rothwell 11 Aug '08

11 Aug '08

Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- drivers/hwmon/ams/ams.h | 2 +- sound/aoa/soundbus/soundbus.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) These are the last two users of asm/of_device.h. If everyone is happy, we can put this through the powerpc tree. I have build tested for ppc64_defconfig and ppc6xx_defconfig (which uses the relevant drivers). After this we can remove the include of linux/of_device.h from asm/of_device.h diff --git a/drivers/hwmon/ams/ams.h b/drivers/hwmon/ams/ams.h index a6221e5..221ef69 100644 --- a/drivers/hwmon/ams/ams.h +++ b/drivers/hwmon/ams/ams.h @@ -4,7 +4,7 @@ #include <linux/mutex.h> #include <linux/spinlock.h> #include <linux/types.h> -#include <asm/of_device.h> +#include <linux/of_device.h> enum ams_irq { AMS_IRQ_FREEFALL = 0x01, diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 622cd37..a0f223c 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -8,7 +8,7 @@ #ifndef __SOUNDBUS_H #define __SOUNDBUS_H -#include <asm/of_device.h> +#include <linux/of_device.h> #include <sound/pcm.h> #include <linux/list.h> -- 1.5.6.3 -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au http://www.canb.auug.org.au/~sfr/

3 2

[alsa-devel] clicking/noise when upsampling with ESI juli@
by Amaury De Ganseman 10 Aug '08

10 Aug '08

Hi, There's a problem when I start to play a file the sound is bad during 1sec or less (clicking,cracking,...) or when I switch between tracks I hear clicks. Any idea ? Note: It works very well with OSS driver (last mercurial) Thanks for help. Here my asoundrc: defaults.pcm.rate_converter "speexrate_best" pcm.!default { type plug slave.pcm { type dmix ipc_key 1478 slave { pcm "hw:0,1" format S32_LE #period_time 0 period_size 2048 buffer_size 8192 rate 48000 } } #rate_converter "speexrate_best" #rate_converter "samplerate_best" }

1 0

[alsa-devel] [PATCH] wm8750: it's MONO1, not MONO
by Dmitry Baryshkov 09 Aug '08

09 Aug '08

Since first commit wm8750 contained output named MONO, but all routes mentioned MONO1. Correct MONO to be MONO1. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-ac97-lib.c | 9 ++++----- sound/soc/codecs/wm8750.c | 2 +- 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index be500d3..3802d68 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -8,11 +8,10 @@ #include <sound/pxa2xx-lib.h> #include <asm/irq.h> -#include <asm/hardware.h> - -#include <asm/arch/pxa-regs.h> -#include <asm/arch/pxa2xx-gpio.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/pxa2xx-gpio.h> +#include <mach/audio.h> static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e23cb09..2e71394 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -348,7 +348,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), SND_SOC_DAPM_OUTPUT("ROUT2"), - SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), SND_SOC_DAPM_INPUT("LINPUT1"), -- 1.5.6.3

1 1

[alsa-devel] ALSA OSS Emulation - Read - EAGAIN Errors
by Rahul Iyer 09 Aug '08

09 Aug '08

Hi ALSA Developers I have a full duplex voice application, that records voice and sends (over a network) as well as receives and plays voice. OSS APIs are used to capture and play voice samples. The voice quality is noticeably poor in SUSE 10. After further investigation, I see most of the 'read' system calls fail with EAGAIN (resource unavailable) error. This is not the case in SUSE 9.2. This results in poor voice quality -- broken voice, due to loss of voice samples. Please take a look at the strace stats below for SUSE 9.2 and 10. Check the errors column for SUSE 10. SUSE 9.2 Process 17533 detached % time seconds usecs/call calls errors syscall ------ ----------- ----------- --------- --------- ---------------- 51.54 0.824285 48 17152 gettimeofday 14.36 0.229609 52 4393 693 select 9.85 0.157454 76 2081 692 sigreturn 7.89 0.126199 14 8784 rt_sigprocmask 7.74 0.123765 20 6201 ioctl 4.31 0.068929 33 2067 read 4.31 0.068898 33 2067 write 0.01 0.000083 6 14 alarm 0.00 0.000034 11 3 munmap 0.00 0.000007 4 2 rt_sigaction 0.00 0.000003 3 1 1 open ------ ----------- ----------- --------- --------- ---------------- 100.00 1.599266 42765 1386 total SUSE 10 Process 11342 detached % time seconds usecs/call calls errors syscall ------ ----------- ----------- --------- --------- ---------------- 49.58 1.006722 1 1191090 1189103 read 49.44 1.003873 1 1194607 ioctl 0.95 0.019285 11 1698 write 0.02 0.000453 0 1152 551 select 0.00 0.000038 13 3 munmap 0.00 0.000015 0 9230 gettimeofday 0.00 0.000000 0 1 1 open 0.00 0.000000 0 11 alarm 0.00 0.000000 0 0.00 0.000000 0 793 624 sigreturn 0.00 0.000000 0 6 sched_yield 0.00 0.000000 0 2 rt_sigaction 0.00 0.000000 0 2303 rt_sigprocmask ------ ----------- ----------- --------- --------- ---------------- 100.00 2.030386 2400896 1190279 total BTW, the SUSE 10 machine runs on newer hardware with on board Intel HDA Sound (Realtek ALC262). Any thoughts on what causes so many read failures? Could this be hardware/driver related? There is nothing fancy about this sound card, its a regular on board card. It would be great if someone can confirm their experience with this card. Any suggestions are also most welcome... Thanks for the help.-- Rahul Iyer

3 3

[alsa-devel] Suggestion for http://git.alsa-project.org/
by Timur Tabi 08 Aug '08

08 Aug '08

If you go to http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git, you will see this text: description Sound development tree owner Takashi Iwai last change Fri, 8 Aug 2008 14:50:58 +0000 URL git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git http://www.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git This is very helpful information. It makes git-clone much easier, because I can just copy/psate the URL. Can we have the same thing at http://git.alsa-project.org/? I don't want to have to keep figuring out the http URL for cloning alsa-lib, every time I need to do it. -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] [PATCH] ALSA: wss_lib: missing closing brace in (ifdeffed out) debug function.
by Rene Herman 08 Aug '08

08 Aug '08

2 1

[PATCH] ALSA: wss_lib: missing closing brace in (ifdeffed out) debug function.
by Rene Herman 08 Aug '08

08 Aug '08

Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/wss/wss_lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 011da7a..6b7a0fc 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -282,7 +282,7 @@ static void snd_wss_debug(struct snd_wss *chip) printk(KERN_DEBUG "CS4231 REGS: INDEX = 0x%02x " " STATUS = 0x%02x\n", - wss_inb(chip, CS4231P(REGSEL), + wss_inb(chip, CS4231P(REGSEL)), wss_inb(chip, CS4231P(STATUS))); printk(KERN_DEBUG " 0x00: left input = 0x%02x " -- 1.5.5

1 0

Re: [alsa-devel] [PATCH 03/11] wss_lib: rename cs4321_foo to wss_foo
by Rene Herman 08 Aug '08

08 Aug '08

On 31-07-08 21:03, Krzysztof Helt wrote: Takashi, you already applied these but is it possible to still get my Acked-by (or Reviewed-by, whatever you want) on them? When a problem is bisected to one of these, people tend to put the people in the patch tags in CC and I looked at them and would like to be informed of any trouble. I tested most users by now, and we'll get the azt2320 fixed before this hits the streets but I still like to be in the loop on any traffic concerning them. Acked-by: Rene Herman <rene.herman(a)gmail.com> (or Reviewed-by, Tested-by, ...) Below comments not important: > @@ -679,23 +679,25 @@ static int __devinit snd_opl3sa2_probe(s [ ... ] > - chip->cs4231 = cs4231; > - if ((err = snd_cs4231_pcm(cs4231, 0, NULL)) < 0) > + chip->wss = wss; > + err = snd_wss_pcm(wss, 0, NULL); > + if (err < 0) > return err; > - if ((err = snd_cs4231_mixer(cs4231)) < 0) > + err = snd_wss_mixer(wss); > + if (err < 0) > return err; > if ((err = snd_opl3sa2_mixer(chip)) < 0) > return err; <mumble> > - if ((err = snd_cs4231_timer(cs4231, 0, NULL)) < 0) > + err = snd_wss_timer(wss, 0, NULL); > + if (err < 0) > return err; [ ... ] > +static void snd_wss_debug(struct snd_wss *chip) > + printk(KERN_DEBUG > + " 0x00: left input = 0x%02x " > + " 0x10: alt 1 (CFIG 2) = 0x%02x\n", > + snd_wss_in(chip, 0x00), > + snd_wss_in(chip, 0x10)); > + printk(KERN_DEBUG > + " 0x01: right input = 0x%02x " > + " 0x11: alt 2 (CFIG 3) = 0x%02x\n", > + snd_wss_in(chip, 0x01), > + snd_wss_in(chip, 0x11)); I'll admit that's actually half-way passable... > @@ -553,79 +628,88 @@ static void snd_wss_playback_format(s [ ... ] > if (full_calib) { > - snd_cs4231_mce_up(chip); > + snd_wss_mce_up(chip); > spin_lock_irqsave(&chip->reg_lock, flags); > - if (chip->hardware != CS4231_HW_INTERWAVE && !chip->single_dma) { > - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, > - (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? > - (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : > - pdfr); > + if (chip->hardware != WSS_HW_INTERWAVE && !chip->single_dma) { > + if (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) > + pdfr = (pdfr & 0xf0) | > + (chip->image[CS4231_REC_FORMAT] & 0x0f); > } else { > - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); > + chip->image[CS4231_PLAYBK_FORMAT] = pdfr; > } > + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, pdfr); > spin_unlock_irqrestore(&chip->reg_lock, flags); Much beter. > +static void snd_wss_capture_format(struct snd_wss *chip, [ ... ] > - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); > - snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT] = cdfr); > - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); > + snd_wss_out(chip, CS4231_ALT_FEATURE_1, > + chip->image[CS4231_ALT_FEATURE_1] | 0x20); > + snd_wss_out(chip, CS4231_REC_FORMAT, > + chip->image[CS4231_REC_FORMAT] = cdfr); Not much better... > + snd_wss_out(chip, CS4231_ALT_FEATURE_1, > + chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); Same level of betterness. Rene.

2 4

[alsa-devel] Application hangs with different period sizes
by anuj aggarwal 08 Aug '08

08 Aug '08

My audio driver supports buffer sizes from 128 to 32768 and period sizes from 8 to 2048. I am trying to set the period size as 1030 (just a random number between 8 & 2048, but not multiple of 2^n) and buffer size as 16*buffer_size (i.e. 16480). The output is as follows: Buffer size range from 128 to 32768 Period size range from 8 to 2048 Request period size 1030 and got 1030 Plug PCM: Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16480 period_size : 1030 period_time : 23356 tstamp_mode : NONE period_step : 1 avail_min : 1030 start_threshold : 16480 stop_threshold : 16480 silence_threshold: 0 silence_size : 0 boundary : 1080033280 The problem with this setup is the application just hangs without playing anything. If I use period size as 2048, the app plays the song but clips the last part of it. If I use 1024/512, it works fine. I have tried alsa lib version 1.0.15 & 1.0.16 but the problem persists. Is there anything wrong with my app or audio driver? Please help. Thanks in advance, Anuj Aggarwal

3 7

[alsa-devel] Can't build alsa-driver git, missing isa/wss directory.
by Eliot Blennerhassett 08 Aug '08

08 Aug '08

after recent git-pull of alsa-kmirror, I can no longer build alsa-driver Seems the isa/wss (present in alsa-kmirror) directory is missing from alsa-driver? regards -- Eliot Blennerhassett www.audioscience.com

2 1

[alsa-devel] Esi Maya44 pci:include the existing patch in the next alsa release
by Giuseppe Di Lernia 08 Aug '08

08 Aug '08

Hi, I found this useful howto http://ubuntuforums.org/showthread.php?p=5542917#post5542917 , to make it work the Egosys Maya44 pci under linux. Unfortunately as the poster says, the patch for this card was not included in the last Alsa driver release.So as the patch is working nicely I would like request if the patch can be included in the next alsa driver release. Regards Giuseppe Di Lernia

2 1

[alsa-devel] Why is substream->pcm->dev == NULL?
by Timur Tabi 08 Aug '08

08 Aug '08

In my _pointer callback function, I have this: printk(KERN_INFO "%s:%u substream->pcm->dev=%p, substream->pcm->card->dev=%p\n", __func__, __LINE__, substream->pcm->dev, substream->pcm->card->dev); During playback, it displays this: fsl_dma_pointer:698 substream->pcm->dev=00000000, substream->pcm->card->dev=dfa50a08 Why is substream->pcm->dev == NULL? What's the point of having this field if you're not going to use it? -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] [PATCH] ALSA: fix DMA channel selection in Freescale MPC8610 sound drivers
by Timur Tabi 07 Aug '08

07 Aug '08

On the Freescale MPC8610, SSI1 is hard-coded to use DMA channels 0 and 1 for playback and capture, and SSI2 is hard-coded to use DMA channels 2 and 3. This patch fixes the fabric driver so that it uses the right channels. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 26 +++++++++++++++----------- 1 files changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 4bdc9d8..94f89de 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -68,10 +68,6 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], 0); - guts_set_pmuxcr_dma(machine_data->guts, 1, 0, 0); - guts_set_pmuxcr_dma(machine_data->guts, 1, 3, 0); - guts_set_pmuxcr_dma(machine_data->guts, 0, 3, 0); - switch (machine_data->ssi_id) { case 0: clrsetbits_be32(&machine_data->guts->pmuxcr, @@ -230,6 +226,8 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, struct fsl_ssi_info ssi_info; struct fsl_dma_info dma_info; int ret = -ENODEV; + unsigned int playback_dma_channel; + unsigned int capture_dma_channel; machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); if (!machine_data) @@ -381,8 +379,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, goto error; } - /* Find the DMA channels to use. For now, we always use the first DMA - controller. */ + /* Find the DMA channels to use. Both SSIs need to use the same DMA + * controller, so let's use DMA#1. + */ for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { iprop = of_get_property(dma_np, "cell-index", NULL); if (iprop && (*iprop == 0)) { @@ -397,14 +396,19 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } machine_data->dma_id = *iprop; + /* SSI1 needs to use DMA Channels 0 and 1, and SSI2 needs to use DMA + * channels 2 and 3. This is just how the MPC8610 is wired + * internally. + */ + playback_dma_channel = (machine_data->ssi_id == 0) ? 0 : 2; + capture_dma_channel = (machine_data->ssi_id == 0) ? 1 : 3; + /* - * Find the DMA channels to use. For now, we always use DMA channel 0 - * for playback, and DMA channel 1 for capture. + * Find the DMA channels to use. */ while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { iprop = of_get_property(dma_channel_np, "cell-index", NULL); - /* Is it DMA channel 0? */ - if (iprop && (*iprop == 0)) { + if (iprop && (*iprop == playback_dma_channel)) { /* dma_channel[0] and dma_irq[0] are for playback */ dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); dma_info.dma_irq[0] = @@ -412,7 +416,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, machine_data->dma_channel_id[0] = *iprop; continue; } - if (iprop && (*iprop == 1)) { + if (iprop && (*iprop == capture_dma_channel)) { /* dma_channel[1] and dma_irq[1] are for capture */ dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); dma_info.dma_irq[1] = -- 1.5.5

2 1

[alsa-devel] Dell Studio 15 - aka Inspiron 1535 sound problems: HDA intel and codec IDT (STAC) 92HD73C1X5
by Andrea Venturi 07 Aug '08

07 Aug '08

hi, i own a brand new Dell laptop Studio 15 the audio speakers stay mute if i play an audio file. :-| i'm using debian lenny kernel. it seems to me a wrong port mapping. i put in place a newer Alsa 1.0.17 kernel modules with debug enabled, but still no sound. i append some relevant info about HW, alsa dmesg and codec description you can find more info about this laptop and the audio not workig here (look for audio support): http://www.avalpa.com/assets/andrea/studio15/debian_on_dell_studio15.html this is the dmesg detection WHEN i tried to remove the Subsystem Id: 0x10280254 autodetection (because it catches my subsystem albeit it says "unknown Dell"!) i'd really like to help more and try to solve myself the quirks of this new laptop but this NID stuff with the port/mixer configuration is really beyond a casual user workload. let me know if you need more info. bye andrea Venturi [36685.486337] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:2069: chipset global capabilities = 0x4401 [36685.514673] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:754: codec_mask = 0x1 [36685.516706] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:3501: hda_codec: Unknown model for STAC92HD73XX, using BIOS defaults [36685.516843] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 0a bios pin config 0321101f [36685.516921] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 0b bios pin config 4f00000f [36685.516992] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 0c bios pin config 4f0000f0 [36685.517068] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 0d bios pin config 90170110 [36685.517138] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 0e bios pin config 03a11020 [36685.517214] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 0f bios pin config 0321101f [36685.517315] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 10 bios pin config 4f0000f0 [36685.517392] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 11 bios pin config 4f0000f0 [36685.517461] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 12 bios pin config 4f0000f0 [36685.517537] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 13 bios pin config 90a60160 [36685.517604] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 14 bios pin config 4f0000f0 [36685.517683] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 1e bios pin config 00000000 [36685.517752] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1813: hda_codec: pin nid 22 bios pin config 4f0000f0 [36685.518466] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3030: autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) [36685.518471] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3034: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [36685.518474] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3038: hp_outs=2 (0xa/0xf/0x0/0x0/0x0) [36685.518477] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3039: mono: mono_out=0x0 [36685.518480] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3047: inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 [36685.518484] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2249: stac92xx_add_dyn_out_pins: total dac count=3 [36685.582821] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Center Playback Volume, skipped [36685.582828] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave LFE Playback Volume, skipped [36685.582832] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Side Playback Volume, skipped [36685.582836] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker Playback Volume, skipped [36685.582840] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave External Speaker Playback Volume, skipped [36685.582843] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker2 Playback Volume, skipped [36685.582850] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Center Playback Switch, skipped [36685.582854] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave LFE Playback Switch, skipped [36685.582857] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Side Playback Switch, skipped [36685.582862] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker Playback Switch, skipped [36685.582865] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave External Speaker Playback Switch, skipped [36685.582869] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker2 Playback Switch, skipped [36685.582872] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave IEC958 Playback Switch, skipped [36685.585858] ACPI: PCI Interrupt 0000:01:00.1[B] -> GSI 17 (level, low) -> IRQ 17 [36685.585886] PCI: Setting latency timer of device 0000:01:00.1 to 64 [36685.585891] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:2069: chipset global capabilities = 0x1001 [36685.600103] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:754: codec_mask = 0x1 [35851.601774] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1390: azx_pcm_prepare: bufsize=0x10000, format=0x31 [35851.601807] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x15, stream=0x5, channel=0, format=0x31 [35851.609097] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x17, stream=0x5, channel=0, format=0x31 [35851.617097] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x16, stream=0x5, channel=0, format=0x31 [35851.628764] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x17, stream=0x5, channel=0, format=0x31 [35855.924252] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x15 [35855.924252] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x16 [35855.924252] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 [35855.924252] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 andrea@nb-andrea:~/mp3$ Codec: IDT 92HD73C1X5 Address: 0 Vendor Id: 0x111d7675 Subsystem Id: 0x10280254 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x0a [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=30, enabled=1 Connection: 4 0x15 0x16* 0x17 0x1e Node 0x0b [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x4f00000f: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0xf Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 4 0x15* 0x16 0x17 0x1e Node 0x0c [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 4 0x15* 0x16 0x17 0x1e Node 0x0d [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x083f: IN OUT HP Detect Trigger ImpSense Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 4 0x15* 0x16 0x17 0x1e Node 0x0e [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x03a11020: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 4 0x15* 0x16 0x17 0x1e Node 0x0f [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x40: OUT Unsolicited: tag=35, enabled=1 Connection: 4 0x15 0x16 0x17* 0x1e Node 0x10 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 4 0x15* 0x16 0x17 0x1e Node 0x11 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 4 0x15* 0x16 0x17 0x1e Node 0x12 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x13 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0820: IN Pin Default 0x90a60160: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0820: IN Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Node 0x15 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x41 0x41] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x16 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x5a 0x5a] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x17 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x0f 0x0f] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x18 [Vendor Defined Widget] wcaps 0xfd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: setting=D0, actual=D3 Delay: 13 samples Node 0x19 [Vendor Defined Widget] wcaps 0xfd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: setting=D0, actual=D3 Delay: 13 samples Node 0x1a [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x20 Processing caps: benign=0, ncoeff=0 Node 0x1b [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x21 Processing caps: benign=0, ncoeff=0 Node 0x1c [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x1d [Audio Mixer] wcaps 0x20010b: 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] Connection: 5 0x28 0x29 0x2a 0x2b 0x12 Node 0x1e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x1d Node 0x1f [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=108 Connection: 3 0x15* 0x16 0x17 Node 0x20 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d* Node 0x21 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d* Node 0x22 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Connection: 3 0x25* 0x20 0x21 Node 0x23 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x4f00000f: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0xf Pin-ctls: 0x00: Connection: 3 0x26* 0x20 0x21 Node 0x24 [Pin Complex] wcaps 0x400681: Stereo Digital Pincap 0x0810024: IN EAPD Detect EAPD 0x0: Pin Default 0x4f00000f: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0xf Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x26 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x27 [Audio Input] wcaps 0x14031b: Stereo Digital Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x24 Node 0x28 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a 0x0b* 0x0d 0x0f Node 0x29 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a 0x0e* 0x10 0x11 Node 0x2a [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0b 0x0c* 0x10 0x11 Node 0x2b [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x15* 0x16 0x17 [28308.279209] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 21 (level, low) -> IRQ 21 [28308.279241] PCI: Setting latency timer of device 0000:00:1b.0 to 64 [28308.279248] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:2069: chipset global capabilities = 0x4401 [28308.311620] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:754: codec_mask = 0x1 [28308.313776] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:2359: hda_codec: model 'dell-m6' is selected for config 1028:254 (unknown Dell) [28308.313926] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 0a pin config 0321101f [28308.314064] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 0b pin config 4f00000f [28308.314227] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 0c pin config 4f0000f0 [28308.314355] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 0d pin config 90170110 [28308.314499] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 0e pin config 03a11020 [28308.314670] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 0f pin config 0321101f [28308.314810] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 10 pin config 4f0000f0 [28308.314986] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 11 pin config 4f0000f0 [28308.315122] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 12 pin config 4f0000f0 [28308.315298] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 13 pin config 90a60160 [28308.315607] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 14 pin config 4f0000f0 [28308.315750] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 1e pin config 00000000 [28308.315896] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 22 pin config 4f0000f0 [28308.316125] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:1840: hda_codec: pin nid 13 pin config 90a60160 [28308.316771] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3030: autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) [28308.316778] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3034: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [28308.316783] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3038: hp_outs=2 (0xa/0xf/0x0/0x0/0x0) [28308.316789] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3039: mono: mono_out=0x0 [28308.316793] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:3047: inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 [28308.316800] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2249: stac92xx_add_dyn_out_pins: total dac count=3 [28308.385231] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Center Playback Volume, skipped [28308.385237] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave LFE Playback Volume, skipped [28308.385241] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Side Playback Volume, skipped [28308.385245] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker Playback Volume, skipped [28308.385249] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave External Speaker Playback Volume, skipped [28308.385252] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker2 Playback Volume, skipped [28308.385260] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Center Playback Switch, skipped [28308.385264] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave LFE Playback Switch, skipped [28308.385268] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Side Playback Switch, skipped [28308.385274] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker Playback Switch, skipped [28308.385278] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave External Speaker Playback Switch, skipped [28308.385283] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave Speaker2 Playback Switch, skipped [28308.385287] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:1136: Cannot find slave IEC958 Playback Switch, skipped [28308.385838] ACPI: PCI Interrupt 0000:01:00.1[B] -> GSI 17 (level, low) -> IRQ 17 [28308.385865] PCI: Setting latency timer of device 0000:01:00.1 to 64 [28308.385870] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:2069: chipset global capabilities = 0x1001 [28308.401066] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:754: codec_mask = 0x1 [28967.997654] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1390: azx_pcm_prepare: bufsize=0x10000, format=0x31 [28967.997678] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x15, stream=0x5, channel=0, format=0x31 [28968.004577] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x17, stream=0x5, channel=0, format=0x31 [28968.016553] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x16, stream=0x5, channel=0, format=0x31 [28968.024471] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x17, stream=0x5, channel=0, format=0x31 [29032.180075] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x15 [29032.180075] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x16 [29032.180075] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 [29032.180075] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 [29429.525365] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1390: azx_pcm_prepare: bufsize=0xff00, format=0x15 [29429.525395] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x15, stream=0x5, channel=0, format=0x15 [29429.529905] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x17, stream=0x5, channel=0, format=0x15 [29429.537901] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x16, stream=0x5, channel=2, format=0x15 [29429.557564] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:725: hda_codec_setup_stream: NID=0x17, stream=0x5, channel=4, format=0x15 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x15 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x16 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x15 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x16 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 [28761.447729] ALSA /home/andrea/src/alsa/alsa-driver-1.0.17/pci/hda/hda_codec.c:737: hda_codec_cleanup_stream: NID=0x17 nb-andrea description: Portable Computer product: Studio 1535 vendor: Dell Inc. serial: BWZ7T3J width: 32 bits capabilities: smbios-2.4 dmi-2.4 configuration: boot=normal chassis=portable uuid=44454C4C-5700-105A-8037-C2C04F54334A *-core description: Motherboard product: 0F700C vendor: Dell Inc. physical id: 0 serial: .BWZ7T3J.CN486438673307. slot: @ *-firmware description: BIOS vendor: Dell Inc. physical id: 0 version: A03 (07/02/2008) size: 64KiB capacity: 1984KiB capabilities: isa pci pcmcia pnp upgrade shadowing cdboot bootselect int13floppy720 int5printscreen int9keyboard int14serial int17printer int10video acpi usb agp smartbattery biosbootspecification netboot *-cpu description: CPU product: Intel(R) Core(TM)2 Duo CPU T8100 @ 2.10GHz vendor: Intel Corp. physical id: 400 bus info: cpu@0 version: 6.7.6 serial: 0001-0676-0000-0000-0000-0000 slot: Microprocessor size: 2101MHz capacity: 2101MHz width: 64 bits clock: 200MHz capabilities: fpu fpu_exception wp vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx x86-64 constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr sse4_1 lahf_lm ida cpufreq configuration: id=0 *-cache:0 description: L1 cache physical id: 700 size: 32KiB capacity: 32KiB capabilities: internal write-back data *-cache:1 description: L2 cache physical id: 701 size: 3MiB capacity: 3MiB clock: 66MHz (15.0ns) capabilities: pipeline-burst internal varies unified *-logicalcpu:0 description: Logical CPU physical id: 0.1 width: 64 bits capabilities: logical *-logicalcpu:1 description: Logical CPU physical id: 0.2 width: 64 bits capabilities: logical *-memory description: System Memory physical id: 1000 slot: System board or motherboard size: 4GiB *-bank:0 description: DIMM DDR Synchronous 667 MHz (1.5 ns) product: HYMP125S64CP8-Y5 vendor: AD00000000000000 physical id: 0 serial: 00001106 slot: DIMM_A size: 2GiB width: 64 bits clock: 667MHz (1.5ns) *-bank:1 description: DIMM DDR Synchronous 667 MHz (1.5 ns) 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34xx Series] vendor: ATI Technologies Inc physical id: 0.1 bus info: pci@0000:01:00.1 version: 00 width: 32 bits clock: 33MHz capabilities: pm pciexpress msi bus_master cap_list configuration: driver=HDA Intel latency=0 module=snd_hda_intel *-usb:0 description: USB Controller product: 82801H (ICH8 Family) USB UHCI Controller #4 vendor: Intel Corporation physical id: 1a bus info: pci@0000:00:1a.0 version: 03 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=uhci_hcd latency=0 module=uhci_hcd *-usbhost product: UHCI Host Controller vendor: Linux 2.6.25-2-686 uhci_hcd physical id: 1 bus info: usb@1 logical name: usb1 version: 2.06 capabilities: usb-1.10 configuration: driver=hub slots=2 speed=12.0MB/s *-usb:1 description: USB Controller product: 82801H (ICH8 Family) USB UHCI Controller #5 vendor: Intel Corporation physical id: 1a.1 bus info: pci@0000:00:1a.1 version: 03 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=uhci_hcd latency=0 module=uhci_hcd *-usbhost product: UHCI Host Controller vendor: Linux 2.6.25-2-686 uhci_hcd physical id: 1 bus info: usb@2 logical name: usb2 version: 2.06 capabilities: usb-1.10 configuration: driver=hub slots=2 speed=12.0MB/s *-usb:2 description: USB Controller product: 82801H (ICH8 Family) USB2 EHCI Controller #2 vendor: Intel Corporation physical id: 1a.7 bus info: pci@0000:00:1a.7 version: 03 width: 32 bits clock: 33MHz capabilities: pm debug bus_master cap_list configuration: driver=ehci_hcd latency=0 module=ehci_hcd *-usbhost product: EHCI Host Controller vendor: Linux 2.6.25-2-686 ehci_hcd physical id: 1 bus info: usb@6 logical name: usb6 version: 2.06 capabilities: usb-2.00 configuration: driver=hub slots=4 speed=480.0MB/s *-multimedia description: Audio device product: 82801H (ICH8 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 03 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=HDA Intel latency=0 module=snd_hda_intel *-pci:1 description: PCI bridge product: 82801H (ICH8 Family) PCI Express Port 1 vendor: Intel Corporation physical id: 1c bus info: pci@0000:00:1c.0 version: 03 width: 32 bits clock: 33MHz capabilities: pci pciexpress msi pm bus_master cap_list configuration: driver=pcieport-driver *-pci:2 description: PCI bridge product: 82801H (ICH8 Family) PCI Express Port 2 vendor: Intel Corporation physical id: 1c.1 bus info: pci@0000:00:1c.1 version: 03 width: 32 bits clock: 33MHz capabilities: pci pciexpress msi pm bus_master cap_list configuration: driver=pcieport-driver *-network UNCLAIMED description: Network controller product: BCM4322 802.11a/b/g/n Wireless LAN Controller vendor: Broadcom Corporation physical id: 0 bus info: pci@0000:0c:00.0 version: 01 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: latency=0 *-pci:3 description: PCI bridge product: 82801H (ICH8 Family) PCI Express Port 4 vendor: Intel Corporation physical id: 1c.3 bus info: pci@0000:00:1c.3 version: 03 width: 32 bits clock: 33MHz capabilities: pci pciexpress msi pm bus_master cap_list configuration: driver=pcieport-driver *-pci:4 description: PCI bridge product: 82801H (ICH8 Family) PCI Express Port 6 vendor: Intel Corporation physical id: 1c.5 bus info: pci@0000:00:1c.5 version: 03 width: 32 bits clock: 33MHz capabilities: pci pciexpress msi pm bus_master cap_list configuration: driver=pcieport-driver *-network description: Ethernet interface product: NetLink BCM5784M Gigabit Ethernet PCIe vendor: Broadcom Corporation physical id: 0 bus info: pci@0000:09:00.0 logical name: eth0 version: 10 serial: 00:21:70:71:70:82 capacity: 1GB/s width: 64 bits clock: 33MHz capabilities: pm vpd msi pciexpress bus_master cap_list ethernet physical tp 10bt 10bt-fd 100bt 100bt-fd 1000bt 1000bt-fd autonegotiation configuration: autonegotiation=on broadcast=yes driver=tg3 driverversion=3.91 latency=0 link=no module=tg3 multicast=yes port=twisted pair *-usb:3 description: USB Controller product: 82801H (ICH8 Family) USB UHCI Controller #1 vendor: Intel Corporation physical id: 1d bus info: pci@0000:00:1d.0 version: 03 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=uhci_hcd latency=0 module=uhci_hcd *-usbhost product: UHCI Host Controller vendor: Linux 2.6.25-2-686 uhci_hcd physical id: 1 bus info: usb@3 logical name: usb3 version: 2.06 capabilities: usb-1.10 configuration: driver=hub slots=2 speed=12.0MB/s *-usb:4 description: USB Controller product: 82801H (ICH8 Family) USB UHCI Controller #2 vendor: Intel Corporation physical id: 1d.1 bus info: pci@0000:00:1d.1 version: 03 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=uhci_hcd latency=0 module=uhci_hcd *-usbhost product: UHCI Host Controller vendor: Linux 2.6.25-2-686 uhci_hcd physical id: 1 bus info: usb@4 logical name: usb4 version: 2.06 capabilities: usb-1.10 configuration: driver=hub slots=2 speed=12.0MB/s *-usb:5 description: USB Controller product: 82801H (ICH8 Family) USB UHCI Controller #3 vendor: Intel Corporation physical id: 1d.2 bus info: pci@0000:00:1d.2 version: 03 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=uhci_hcd latency=0 module=uhci_hcd *-usbhost product: UHCI Host Controller vendor: Linux 2.6.25-2-686 uhci_hcd physical id: 1 bus info: usb@5 logical name: usb5 version: 2.06 capabilities: usb-1.10 configuration: driver=hub slots=2 speed=12.0MB/s *-usb:6 description: USB Controller product: 82801H (ICH8 Family) USB2 EHCI Controller #1 vendor: Intel Corporation physical id: 1d.7 bus info: pci@0000:00:1d.7 version: 03 width: 32 bits clock: 33MHz capabilities: pm debug bus_master cap_list configuration: driver=ehci_hcd latency=0 module=ehci_hcd *-usbhost product: EHCI Host Controller vendor: Linux 2.6.25-2-686 ehci_hcd physical id: 1 bus info: usb@7 logical name: usb7 version: 2.06 capabilities: usb-2.00 configuration: driver=hub slots=6 speed=480.0MB/s *-usb:0 description: Generic USB device product: Compact Wireless-G USB Adapter vendor: Cisco-Linksys physical id: 3 bus info: usb@7:3 version: 0.01 capabilities: usb-2.00 configuration: driver=rt73usb maxpower=300mA speed=480.0MB/s *-usb:1 UNCLAIMED description: Video product: Laptop_Integrated_Webcam_2M vendor: SONiX Technology Inc. physical id: 5 bus info: usb@7:5 version: 82.29 serial: CN0TX613724878570GRK capabilities: usb-2.00 configuration: maxpower=168mA speed=480.0MB/s *-pci:5 description: PCI bridge product: 82801 Mobile PCI Bridge vendor: Intel Corporation physical id: 1e bus info: pci@0000:00:1e.0 version: f3 width: 32 bits clock: 33MHz capabilities: pci bus_master cap_list *-firewire description: FireWire (IEEE 1394) product: R5C832 IEEE 1394 Controller vendor: Ricoh Co Ltd physical id: 1 bus info: pci@0000:03:01.0 version: 05 width: 32 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: driver=firewire_ohci latency=64 maxlatency=4 mingnt=2 module=firewire_ohci *-system:0 description: SD Host controller product: R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter vendor: Ricoh Co Ltd physical id: 1.1 bus info: pci@0000:03:01.1 version: 22 width: 32 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: driver=sdhci latency=64 module=sdhci *-system:1 description: System peripheral product: R5C592 Memory Stick Bus Host Adapter vendor: Ricoh Co Ltd physical id: 1.2 bus info: pci@0000:03:01.2 version: 12 width: 32 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: driver=ricoh-mmc latency=64 module=ricoh_mmc *-system:2 UNCLAIMED description: System peripheral product: xD-Picture Card Controller vendor: Ricoh Co Ltd physical id: 1.3 bus info: pci@0000:03:01.3 version: 12 width: 32 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: latency=64 *-generic UNCLAIMED product: Illegal Vendor ID vendor: Illegal Vendor ID physical id: 1.4 bus info: pci@0000:03:01.4 version: ff width: 32 bits clock: 66MHz capabilities: bus_master vga_palette cap_list configuration: latency=255 maxlatency=255 mingnt=255 *-isa description: ISA bridge product: 82801HEM (ICH8M) LPC Interface Controller vendor: Intel Corporation physical id: 1f bus info: pci@0000:00:1f.0 version: 03 width: 32 bits clock: 33MHz capabilities: isa bus_master cap_list configuration: latency=0 *-storage description: SATA controller product: 82801HBM/HEM (ICH8M/ICH8M-E) SATA AHCI Controller vendor: Intel Corporation physical id: 1f.2 bus info: pci@0000:00:1f.2 logical name: scsi0 logical name: scsi1 version: 03 width: 32 bits clock: 66MHz capabilities: storage msi pm bus_master cap_list emulated configuration: driver=ahci latency=0 module=ahci *-disk description: ATA Disk product: WDC WD2500BEVS-7 vendor: Western Digital physical id: 0 bus info: scsi@0:0.0.0 logical name: /dev/sda version: 01.0 serial: WD-WXC508631049 size: 232GiB (250GB) capabilities: partitioned 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filesystem=ntfs label=OS state=clean *-volume:3 description: Extended partition physical id: 4 bus info: scsi@0:0.0.0,4 logical name: /dev/sda4 size: 99GiB capacity: 99GiB capabilities: primary extended partitioned partitioned:extended *-logicalvolume:0 description: Linux filesystem partition physical id: 5 logical name: /dev/sda5 capacity: 9538MiB *-logicalvolume:1 description: Linux filesystem partition physical id: 6 logical name: /dev/sda6 logical name: / capacity: 9538MiB configuration: mount.fstype=ext3 mount.options=rw,noatime,errors=remount-ro,data=ordered state=mounted *-logicalvolume:2 description: Linux filesystem partition physical id: 7 logical name: /dev/sda7 logical name: /home capacity: 80GiB configuration: mount.fstype=ext3 mount.options=rw,errors=continue,data=ordered state=mounted *-cdrom description: DVD-RAM writer product: DVDRWBD BC-5600S vendor: Optiarc physical id: 1 bus info: scsi@1:0.0.0 logical name: /dev/cdrom logical name: /dev/cdrw logical name: /dev/dvd 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Battery Bay capacity: 52000mWh configuration: voltage=11.1V *-network description: Wireless interface physical id: 2 logical name: wlan0 serial: 00:1d:7e:97:83:82 capabilities: ethernet physical wireless configuration: broadcast=yes ip=192.168.1.69 multicast=yes wireless=IEEE 802.11g

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Re: [alsa-devel] WM9712 Not Giving output
by Marek Vasut 07 Aug '08

07 Aug '08

Dne Friday 01 of August 2008 19:59:44 Nitin Mahajan napsal(a): > HI! > > --- On Fri, 1/8/08, Marek Vasut <marek.vasut(a)gmail.com> wrote: > > From: Marek Vasut <marek.vasut(a)gmail.com> > > Subject: Re: [alsa-devel] WM9712 Not Giving output > > To: alsa-devel(a)alsa-project.org, nitinm76(a)yahoo.com > > Date: Friday, 1 August, 2008, 11:10 PM > > Dne Friday 01 of August 2008 19:27:24 Nitin Mahajan > > > > napsal(a): > > > HI! > > > > > > --- On Fri, 1/8/08, Mark Brown > > > > <broonie(a)opensource.wolfsonmicro.com> wrote: > > > > From: Mark Brown > > > > <broonie(a)opensource.wolfsonmicro.com> > > > > > > Subject: Re: [alsa-devel] WM9712 Not Giving > > > > output > > > > > > To: "Nitin Mahajan" > > > > <nitinm76(a)yahoo.com> > > > > > > Cc: alsa-devel(a)alsa-project.org, > > > > "Manuel" <mano(a)roarinelk.homelinux.net> > > > > > > Date: Friday, 1 August, 2008, 9:32 PM > > > > On Fri, Aug 01, 2008 at 08:31:03AM -0700, Nitin > > > > Mahajan > > > > > > wrote: > > > > > I am using a WM9712 codec chip with the AC97 > > > > > > > > controller on AMD AU1250. > > > > > > > > > Linux kernel version 2.6.24.2. > > > > > > > > > > On AC97 controller's side, I can see the > > > > DMA > > > > > > happening on the host > > > > > > > > > processor, interrupts also coming. > > > > > > > > This suggests a mixer configuration problem. > > > > > > > > > On the mixer front, I did unmute and set the > > > > vol to > > > > > > max for PCM and > > > > > > > > > Headphone. I did not find any Master control > > > > with this > > > > > > chip. Isn't > > > > > > > > > there any master control with this chip? > > > > > > > > No. > > > > > > If Master is not there, logically are the enough for > > > > playback? > > > > > amixer set 'PCM' 255 unmute > > > amixer set 'Headphone' 255 unmute > > > > > > > > Can some one please throw more light on the > > > > Mixer > > > > > > part, specific to > > > > > > > > > this chip. > > > > > > > > > > What Mixer controls do I need to set with > > > > what values > > > > > > for playback and > > > > > > > > > capture respectively with this chip? > > > > > > > > Check the configuration of the Left HP Mixer and > > > > Right HP > > > > > > Mixer PCM > > > > Playback switches. These should be enabled in > > > > order to > > > > > > play back via > > > > the headphone output. > > > > > > While writing the reply , the hardware is not with me. > > > But I wil check these tommorow. > > > > > > > If you still can't get audio output with that > > > > please > > > > > > send a copy of your > > > > mixer configuration - run the command > > > > > > > > alsactl store -f mixer-config > > > > > > > > to write a file 'mixer-config' with this > > > > information in it. Please > > > > also check your kernel log from when the driver > > > > is loaded > > > > > > and when you > > > > configure the mixer. > > > > > > > > There is documentation of the audio paths > > > > supported by the > > > > > > chip in the > > > > > > I saw the figure in the datasheet, its quite > > > > explainatory, as to which > > > > > registers are involved in which path. > > > > > > > datasheet, available from: > > > > > > > > http://www.wolfsonmicro.com/products/WM9712/ > > > > > > > > > > > > and you may find the audio route mappings in the > > > > WM9712 > > > > > > driver useful to > > > > refer to - the 'audio_map' table in the > > > > driver > > > > > > which shows how the parts > > > > of the device are connected. The format is { > > > > destination, > > > > > > control, > > > > source }. > > > > > > Thanks and regards > > > > > > -Nitin > > > > > > Send instant messages to your online friends > > > > http://uk.messenger.yahoo.com > > > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > Hi, > > try searching the mailing list for "[PATCH] wm9712 > > mono mixer". That may help > > you. > > Interesting, do you mean that "Right HP Mixer" and "Left HP Mixer" should > be "Right Mixer" and "Left Mixer" ? > > thanks and regards > > -Nitin yea, and you should see alsa warning about it in kernel log as well. > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Send instant messages to your online friends http://uk.messenger.yahoo.com This message wasnt CCed to alsa-devel, re-adding.

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[alsa-devel] Realtek ALC1200
by Stanislav David 06 Aug '08

06 Aug '08

Hello, do you have any information about realtek codec ALC1200. Is it supported in ALSA. I was not able to find any informations about it. Thank you Stanislav David ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

2 1

[alsa-devel] [PATCH] ASoC: Add WM8580 CODEC driver
by Mark Brown 06 Aug '08

06 Aug '08

The WM8580 is an audio CODEC designed for DVD and surround sound applications, offering three stereo DACs and a stereo ADC. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8580.c | 1055 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8580.h | 42 ++ 4 files changed, 1103 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8580.c create mode 100644 sound/soc/codecs/wm8580.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d7bacf6..eb79c5c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 select SND_SOC_UDA1380 select SND_SOC_WM8510 + select SND_SOC_WM8580 select SND_SOC_WM8731 select SND_SOC_WM8750 select SND_SOC_WM8753 @@ -38,6 +39,9 @@ config SND_SOC_UDA1380 config SND_SOC_WM8510 tristate +config SND_SOC_WM8580 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 98808d9..7c694ca 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ak4535-objs := ak4535.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o +snd-soc-wm8580-objs := wm8580.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -19,6 +20,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o +obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c new file mode 100644 index 0000000..df1ffbe --- /dev/null +++ b/sound/soc/codecs/wm8580.c @@ -0,0 +1,1055 @@ +/* + * wm8580.c -- WM8580 ALSA Soc Audio driver + * + * Copyright 2008 Wolfson Microelectronics PLC. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Notes: + * The WM8580 is a multichannel codec with S/PDIF support, featuring six + * DAC channels and two ADC channels. + * + * Currently only the primary audio interface is supported - S/PDIF and + * the secondary audio interfaces are not. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.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/tlv.h> +#include <sound/initval.h> +#include <asm/div64.h> + +#include "wm8580.h" + +#define AUDIO_NAME "wm8580" +#define WM8580_VERSION "0.1" + +struct pll_state { + unsigned int in; + unsigned int out; +}; + +/* codec private data */ +struct wm8580_priv { + struct pll_state a; + struct pll_state b; +}; + +/* WM8580 register space */ +#define WM8580_PLLA1 0x00 +#define WM8580_PLLA2 0x01 +#define WM8580_PLLA3 0x02 +#define WM8580_PLLA4 0x03 +#define WM8580_PLLB1 0x04 +#define WM8580_PLLB2 0x05 +#define WM8580_PLLB3 0x06 +#define WM8580_PLLB4 0x07 +#define WM8580_CLKSEL 0x08 +#define WM8580_PAIF1 0x09 +#define WM8580_PAIF2 0x0A +#define WM8580_SAIF1 0x0B +#define WM8580_PAIF3 0x0C +#define WM8580_PAIF4 0x0D +#define WM8580_SAIF2 0x0E +#define WM8580_DAC_CONTROL1 0x0F +#define WM8580_DAC_CONTROL2 0x10 +#define WM8580_DAC_CONTROL3 0x11 +#define WM8580_DAC_CONTROL4 0x12 +#define WM8580_DAC_CONTROL5 0x13 +#define WM8580_DIGITAL_ATTENUATION_DACL1 0x14 +#define WM8580_DIGITAL_ATTENUATION_DACR1 0x15 +#define WM8580_DIGITAL_ATTENUATION_DACL2 0x16 +#define WM8580_DIGITAL_ATTENUATION_DACR2 0x17 +#define WM8580_DIGITAL_ATTENUATION_DACL3 0x18 +#define WM8580_DIGITAL_ATTENUATION_DACR3 0x19 +#define WM8580_MASTER_DIGITAL_ATTENUATION 0x1C +#define WM8580_ADC_CONTROL1 0x1D +#define WM8580_SPDTXCHAN0 0x1E +#define WM8580_SPDTXCHAN1 0x1F +#define WM8580_SPDTXCHAN2 0x20 +#define WM8580_SPDTXCHAN3 0x21 +#define WM8580_SPDTXCHAN4 0x22 +#define WM8580_SPDTXCHAN5 0x23 +#define WM8580_SPDMODE 0x24 +#define WM8580_INTMASK 0x25 +#define WM8580_GPO1 0x26 +#define WM8580_GPO2 0x27 +#define WM8580_GPO3 0x28 +#define WM8580_GPO4 0x29 +#define WM8580_GPO5 0x2A +#define WM8580_INTSTAT 0x2B +#define WM8580_SPDRXCHAN1 0x2C +#define WM8580_SPDRXCHAN2 0x2D +#define WM8580_SPDRXCHAN3 0x2E +#define WM8580_SPDRXCHAN4 0x2F +#define WM8580_SPDRXCHAN5 0x30 +#define WM8580_SPDSTAT 0x31 +#define WM8580_PWRDN1 0x32 +#define WM8580_PWRDN2 0x33 +#define WM8580_READBACK 0x34 +#define WM8580_RESET 0x35 + +/* PLLB4 (register 7h) */ +#define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60 +#define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20 +#define WM8580_PLLB4_MCLKOUTSRC_PLLB 0x40 +#define WM8580_PLLB4_MCLKOUTSRC_OSC 0x60 + +#define WM8580_PLLB4_CLKOUTSRC_MASK 0x180 +#define WM8580_PLLB4_CLKOUTSRC_PLLACLK 0x080 +#define WM8580_PLLB4_CLKOUTSRC_PLLBCLK 0x100 +#define WM8580_PLLB4_CLKOUTSRC_OSCCLK 0x180 + +/* CLKSEL (register 8h) */ +#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 +#define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 + +/* AIF control 1 (registers 9h-bh) */ +#define WM8580_AIF_RATE_MASK 0x7 +#define WM8580_AIF_RATE_128 0x0 +#define WM8580_AIF_RATE_192 0x1 +#define WM8580_AIF_RATE_256 0x2 +#define WM8580_AIF_RATE_384 0x3 +#define WM8580_AIF_RATE_512 0x4 +#define WM8580_AIF_RATE_768 0x5 +#define WM8580_AIF_RATE_1152 0x6 + +#define WM8580_AIF_BCLKSEL_MASK 0x18 +#define WM8580_AIF_BCLKSEL_64 0x00 +#define WM8580_AIF_BCLKSEL_128 0x08 +#define WM8580_AIF_BCLKSEL_256 0x10 +#define WM8580_AIF_BCLKSEL_SYSCLK 0x18 + +#define WM8580_AIF_MS 0x20 + +#define WM8580_AIF_CLKSRC_MASK 0xc0 +#define WM8580_AIF_CLKSRC_PLLA 0x40 +#define WM8580_AIF_CLKSRC_PLLB 0x40 +#define WM8580_AIF_CLKSRC_MCLK 0xc0 + +/* AIF control 2 (registers ch-eh) */ +#define WM8580_AIF_FMT_MASK 0x03 +#define WM8580_AIF_FMT_RIGHTJ 0x00 +#define WM8580_AIF_FMT_LEFTJ 0x01 +#define WM8580_AIF_FMT_I2S 0x02 +#define WM8580_AIF_FMT_DSP 0x03 + +#define WM8580_AIF_LENGTH_MASK 0x0c +#define WM8580_AIF_LENGTH_16 0x00 +#define WM8580_AIF_LENGTH_20 0x04 +#define WM8580_AIF_LENGTH_24 0x08 +#define WM8580_AIF_LENGTH_32 0x0c + +#define WM8580_AIF_LRP 0x10 +#define WM8580_AIF_BCP 0x20 + +/* Powerdown Register 1 (register 32h) */ +#define WM8580_PWRDN1_PWDN 0x001 +#define WM8580_PWRDN1_ALLDACPD 0x040 + +/* Powerdown Register 2 (register 33h) */ +#define WM8580_PWRDN2_OSSCPD 0x001 +#define WM8580_PWRDN2_PLLAPD 0x002 +#define WM8580_PWRDN2_PLLBPD 0x004 +#define WM8580_PWRDN2_SPDIFPD 0x008 +#define WM8580_PWRDN2_SPDIFTXD 0x010 +#define WM8580_PWRDN2_SPDIFRXD 0x020 + +#define WM8580_DAC_CONTROL5_MUTEALL 0x10 + +/* + * wm8580 register cache + * We can't read the WM8580 register space when we + * are using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8580_reg[] = { + 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ + 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ + 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/ + 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ + 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ + 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ + 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R27*/ + 0x01f0, 0x0040, 0x0000, 0x0000, /*R31(0x1F)*/ + 0x0000, 0x0000, 0x0031, 0x000b, /*R35*/ + 0x0039, 0x0000, 0x0010, 0x0032, /*R39*/ + 0x0054, 0x0076, 0x0098, 0x0000, /*R43(0x2B)*/ + 0x0000, 0x0000, 0x0000, 0x0000, /*R47*/ + 0x0000, 0x0000, 0x005e, 0x003e, /*R51(0x33)*/ + 0x0000, 0x0000 /*R53*/ +}; + +/* + * read wm8580 register cache + */ +static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + return cache[reg]; +} + +/* + * write wm8580 register cache + */ +static inline void wm8580_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + + cache[reg] = value; +} + +/* + * write to the WM8580 register space + */ +static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + + /* Registers are 9 bits wide */ + value &= 0x1ff; + + switch (reg) { + case WM8580_RESET: + /* Uncached */ + break; + default: + if (value == wm8580_read_reg_cache(codec, reg)) + return 0; + } + + /* data is + * D15..D9 WM8580 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8580_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static inline unsigned int wm8580_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + default: + return wm8580_read_reg_cache(codec, reg); + } +} + +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); + +static int wm8580_out_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int reg2 = (kcontrol->private_value >> 24) & 0xff; + int ret; + u16 val; + + /* Clear the register cache so we write without VU set */ + wm8580_write_reg_cache(codec, reg, 0); + wm8580_write_reg_cache(codec, reg2, 0); + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* Now write again with the volume update bit set */ + val = wm8580_read_reg_cache(codec, reg); + wm8580_write(codec, reg, val | 0x0100); + + val = wm8580_read_reg_cache(codec, reg2); + wm8580_write(codec, reg2, val | 0x0100); + + return 0; +} + +#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \ + .private_value = (reg_left) | ((shift) << 8) | \ + ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } + +static const struct snd_kcontrol_new wm8580_snd_controls[] = { +SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL1, + WM8580_DIGITAL_ATTENUATION_DACR1, + 0, 0xff, 0, dac_tlv), +SOC_WM8580_OUT_DOUBLE_R_TLV("DAC2 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL2, + WM8580_DIGITAL_ATTENUATION_DACR2, + 0, 0xff, 0, dac_tlv), +SOC_WM8580_OUT_DOUBLE_R_TLV("DAC3 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL3, + WM8580_DIGITAL_ATTENUATION_DACR3, + 0, 0xff, 0, dac_tlv), + +SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0), +SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0), +SOC_SINGLE("DAC3 Deemphasis Switch", WM8580_DAC_CONTROL3, 2, 1, 0), + +SOC_DOUBLE("DAC1 Invert Switch", WM8580_DAC_CONTROL4, 0, 1, 1, 0), +SOC_DOUBLE("DAC2 Invert Switch", WM8580_DAC_CONTROL4, 2, 3, 1, 0), +SOC_DOUBLE("DAC3 Invert Switch", WM8580_DAC_CONTROL4, 4, 5, 1, 0), + +SOC_SINGLE("DAC ZC Switch", WM8580_DAC_CONTROL5, 5, 1, 0), +SOC_SINGLE("DAC1 Switch", WM8580_DAC_CONTROL5, 0, 1, 0), +SOC_SINGLE("DAC2 Switch", WM8580_DAC_CONTROL5, 1, 1, 0), +SOC_SINGLE("DAC3 Switch", WM8580_DAC_CONTROL5, 2, 1, 0), + +SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0), +SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), +}; + +/* Add non-DAPM controls */ +static int wm8580_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8580_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + return 0; +} +static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1), +SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1), +SND_SOC_DAPM_DAC("DAC3", "Playback", WM8580_PWRDN1, 4, 1), + +SND_SOC_DAPM_OUTPUT("VOUT1L"), +SND_SOC_DAPM_OUTPUT("VOUT1R"), +SND_SOC_DAPM_OUTPUT("VOUT2L"), +SND_SOC_DAPM_OUTPUT("VOUT2R"), +SND_SOC_DAPM_OUTPUT("VOUT3L"), +SND_SOC_DAPM_OUTPUT("VOUT3R"), + +SND_SOC_DAPM_ADC("ADC", "Capture", WM8580_PWRDN1, 1, 1), + +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + { "VOUT1L", NULL, "DAC1" }, + { "VOUT1R", NULL, "DAC1" }, + + { "VOUT2L", NULL, "DAC2" }, + { "VOUT2R", NULL, "DAC2" }, + + { "VOUT3L", NULL, "DAC3" }, + { "VOUT3R", NULL, "DAC3" }, + + { "ADC", NULL, "AINL" }, + { "ADC", NULL, "AINR" }, +}; + +static int wm8580_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets, + ARRAY_SIZE(wm8580_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 prescale:1; + u32 postscale:1; + u32 freqmode:2; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the pll divide */ +#define FIXED_PLL_SIZE (1 << 22) + +/* PLL rate to output rate divisions */ +static struct { + unsigned int div; + unsigned int freqmode; + unsigned int postscale; +} post_table[] = { + { 2, 0, 0 }, + { 4, 0, 1 }, + { 4, 1, 0 }, + { 8, 1, 1 }, + { 8, 2, 0 }, + { 16, 2, 1 }, + { 12, 3, 0 }, + { 24, 3, 1 } +}; + +static int pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + int i; + + pr_debug("wm8580: PLL %dHz->%dHz\n", source, target); + + /* Scale the output frequency up; the PLL should run in the + * region of 90-100MHz. + */ + for (i = 0; i < ARRAY_SIZE(post_table); i++) { + if (target * post_table[i].div >= 90000000 && + target * post_table[i].div <= 100000000) { + pll_div->freqmode = post_table[i].freqmode; + pll_div->postscale = post_table[i].postscale; + target *= post_table[i].div; + break; + } + } + + if (i == ARRAY_SIZE(post_table)) { + printk(KERN_ERR "wm8580: Unable to scale output frequency " + "%u\n", target); + return -EINVAL; + } + + Ndiv = target / source; + + if (Ndiv < 5) { + source /= 2; + pll_div->prescale = 1; + Ndiv = target / source; + } else + pll_div->prescale = 0; + + if ((Ndiv < 5) || (Ndiv > 13)) { + printk(KERN_ERR + "WM8580 N=%d outside supported range\n", Ndiv); + return -EINVAL; + } + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + pll_div->k = K; + + pr_debug("PLL %x.%x prescale %d freqmode %d postscale %d\n", + pll_div->n, pll_div->k, pll_div->prescale, pll_div->freqmode, + pll_div->postscale); + + return 0; +} + +static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + int offset; + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8580_priv *wm8580 = codec->private_data; + struct pll_state *state; + struct _pll_div pll_div; + unsigned int reg; + unsigned int pwr_mask; + int ret; + + /* GCC isn't able to work out the ifs below for initialising/using + * pll_div so suppress warnings. + */ + memset(&pll_div, 0, sizeof(pll_div)); + + switch (pll_id) { + case WM8580_PLLA: + state = &wm8580->a; + offset = 0; + pwr_mask = WM8580_PWRDN2_PLLAPD; + break; + case WM8580_PLLB: + state = &wm8580->b; + offset = 4; + pwr_mask = WM8580_PWRDN2_PLLBPD; + break; + default: + return -ENODEV; + } + + if (freq_in && freq_out) { + ret = pll_factors(&pll_div, freq_out, freq_in); + if (ret != 0) + return ret; + } + + state->in = freq_in; + state->out = freq_out; + + /* Always disable the PLL - it is not safe to leave it running + * while reprogramming it. + */ + reg = wm8580_read(codec, WM8580_PWRDN2); + wm8580_write(codec, WM8580_PWRDN2, reg | pwr_mask); + + if (!freq_in || !freq_out) + return 0; + + wm8580_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff); + wm8580_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff); + wm8580_write(codec, WM8580_PLLA3 + offset, + (pll_div.k >> 18 & 0xf) | (pll_div.n << 4)); + + reg = wm8580_read(codec, WM8580_PLLA4 + offset); + reg &= ~0x3f; + reg |= pll_div.prescale | pll_div.postscale << 1 | + pll_div.freqmode << 4; + + wm8580_write(codec, WM8580_PLLA4 + offset, reg); + + /* All done, turn it on */ + reg = wm8580_read(codec, WM8580_PWRDN2); + wm8580_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id); + + paifb &= ~WM8580_AIF_LENGTH_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + paifb |= WM8580_AIF_LENGTH_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + paifb |= WM8580_AIF_LENGTH_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + paifb |= WM8580_AIF_LENGTH_24; + break; + default: + return -EINVAL; + } + + wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb); + return 0; +} + +static int wm8580_set_paif_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int aifa; + unsigned int aifb; + int can_invert_lrclk; + + aifa = wm8580_read(codec, WM8580_PAIF1 + codec_dai->id); + aifb = wm8580_read(codec, WM8580_PAIF3 + codec_dai->id); + + aifb &= ~(WM8580_AIF_FMT_MASK | WM8580_AIF_LRP | WM8580_AIF_BCP); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aifa &= ~WM8580_AIF_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aifa |= WM8580_AIF_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + can_invert_lrclk = 1; + aifb |= WM8580_AIF_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + can_invert_lrclk = 1; + aifb |= WM8580_AIF_FMT_RIGHTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + can_invert_lrclk = 1; + aifb |= WM8580_AIF_FMT_LEFTJ; + break; + case SND_SOC_DAIFMT_DSP_A: + can_invert_lrclk = 0; + aifb |= WM8580_AIF_FMT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B: + can_invert_lrclk = 0; + aifb |= WM8580_AIF_FMT_DSP; + aifb |= WM8580_AIF_LRP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_IB_IF: + if (!can_invert_lrclk) + return -EINVAL; + aifb |= WM8580_AIF_BCP; + aifb |= WM8580_AIF_LRP; + break; + + case SND_SOC_DAIFMT_IB_NF: + aifb |= WM8580_AIF_BCP; + break; + + case SND_SOC_DAIFMT_NB_IF: + if (!can_invert_lrclk) + return -EINVAL; + aifb |= WM8580_AIF_LRP; + break; + + default: + return -EINVAL; + } + + wm8580_write(codec, WM8580_PAIF1 + codec_dai->id, aifa); + wm8580_write(codec, WM8580_PAIF3 + codec_dai->id, aifb); + + return 0; +} + +static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg; + + switch (div_id) { + case WM8580_MCLK: + reg = wm8580_read(codec, WM8580_PLLB4); + reg &= ~WM8580_PLLB4_MCLKOUTSRC_MASK; + + switch (div) { + case WM8580_CLKSRC_MCLK: + /* Input */ + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_PLLB4_MCLKOUTSRC_PLLA; + break; + case WM8580_CLKSRC_PLLB: + reg |= WM8580_PLLB4_MCLKOUTSRC_PLLB; + break; + + case WM8580_CLKSRC_OSC: + reg |= WM8580_PLLB4_MCLKOUTSRC_OSC; + break; + + default: + return -EINVAL; + } + wm8580_write(codec, WM8580_PLLB4, reg); + break; + + case WM8580_DAC_CLKSEL: + reg = wm8580_read(codec, WM8580_CLKSEL); + reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK; + + switch (div) { + case WM8580_CLKSRC_MCLK: + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLB; + break; + + default: + return -EINVAL; + } + wm8580_write(codec, WM8580_CLKSEL, reg); + break; + + case WM8580_CLKOUTSRC: + reg = wm8580_read(codec, WM8580_PLLB4); + reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; + + switch (div) { + case WM8580_CLKSRC_NONE: + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_PLLB4_CLKOUTSRC_PLLACLK; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_PLLB4_CLKOUTSRC_PLLBCLK; + break; + + case WM8580_CLKSRC_OSC: + reg |= WM8580_PLLB4_CLKOUTSRC_OSCCLK; + break; + + default: + return -EINVAL; + } + wm8580_write(codec, WM8580_PLLB4, reg); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg; + + reg = wm8580_read(codec, WM8580_DAC_CONTROL5); + + if (mute) + reg |= WM8580_DAC_CONTROL5_MUTEALL; + else + reg &= ~WM8580_DAC_CONTROL5_MUTEALL; + + wm8580_write(codec, WM8580_DAC_CONTROL5, reg); + + return 0; +} + +static int wm8580_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + reg = wm8580_read(codec, WM8580_PWRDN1); + wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai wm8580_dai[] = { + { + .name = "WM8580 PAIFRX", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8580_FORMATS, + }, + .ops = { + .hw_params = wm8580_paif_hw_params, + }, + .dai_ops = { + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, + .digital_mute = wm8580_digital_mute, + }, + }, + { + .name = "WM8580 PAIFTX", + .id = 1, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8580_FORMATS, + }, + .ops = { + .hw_params = wm8580_paif_hw_params, + }, + .dai_ops = { + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, + }, + }, +}; +EXPORT_SYMBOL_GPL(wm8580_dai); + +/* + * initialise the WM8580 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8580_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8580"; + codec->owner = THIS_MODULE; + codec->read = wm8580_read_reg_cache; + codec->write = wm8580_write; + codec->set_bias_level = wm8580_set_bias_level; + codec->dai = wm8580_dai; + codec->num_dai = ARRAY_SIZE(wm8580_dai); + codec->reg_cache_size = ARRAY_SIZE(wm8580_reg); + codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg), + GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Get the codec into a known state */ + wm8580_write(codec, WM8580_RESET, 0); + + /* Power up and get individual control of the DACs */ + wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) & + ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD)); + + /* Make VMID high impedence */ + wm8580_write(codec, WM8580_ADC_CONTROL1, + wm8580_read(codec, WM8580_ADC_CONTROL1) & ~0x100); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, + SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8580: failed to create pcms\n"); + goto pcm_err; + } + + wm8580_add_controls(codec); + wm8580_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8580: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static struct snd_soc_device *wm8580_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8580 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8580_i2c_driver; +static struct i2c_client client_template; + +static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8580_socdev; + struct wm8580_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8580_init(socdev); + if (ret < 0) { + dev_err(&i2c->dev, "failed to initialise WM8580\n"); + goto err; + } + + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8580_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8580_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8580_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8580_i2c_driver = { + .driver = { + .name = "WM8580 I2C Codec", + .owner = THIS_MODULE, + }, + .attach_adapter = wm8580_i2c_attach, + .detach_client = wm8580_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8580", + .driver = &wm8580_i2c_driver, +}; +#endif + +static int wm8580_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8580_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8580_priv *wm8580; + int ret = 0; + + pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL); + if (wm8580 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = wm8580; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8580_socdev = socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8580_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8580_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8580_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8580 = { + .probe = wm8580_probe, + .remove = wm8580_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); + +MODULE_DESCRIPTION("ASoC WM8580 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h new file mode 100644 index 0000000..589ddab --- /dev/null +++ b/sound/soc/codecs/wm8580.h @@ -0,0 +1,42 @@ +/* + * wm8580.h -- audio driver for WM8580 + * + * Copyright 2008 Samsung Electronics. + * Author: Ryu Euiyoul + * ryu.real(a)gmail.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _WM8580_H +#define _WM8580_H + +#define WM8580_PLLA 1 +#define WM8580_PLLB 2 + +#define WM8580_MCLK 1 +#define WM8580_DAC_CLKSEL 2 +#define WM8580_CLKOUTSRC 3 + +#define WM8580_CLKSRC_MCLK 1 +#define WM8580_CLKSRC_PLLA 2 +#define WM8580_CLKSRC_PLLB 3 +#define WM8580_CLKSRC_OSC 4 +#define WM8580_CLKSRC_NONE 5 + +struct wm8580_setup_data { + unsigned short i2c_address; +}; + +#define WM8580_DAI_PAIFRX 0 +#define WM8580_DAI_PAIFTX 1 + +extern struct snd_soc_dai wm8580_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_wm8580; + +#endif + -- 1.5.6.3

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[alsa-devel] HDA Power management.
by Tobin Davis 06 Aug '08

06 Aug '08

Ok, I've been studying the code for a couple of weeks now, and testing on a mobile type development platform with a Realtek ALC262, but I have not been able to get the audio codec to enter low power mode (D3). I can see it switch when running that other OS. When running, the chip draws ~400mw, but drops to 40mw after switching to low power mode. How do I get it to do the same with the current driver? I'm using the git tip for alsa, and 2.6.26 final kernel. I have tried setting the power mode to D3 using the hda-tool from Takashi, but the codec reverts to D0 (full power) after a second or two, even though there is nothing touching the audio in my test environment. -- Tobin Davis The beauty of a pun is in the "Oy!" of the beholder.

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[alsa-devel] [PATCH] ALSA: wss_lib: rework snd_ad1848_probe()
by Rene Herman 06 Aug '08

06 Aug '08

Hi Takashi. This reworks snd_ad1848_probe(), making for IMO significantly more straightforward reading. The original had a somewhat odd try-loop where it not only waited for the codec to come out of init but also retried a failed basic detect which seems to not make sense. Except for that, the detection logic should be unchanged. I am by the way rather unsure what it is that the spin_lock is trying to protect us from here. What's the error scenario if we just not take it? We aren't live yet... Next step is folding this into snd_wss_probe() directly. Rene.

2 2

[alsa-devel] [PATCH] ALSA: ASoC V2: Improve pause/unpause performance in Freescale 8610 drivers
by Timur Tabi 06 Aug '08

06 Aug '08

Add support for true pause and unpause. Without this, mplayer will drop some audio (less than one second, but still noticeable) when pausing playback. Remove support for PM suspend and resume from the trigger function, since the driver doesn't support PM anyway. Optimize the delay after starting capture. Instead of delaying 1ms, the driver now polls the hardware. The new delay is shorter by over 90% yet still effective. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2 only. sound/soc/fsl/fsl_dma.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 17 +++++++++-------- 2 files changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 83f9c1d..649eac6 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -122,7 +122,8 @@ struct fsl_dma_private { static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7ccd37d..916743f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -409,28 +409,29 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* - * I think we need this delay to allow time for the SSI - * to put data into its FIFO. Without it, ALSA starts - * to complain about overruns. + * Wait until the SSI has filled its FIFO. Without this + * delay, ALSA complains about overruns. When the FIFO + * is full, the DMA controller initiates its first + * transfer. Until then, however, the DMA's DAR + * register is zero, which translates to an + * out-of-bounds pointer. This makes ALSA think an + * overrun has occurred. */ - mdelay(1); + while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)); } break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); -- 1.5.5

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Re: [alsa-devel] [PATCH] ALSA: wss_lib: rework snd_ad1848_probe()
by krzysztof.h1＠poczta.fm 06 Aug '08

06 Aug '08

> Hi Takashi. > > This reworks snd_ad1848_probe(), making for IMO significantly more > straightforward reading. > > The original had a somewhat odd try-loop where it not only waited for > the codec to come out of init but also retried a failed basic detect > which seems to not make sense. Except for that, the detection logic > should be unchanged. > Acked-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > I am by the way rather unsure what it is that the spin_lock is trying to > protect us from here. What's the error scenario if we just not take it? > We aren't live yet... > > Next step is folding this into snd_wss_probe() directly. > I would like to leave it separate as 1. End of this function is good check point of correct detection for ad1848 chips, 2. The snd_wss_probe() is long enough. But I won't oppose it fiercely. Krzysztof ---------------------------------------------------------------------- Tani Telefon! Sprawdz >>> http://link.interia.pl/f1ec0

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[PATCH] ALSA: wss_lib: rework snd_ad1848_probe()
by Rene Herman 06 Aug '08

06 Aug '08

Make snd_ad1848_probe() easier to follow. With an exception for only trying once as soon as the codec is out of init (which should be all that's needed) the detection logic should be unchanged. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/wss/wss_lib.c | 131 ++++++++++++++++++++++++----------------------- 1 files changed, 67 insertions(+), 64 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 955db45..802474b 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1147,79 +1147,82 @@ static snd_pcm_uframes_t snd_wss_capture_pointer(struct snd_pcm_substream *subst static int snd_ad1848_probe(struct snd_wss *chip) { + unsigned long timeout = jiffies + msecs_to_jiffies(1000); unsigned long flags; - int i, id, rev, ad1847; + unsigned char r; + unsigned short hardware = 0; + int err = 0; + int i; - id = 0; - ad1847 = 0; - for (i = 0; i < 1000; i++) { - mb(); - if (inb(chip->port + CS4231P(REGSEL)) & CS4231_INIT) - msleep(1); - else { - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, 0x00); - snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); - snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); - rev = snd_wss_in(chip, CS4231_RIGHT_INPUT); - if (rev == 0x65) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - ad1847 = 1; - break; - } - if (rev == 0x45) { - rev = snd_wss_in(chip, CS4231_LEFT_INPUT); - if (rev == 0xaa || rev == 0x8a) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - break; - } - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - } + while (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { + if (time_after(jiffies, timeout)) + return -ENODEV; + cond_resched(); } - if (id != 1) - return -ENODEV; /* no valid device found */ - id = 0; - if (chip->hardware == WSS_HW_DETECT) - id = ad1847 ? WSS_HW_AD1847 : WSS_HW_AD1848; - spin_lock_irqsave(&chip->reg_lock, flags); - inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ - outb(0, chip->port + CS4231P(STATUS)); - mb(); - if (id == WSS_HW_AD1848) { - /* check if there are more than 16 registers */ - rev = snd_wss_in(chip, CS4231_MISC_INFO); - snd_wss_out(chip, CS4231_MISC_INFO, 0x40); - for (i = 0; i < 16; ++i) { - if (snd_wss_in(chip, i) != snd_wss_in(chip, i + 16)) { - id = WSS_HW_CMI8330; - break; - } + + /* set CS423x MODE 1 */ + snd_wss_out(chip, CS4231_MISC_INFO, 0); + + snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */ + r = snd_wss_in(chip, CS4231_RIGHT_INPUT); + if (r != 0x45) { + /* RMGE always high on AD1847 */ + if ((r & ~CS4231_ENABLE_MIC_GAIN) != 0x45) { + err = -ENODEV; + goto out; + } + hardware = WSS_HW_AD1847; + } else { + snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); + r = snd_wss_in(chip, CS4231_LEFT_INPUT); + /* L/RMGE always low on AT2320 */ + if ((r | CS4231_ENABLE_MIC_GAIN) != 0xaa) { + err = -ENODEV; + goto out; } - snd_wss_out(chip, CS4231_MISC_INFO, 0x00); - if (id != WSS_HW_CMI8330 && (rev & 0x80)) - id = WSS_HW_CS4248; - if (id == WSS_HW_CMI8330 && (rev & 0x0f) != 0x0a) - id = 0; } - if (id == WSS_HW_CMI8330) { - /* verify it is not CS4231 by changing the version register */ - /* on CMI8330 it is volume control register and can be set 0 */ - snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - snd_wss_dout(chip, CS4231_VERSION, 0x00); - rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; - if (rev) - id = 0; - snd_wss_out(chip, CS4231_MISC_INFO, 0); + + /* clear pending IRQ */ + wss_inb(chip, CS4231P(STATUS)); + wss_outb(chip, CS4231P(STATUS), 0); + mb(); + + if ((chip->hardware & WSS_HW_TYPE_MASK) != WSS_HW_DETECT) + goto out; + + if (hardware) { + chip->hardware = hardware; + goto out; } - if (id) - chip->hardware = id; + r = snd_wss_in(chip, CS4231_MISC_INFO); + + /* set CS423x MODE 2 */ + snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + for (i = 0; i < 16; i++) { + if (snd_wss_in(chip, i) != snd_wss_in(chip, 16 + i)) { + /* we have more than 16 registers: check ID */ + if ((r & 0xf) != 0xa) + goto out; + /* + * on CMI8330, CS4231_VERSION is volume control and + * can be set to 0 + */ + snd_wss_dout(chip, CS4231_VERSION, 0); + r = snd_wss_in(chip, CS4231_VERSION) & 0xe7; + if (!r) + chip->hardware = WSS_HW_CMI8330; + goto out; + } + } + if (r & 0x80) + chip->hardware = WSS_HW_CS4248; + else + chip->hardware = WSS_HW_AD1848; +out: spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; /* all things are ok.. */ + return err; } static int snd_wss_probe(struct snd_wss *chip) -- 1.5.5 --------------060807000500060608070809--

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[PATCH] ALSA: wss_lib: rework snd_ad1848_probe()
by Rene Herman 06 Aug '08

06 Aug '08

Make snd_ad1848_probe() easier to follow. With an exception for only trying once as soon as the codec is out of init (which should be all that's needed) the detection logic should be unchanged. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/wss/wss_lib.c | 133 ++++++++++++++++++++++++---------------------- 1 files changed, 69 insertions(+), 64 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 955db45..cbe9565 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1147,79 +1147,84 @@ static snd_pcm_uframes_t snd_wss_capture_pointer(struct snd_pcm_substream *subst static int snd_ad1848_probe(struct snd_wss *chip) { + unsigned long timeout = jiffies + msecs_to_jiffies(1000); unsigned long flags; - int i, id, rev, ad1847; + unsigned char r; + unsigned short hardware = 0; + int err = 0; + int i; - id = 0; - ad1847 = 0; - for (i = 0; i < 1000; i++) { - mb(); - if (inb(chip->port + CS4231P(REGSEL)) & CS4231_INIT) - msleep(1); - else { - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, 0x00); - snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); - snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); - rev = snd_wss_in(chip, CS4231_RIGHT_INPUT); - if (rev == 0x65) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - ad1847 = 1; - break; - } - if (rev == 0x45) { - rev = snd_wss_in(chip, CS4231_LEFT_INPUT); - if (rev == 0xaa || rev == 0x8a) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - break; - } - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - } + while (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { + if (time_after(jiffies, timeout)) + return -ENODEV; + cond_resched(); } - if (id != 1) - return -ENODEV; /* no valid device found */ - id = 0; - if (chip->hardware == WSS_HW_DETECT) - id = ad1847 ? WSS_HW_AD1847 : WSS_HW_AD1848; - spin_lock_irqsave(&chip->reg_lock, flags); - inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ - outb(0, chip->port + CS4231P(STATUS)); - mb(); - if (id == WSS_HW_AD1848) { - /* check if there are more than 16 registers */ - rev = snd_wss_in(chip, CS4231_MISC_INFO); - snd_wss_out(chip, CS4231_MISC_INFO, 0x40); - for (i = 0; i < 16; ++i) { - if (snd_wss_in(chip, i) != snd_wss_in(chip, i + 16)) { - id = WSS_HW_CMI8330; - break; - } + + /* set CS423x MODE 1 */ + snd_wss_out(chip, CS4231_MISC_INFO, 0); + + snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */ + r = snd_wss_in(chip, CS4231_RIGHT_INPUT); + if (r != 0x45) { + /* RMGE always high on AD1847 */ + if ((r & ~CS4231_ENABLE_MIC_GAIN) != 0x45) { + err = -ENODEV; + goto out; + } + hardware = WSS_HW_AD1847; + } else { + snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); + r = snd_wss_in(chip, CS4231_LEFT_INPUT); + /* L/RMGE always low on AT2320 */ + if ((r | CS4231_ENABLE_MIC_GAIN) != 0xaa) { + err = -ENODEV; + goto out; } - snd_wss_out(chip, CS4231_MISC_INFO, 0x00); - if (id != WSS_HW_CMI8330 && (rev & 0x80)) - id = WSS_HW_CS4248; - if (id == WSS_HW_CMI8330 && (rev & 0x0f) != 0x0a) - id = 0; } - if (id == WSS_HW_CMI8330) { - /* verify it is not CS4231 by changing the version register */ - /* on CMI8330 it is volume control register and can be set 0 */ - snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - snd_wss_dout(chip, CS4231_VERSION, 0x00); - rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; - if (rev) - id = 0; - snd_wss_out(chip, CS4231_MISC_INFO, 0); + + /* clear pending IRQ */ + wss_inb(chip, CS4231P(STATUS)); + wss_outb(chip, CS4231P(STATUS), 0); + mb(); + + if ((chip->hardware & WSS_HW_TYPE_MASK) != WSS_HW_DETECT) + goto out; + + if (hardware) { + chip->hardware = hardware; + goto out; } - if (id) - chip->hardware = id; + r = snd_wss_in(chip, CS4231_MISC_INFO); + + /* set CS423x MODE 2 */ + snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + for (i = 0; i < 16; i++) { + if (snd_wss_in(chip, i) != snd_wss_in(chip, 16 + i)) { + /* we have more than 16 registers: check ID */ + if ((r & 0xf) != 0xa) + goto out_mode; + /* + * on CMI8330, CS4231_VERSION is volume control and + * can be set to 0 + */ + snd_wss_dout(chip, CS4231_VERSION, 0); + r = snd_wss_in(chip, CS4231_VERSION) & 0xe7; + if (!r) + chip->hardware = WSS_HW_CMI8330; + goto out_mode; + } + } + if (r & 0x80) + chip->hardware = WSS_HW_CS4248; + else + chip->hardware = WSS_HW_AD1848; +out_mode: + snd_wss_out(chip, CS4231_MISC_INFO, 0); +out: spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; /* all things are ok.. */ + return err; } static int snd_wss_probe(struct snd_wss *chip) -- 1.5.5 --------------080803020607030707060103--

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[alsa-devel] dmix in a forked process
by Bankim Bhavsar 05 Aug '08

05 Aug '08

Don't know why the earlier message got truncated so here it is again... Reading the source alsa-lib I see that dmix runs in a forked process *without* exec. file: src/pcm/pcm_direct.c function: snd_pcm_direct_server_create ret = fork(); if (ret < 0) { close(dmix->server_fd); return ret; } else if (ret == 0) { ret = fork(); if (ret == 0) server_job(dmix); _exit(EXIT_SUCCESS); } else { waitpid(ret, NULL, 0); } _From what I understand, this is a concern since if our app is the first to start dmix then a copy of the process (app) will remain till the system is shutdown ? Also I see that from alsa-lib 1.0.12 a check is performed before starting dmix. file: src/pcm/pcm_dmix.c function: snd_pcm_dmix_open if (dmix->shmptr->use_server) { dmix->server_free = dmix_server_free; ret = snd_pcm_direct_server_create(dmix); if (ret < 0) { SNDERR("unable to create server"); goto _err; } } When I checked how dmix->shmptr->use_server is set file: src/pcm/pcm_dmix.c function: snd_pcm_direct_initialize_slave int ver = 0; ioctl(spcm->poll_fd, SNDRV_PCM_IOCTL_PVERSION, &ver); if (ver < SNDRV_PROTOCOL_VERSION(2, 0, 8)) dmix->shmptr->use_server = 1; So if sndrv version is older than 2.0.8, use_server is set! So even using alsa-lib greater than 1.0.12 doesn't guarantee that the forked process (without exec) code path will be avoided. I'm currently porting the sound code of our app from OSS to ALSA but this problem is preventing me from doing so. Is there any way to get around this problem? Help here will be greatly appreciated. If this is an incorrect list then kindly fwd it to the appropriate one. Thanks, Bankim. PS: I've tried using the ioctl mentioned above after copying the necessary macros from kernel headers but ioctl returns error "Inappropriate ioctl for this device" on my Ubuntu 7.04 host.

3 4

[alsa-devel] [PATCH] ac97 mono-out w/internal speaker
by george pee 05 Aug '08

05 Aug '08

I have a wireless tablet (webdt310) http://dtresearch.com/product/wt310/index.htm It has cs5536 audio and a realtek alc203 codec. The headphone jack works perfectly, but the internal speaker was not. After a bit a twiddling and reading the ALC203 data sheet, I finally figured out that the function of Pin 37 had to be changed to MONO-OUT to get sound out of the internal speaker. I'm not sure that this is the best approach (maybe there could be some pin sensing?) but I added this as an ac97 quirk. Here's the patch. diff -rub linux-2.6.26.orig/sound/pci/ac97/ac97_codec.c linux-2.6.26.new/sound/pci/ac97/ac97_codec.c --- linux-2.6.26.orig/sound/pci/ac97/ac97_codec.c 2008-07-29 15:30:57.000000000 -0500 +++ linux-2.6.26.new/sound/pci/ac97/ac97_codec.c 2008-08-04 10:58:12.000000000 -0500 @@ -2709,6 +2710,12 @@ return snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_alc_jack_detect, ac97)); } +/* ac97 tune: turn on mono-out */ +static int tune_mono_out(struct snd_ac97 *ac97) +{ + snd_ac97_update_bits(ac97, 0x7a, 0x400, 0x400); + return 1; +} + /* ac97 tune: inversed EAPD bit */ static int tune_inv_eapd(struct snd_ac97 *ac97) { @@ -2801,6 +2808,7 @@ { "inv_eapd", tune_inv_eapd }, { "mute_led", tune_mute_led }, { "hp_mute_led", tune_hp_mute_led }, + { "mono_out", tune_mono_out }, };

2 3

Re: [alsa-devel] [PATCH 07/11] wss_lib: use CS4231P instead of AD1848P (kill the AD1848P)
by krzysztof.h1＠poczta.fm 05 Aug '08

05 Aug '08

> On 31-07-08 21:07, Krzysztof Helt wrote: > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Use CS4231P instead of AD1848P (kill the AD1848P). > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > --- > > Changes since previous version: > > > > 1. No reformatting of the snd_ad1848_debug() > > as the file is going to be killed. > > 1. Nicer formatting of WSS_SINGLE and WSS_DOUBLE > > macros. > > I believe 1 no. 2 is a cut & paste error? There are no such changes in > this patch. > Right. Regards, Krzysztof ---------------------------------------------------------------------- Wymien zeszyty na notebooka! Sprawdz >>> http://link.interia.pl/f1eab

1 0

Re: [alsa-devel] ultra small alsa library
by Marco Schuster 05 Aug '08

05 Aug '08

The problem I have is: As soon as I try to play a mp3 via mpg123, in dmesg comes the following error: Division by zero in kernel. [<c0026e2c>] (dump_stack+0x0/0x14) from [<c0026e58>] (__div0+0x18/0x20) [<c0026e40>] (__div0+0x0/0x20) from [<c011c65c>] (Ldiv0+0x8/0x10) [<bf0267ec>] (iso_stream_find+0x0/0x340 [ehci_hcd]) from [<bf028140>] (ehci_urb_enqueue+0x848/0xdf4 [ehci_hcd]) [<bf0278f8>] (ehci_urb_enqueue+0x0/0xdf4 [ehci_hcd]) from [<bf005228>] (usb_hcd_submit_urb+0x660/0x77c [usbcore]) [<bf004bc8>] (usb_hcd_submit_urb+0x0/0x77c [usbcore]) from [<bf0057fc>] (usb_submit_urb+0x228/0x264 [usbcore]) [<bf0055d4>] (usb_submit_urb+0x0/0x264 [usbcore]) from [<bf1c2820>] (start_urbs+0x1cc/0x308 [snd_usb_audio]) r6:00000000 r5:c0c2dc10 r4:00000010 [<bf1c2654>] (start_urbs+0x0/0x308 [snd_usb_audio]) from [<bf1c4d78>] (snd_usb_pcm_prepare+0xa4/0xb8 [snd_usb_audio]) r7:00000000 r6:00000000 r5:c0c2b800 r4:c0c2dc10 [<bf1c4cd4>] (snd_usb_pcm_prepare+0x0/0xb8 [snd_usb_audio]) from [<bf19bdc4>] (snd_pcm_do_prepare+0x1c/0x3c [snd_pcm]) r6:00000802 r5:c13d02c0 r4:c13d02c0 [<bf19bda8>] (snd_pcm_do_prepare+0x0/0x3c [snd_pcm]) from [<bf19b96c>] (snd_pcm_action_single+0x40/0x7c [snd_pcm]) r4:bf1acd74 [<bf19b92c>] (snd_pcm_action_single+0x0/0x7c [snd_pcm]) from [<bf19c774>] (snd_pcm_action_nonatomic+0x58/0x70 [snd_pcm]) r7:00044930 r6:bf1acd74 r5:00000802 r4:c13d02c0 [<bf19c71c>] (snd_pcm_action_nonatomic+0x0/0x70 [snd_pcm]) from [<bf19eba8>] (snd_pcm_common_ioctl1+0x664/0xf7c [snd_pcm]) r6:c13d02c0 r5:00044930 r4:00044930 [<bf19e544>] (snd_pcm_common_ioctl1+0x0/0xf7c [snd_pcm]) from [<bf19fe18>] (snd_pcm_playback_ioctl1+0x48c/0x4b0 [snd_pcm]) [<bf19f98c>] (snd_pcm_playback_ioctl1+0x0/0x4b0 [snd_pcm]) from [<bf19ff20>] (snd_pcm_playback_ioctl+0x34/0x40 [snd_pcm]) [<bf19feec>] (snd_pcm_playback_ioctl+0x0/0x40 [snd_pcm]) from [<c0097e70>] (do_ioctl+0x34/0x78) [<c0097e3c>] (do_ioctl+0x0/0x78) from [<c0098114>] (vfs_ioctl+0x260/0x290) r5:00044930 r4:c0fbcbc0 [<c0097eb4>] (vfs_ioctl+0x0/0x290) from [<c009819c>] (sys_ioctl+0x58/0x78) r7:00004140 r6:c0fbcbc0 r5:00044930 r4:00000000 [<c0098144>] (sys_ioctl+0x0/0x78) from [<c0022de0>] (ret_fast_syscall+0x0/0x2c) r8:c0022f84 r7:00000036 r6:beebb3a0 r5:beebb708 r4:00043598 Can anyone point me what goes wrong here? Kernel is debian 2.6.24-1-ixp4xx (armv5tel). Thanks, Marco

2 1

Re: [alsa-devel] [PATCH] ALSA: wss_lib: fix AZT2320 probe.
by krzysztof.h1＠poczta.fm 05 Aug '08

05 Aug '08

> Hi Takashi > > The attached is needed and sufficient to get AZT2320 back in line. > Trouble is with non-settable MIC gain bits on AZT2320 which fools > snd_ad1848_probe() and makes snd-azt2320 fail to load after the wss_lib > merge. > > Krzysztof should ACK/NAK this -- he was busy there with CS4248 and might > have other plans for it. > > Krzysztof, do you? I'm rather itching to rewrite that function > otherwise. It's horrible. > Acked-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> If you want to rewrite this function go for it. I'll try to solve cs4248 misdetection and forward you all details (or make a patch after your rewrite). Kind regards, Krzysztof ---------------------------------------------------------------------- Te newsy nakreca Cie na caly dzien! Sprawdz >>> http://link.interia.pl/f1e94

2 1

[alsa-devel] [PATCH] snd_usb_audio: fix SB Extigy IR Remote regression
by Phil Jordan 05 Aug '08

05 Aug '08

From: Phillip Michael Jordan <phil(a)philjordan.eu> The support for the SB Extigy's remote seems to be broken in all recent ALSA versions, including 1.0.17. The driver detects the event correctly, then submits a URB to query the RC code. On the Extigy, the URB is submitted with a length of 2 bytes. My hardware, however, only replies with 1 byte, containing the correct RC button code. The current implementation discards this as being too short. (line 1784 of usbmixer.c) This patch specifies a "minimum packet length" in the remote control configuration. I've left the values for the Audigy 2/Live! the same as the packet length, as I'm assuming the existing code works with them. (I don't have the hardware to confirm) This fixes the Extigy RC support, e.g. for use with Lirc. Signed-off-by: Phillip Michael Jordan <phil(a)philjordan.eu> --- An alternative, simpler fix would be to change the check on line 1784 to if (urb->status < 0 || urb->actual_length < rc->offset + rc->length) On the Extigy, this works out the same as the fix proposed in the patch (urb->actual_length < 1), but I have no idea how this would behave with the other types of hardware. Comments & suggestions welcome! phil --- a/sound/usb/usbmixer.c 2008-08-05 10:30:12.729535006 +0200 +++ b/sound/usb/usbmixer.c 2008-08-05 10:29:23.497533626 +0200 @@ -59,12 +59,13 @@ static const struct rc_config { u8 offset; u8 length; u8 packet_length; + u8 min_packet_length; /* minimum accepted length of the URB result */ u8 mute_mixer_id; u32 mute_code; } rc_configs[] = { - { USB_ID(0x041e, 0x3000), 0, 1, 2, 18, 0x0013 }, /* Extigy */ - { USB_ID(0x041e, 0x3020), 2, 1, 6, 18, 0x0013 }, /* Audigy 2 NX */ - { USB_ID(0x041e, 0x3040), 2, 2, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ + { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ + { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ }; struct usb_mixer_interface { @@ -1781,7 +1782,7 @@ static void snd_usb_soundblaster_remote_ const struct rc_config *rc = mixer->rc_cfg; u32 code; - if (urb->status < 0 || urb->actual_length < rc->packet_length) + if (urb->status < 0 || urb->actual_length < rc->min_packet_length) return; code = mixer->rc_buffer[rc->offset];

2 1

[alsa-devel] [PATCH] ALSA: wss_lib: fix AZT2320 probe.
by Rene Herman 05 Aug '08

05 Aug '08

Hi Takashi The attached is needed and sufficient to get AZT2320 back in line. Trouble is with non-settable MIC gain bits on AZT2320 which fools snd_ad1848_probe() and makes snd-azt2320 fail to load after the wss_lib merge. Krzysztof should ACK/NAK this -- he was busy there with CS4248 and might have other plans for it. Krzysztof, do you? I'm rather itching to rewrite that function otherwise. It's horrible. Rene.

1 0

[PATCH] ALSA: wss_lib: fix AZT2320 probe.
by Rene Herman 05 Aug '08

05 Aug '08

After the transition from cs4321_lib to wss_lib, azt2320 probe visits snd_ad1848_probe during detection. It expects register 0 (LEFT_INPUT) to be able to retain the value 0xaa during detection but AZT2320 does not support MIC gain meaning it reads back as 0x8a instead. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/wss/wss_lib.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index f2df77b..955db45 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1168,11 +1168,13 @@ static int snd_ad1848_probe(struct snd_wss *chip) ad1847 = 1; break; } - if (snd_wss_in(chip, CS4231_LEFT_INPUT) == 0xaa && - rev == 0x45) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - break; + if (rev == 0x45) { + rev = snd_wss_in(chip, CS4231_LEFT_INPUT); + if (rev == 0xaa || rev == 0x8a) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + id = 1; + break; + } } spin_unlock_irqrestore(&chip->reg_lock, flags); } -- 1.5.5 --------------010905020004010504060906--

1 0

Re: [alsa-devel] [PATCH 03/11] wss_lib: rename cs4321_foo to wss_foo
by krzysztof.h1＠poczta.fm 05 Aug '08

05 Aug '08

> On 31-07-08 21:03, Krzysztof Helt wrote: > > Takashi, you already applied these but is it possible to still get my > Acked-by (or Reviewed-by, whatever you want) on them? When a problem is > bisected to one of these, people tend to put the people in the patch > tags in CC and I looked at them and would like to be informed of any > trouble. I tested most users by now, and we'll get the azt2320 fixed > before this hits the streets but I still like to be in the loop on any > traffic concerning them. > > Acked-by: Rene Herman <rene.herman(a)gmail.com> > > (or Reviewed-by, Tested-by, ...) > > Below comments not important: > > > @@ -679,23 +679,25 @@ static int __devinit snd_opl3sa2_probe(s > > [ ... ] > > > - chip->cs4231 = cs4231; > > - if ((err = snd_cs4231_pcm(cs4231, 0, NULL)) < 0) > > + chip->wss = wss; > > + err = snd_wss_pcm(wss, 0, NULL); > > + if (err < 0) > > return err; > > - if ((err = snd_cs4231_mixer(cs4231)) < 0) > > + err = snd_wss_mixer(wss); > > + if (err < 0) > > return err; > > if ((err = snd_opl3sa2_mixer(chip)) < 0) > > return err; > > <mumble> > I think about doing some code improvements in the opl3sa2 e.g. use WSS tlv macros instead of the tlv macros defined localy. Regards, Krzysztof ---------------------------------------------------------------------- Tani Telefon! Sprawdz >>> http://link.interia.pl/f1ec0

2 1

Re: [alsa-devel] [PATCH 05/11] wss_lib: use wss constants instead of ad1848 ones
by krzysztof.h1＠poczta.fm 05 Aug '08

05 Aug '08

> On 31-07-08 21:05, Krzysztof Helt wrote: > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Use wss constants for mode. > > Move ad1848 hardware constants to the wss.h. > > Move mixer tlv macros into the ad1848_lib.c from the ad1848.h. > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > --- > > > > Changes since previous version: > > > > 1. No reformatting of switch in the ad1848_lib.c > > in the snd_ad1848_chip_id() as the file is going > > to be killed anyway. > > Still a little uneasy about the loss of MODE_RUNNING given that the > original looked as something noone would've added without a specific > need but I guess we can keep it in mind and match it up with any > possible regressions. > > I _would_ very much like a comment in the changelog about it. It's > unlikely to be a regression that would be reported quickly with this > being old hardware generally. Something like: > > "Drop the MODE_RUNNING spurious IRQ guard on AD1848 as it doesn not seem > to be needed." > Arghhh. I forgot this. You have mentioned it previously. Takashi, is it possible to add something to patch's description? > Otherwise: > > Acked-by: Rene Herman <rene.herman(a)gmail.com> > > Rene. > > Regards, Krzysztof ---------------------------------------------------------------------- Tani Telefon! Sprawdz >>> http://link.interia.pl/f1ec0

1 0

[alsa-devel] [PATCH 11/11] wss_lib: fix opti93x capture formats limitations
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Limit opti93x cards capture formats to only linear ones. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- No changes since the last version. diff -urp linux-ref/sound/isa/wss/wss_lib.c linux-2.6.26/sound/isa/wss/wss_lib.c --- linux-ref/sound/isa/wss/wss_lib.c 2008-07-31 19:30:47.000000000 +0200 +++ linux-2.6.26/sound/isa/wss/wss_lib.c 2008-07-31 19:30:56.000000000 +0200 @@ -1495,8 +1495,10 @@ static int snd_wss_capture_open(struct s /* hardware limitation of cheap chips */ if (chip->hardware == WSS_HW_CS4235 || - chip->hardware == WSS_HW_CS4239) - runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; + chip->hardware == WSS_HW_CS4239 || + chip->hardware == WSS_HW_OPTI93X) + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE; snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); ---------------------------------------------------------------------- Tanie rozmowy! Sprawdz >>> http://link.interia.pl/f1e91

3 2

[alsa-devel] [PATCH 10/11] wss_lib: use wss detection code instead of ad1848 one
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss detection code and kill the ad1848 library. The library is fully assimilated into the new wss library. This required reworking of the AD1848 family code so the code is changed to correctly detect chips from the AD1848 and CS4231 families. I have tested it on following cards: Gallant SC-6600 (codec: AD1848, driver: snd-sc6600) SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape) SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy) Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x) Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x) Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236) Audio Plus 3D (chip/codec: CMI8330A, driver: snd-cmi8330) Dell Latitude CP (chip/codec: cs4236, driver snd-cs4232) Sound playback and recording works on all these cards. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Changes since previous version: 1. Remove AD1848_LIB symbol from the Kconfig. 2. Actually delete ad1848.h and ad1848_lib.c. 3. Leave different description for opti82x-ad1848 and opti 92x-cs4231 drivers (they are still separated drivers). 4. One less assigment in if. diff -urpN linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 1970-01-01 01:00:00.000000000 +0100 @@ -1,111 +0,0 @@ -#ifndef __SOUND_AD1848_H -#define __SOUND_AD1848_H - -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Definitions for AD1847/AD1848/CS4248 chips - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include "pcm.h" -#include <linux/interrupt.h> - -#include "wss.h" /* temporary till the driver is removed */ - -/* codec registers */ - -#define AD1848_LEFT_INPUT 0x00 /* left input control */ -#define AD1848_RIGHT_INPUT 0x01 /* right input control */ -#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ -#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ -#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ -#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define AD1848_PIN_CTRL 0x0a /* pin control */ -#define AD1848_TEST_INIT 0x0b /* test and initialization */ -#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ -#define AD1848_LOOPBACK 0x0d /* loopback control */ -#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ -#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define AD1848_INIT 0x80 /* CODEC is initializing */ -#define AD1848_MCE 0x40 /* mode change enable */ -#define AD1848_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ - -#define AD1848_ENABLE_MIC_GAIN 0x20 - -#define AD1848_MIXS_LINE1 0x00 -#define AD1848_MIXS_AUX1 0x40 -#define AD1848_MIXS_LINE2 0x80 -#define AD1848_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ - -#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ -#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ -#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define AD1848_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define AD1848_XTAL1 0x00 /* 24.576 crystal */ -#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - AD1848_IFACE_CTRL */ - -#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ -#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ -#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ -#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ -#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ -#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - AD1848_PIN_CTRL */ - -#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ -#define AD1848_XCTL1 0x40 /* external control #1 */ -#define AD1848_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - AD1848_TEST_INIT */ - -#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ - -/* exported functions */ - -void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, - unsigned char value); - -int snd_ad1848_create(struct snd_card *card, - unsigned long port, - int irq, int dma, - unsigned short hardware, - struct snd_wss **chip); - -#endif /* __SOUND_AD1848_H */ diff -urpN linux-ref/sound/isa/Kconfig linux-2.6.26/sound/isa/Kconfig --- linux-ref/sound/isa/Kconfig 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/Kconfig 2008-07-31 19:22:16.000000000 +0200 @@ -4,11 +4,6 @@ config SND_WSS_LIB tristate select SND_PCM -config SND_AD1848_LIB - tristate - select SND_PCM - select SND_WSS_LIB - config SND_SB_COMMON tristate @@ -56,7 +51,7 @@ config SND_AD1816A config SND_AD1848 tristate "Generic AD1848/CS4248 driver" - select SND_AD1848_LIB + select SND_WSS_LIB help Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. @@ -97,7 +92,7 @@ config SND_AZT2320 config SND_CMI8330 tristate "C-Media CMI8330" - select SND_AD1848_LIB + select SND_WSS_LIB select SND_SB16_DSP help Say Y here to include support for soundcards based on the @@ -193,7 +188,7 @@ config SND_ES18XX config SND_SC6000 tristate "Gallant SC-6000, Audio Excel DSP 16" depends on HAS_IOPORT - select SND_AD1848_LIB + select SND_WSS_LIB select SND_OPL3_LIB select SND_MPU401_UART help @@ -280,7 +275,7 @@ config SND_OPTI92X_AD1848 select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART - select SND_AD1848_LIB + select SND_WSS_LIB help Say Y here to include support for soundcards based on Opti 82C92x or OTI-601 chips and using an AD1848 codec. @@ -373,7 +368,7 @@ config SND_SB16_CSP config SND_SGALAXY tristate "Aztech Sound Galaxy" - select SND_AD1848_LIB + select SND_WSS_LIB help Say Y here to include support for Aztech Sound Galaxy soundcards. diff -urpN linux-ref/sound/isa/ad1848/Makefile linux-2.6.26/sound/isa/ad1848/Makefile --- linux-ref/sound/isa/ad1848/Makefile 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/Makefile 2008-07-31 19:17:44.000000000 +0200 @@ -3,10 +3,8 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex(a)perex.cz> # -snd-ad1848-lib-objs := ad1848_lib.o snd-ad1848-objs := ad1848.o # Toplevel Module Dependency obj-$(CONFIG_SND_AD1848) += snd-ad1848.o -obj-$(CONFIG_SND_AD1848_LIB) += snd-ad1848-lib.o diff -urpN linux-ref/sound/isa/ad1848/ad1848.c linux-2.6.26/sound/isa/ad1848/ad1848.c --- linux-ref/sound/isa/ad1848/ad1848.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848.c 2008-07-31 19:17:44.000000000 +0200 @@ -28,7 +28,7 @@ #include <linux/wait.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/initval.h> #define CRD_NAME "Generic AD1848/AD1847/CS4248" @@ -95,8 +95,9 @@ static int __devinit snd_ad1848_probe(st if (!card) return -EINVAL; - error = snd_ad1848_create(card, port[n], irq[n], dma1[n], - thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, &chip); + error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], -1, + thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, + 0, &chip); if (error < 0) goto out; diff -urpN linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 1970-01-01 01:00:00.000000000 +0100 @@ -1,487 +0,0 @@ -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Routines for control of AD1848/AD1847/CS4248 - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#define SNDRV_MAIN_OBJECT_FILE -#include <linux/delay.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/slab.h> -#include <linux/ioport.h> -#include <sound/core.h> -#include <sound/ad1848.h> -#include <sound/control.h> -#include <sound/tlv.h> -#include <sound/pcm_params.h> - -#include <asm/io.h> -#include <asm/dma.h> - -MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); -MODULE_DESCRIPTION("Routines for control of AD1848/AD1847/CS4248"); -MODULE_LICENSE("GPL"); - -#if 0 -#define SNDRV_DEBUG_MCE -#endif - -/* - * Some variables - */ - -static unsigned char snd_ad1848_original_image[16] = -{ - 0x00, /* 00 - lic */ - 0x00, /* 01 - ric */ - 0x9f, /* 02 - la1ic */ - 0x9f, /* 03 - ra1ic */ - 0x9f, /* 04 - la2ic */ - 0x9f, /* 05 - ra2ic */ - 0xbf, /* 06 - loc */ - 0xbf, /* 07 - roc */ - 0x20, /* 08 - dfr */ - AD1848_AUTOCALIB, /* 09 - ic */ - 0x00, /* 0a - pc */ - 0x00, /* 0b - ti */ - 0x00, /* 0c - mi */ - 0x00, /* 0d - lbc */ - 0x00, /* 0e - dru */ - 0x00, /* 0f - drl */ -}; - -/* - * Basic I/O functions - */ - -static void snd_ad1848_wait(struct snd_wss *chip) -{ - int timeout; - - for (timeout = 250; timeout > 0; timeout--) { - if ((inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) == 0) - break; - udelay(100); - } -} - -void snd_ad1848_out(struct snd_wss *chip, - unsigned char reg, - unsigned char value) -{ - snd_ad1848_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - " - "reg = 0x%x, value = 0x%x\n", reg, value); -#endif - outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); - outb(chip->image[reg] = value, chip->port + CS4231P(REG)); - mb(); - snd_printdd("codec out - reg 0x%x = 0x%x\n", - chip->mce_bit | reg, value); -} - -EXPORT_SYMBOL(snd_ad1848_out); - -static unsigned char snd_ad1848_in(struct snd_wss *chip, unsigned char reg) -{ - snd_ad1848_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - " - "reg = 0x%x\n", reg); -#endif - outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); - mb(); - return inb(chip->port + CS4231P(REG)); -} - -#if 0 - -static void snd_ad1848_debug(struct snd_wss *chip) -{ - printk(KERN_DEBUG "AD1848 REGS: INDEX = 0x%02x ", inb(chip->port + CS4231P(REGSEL))); - printk(KERN_DEBUG " STATUS = 0x%02x\n", inb(chip->port + CS4231P(STATUS))); - printk(KERN_DEBUG " 0x00: left input = 0x%02x ", snd_ad1848_in(chip, 0x00)); - printk(KERN_DEBUG " 0x08: playback format = 0x%02x\n", snd_ad1848_in(chip, 0x08)); - printk(KERN_DEBUG " 0x01: right input = 0x%02x ", snd_ad1848_in(chip, 0x01)); - printk(KERN_DEBUG " 0x09: iface (CFIG 1) = 0x%02x\n", snd_ad1848_in(chip, 0x09)); - printk(KERN_DEBUG " 0x02: AUXA left = 0x%02x ", snd_ad1848_in(chip, 0x02)); - printk(KERN_DEBUG " 0x0a: pin control = 0x%02x\n", snd_ad1848_in(chip, 0x0a)); - printk(KERN_DEBUG " 0x03: AUXA right = 0x%02x ", snd_ad1848_in(chip, 0x03)); - printk(KERN_DEBUG " 0x0b: init & status = 0x%02x\n", snd_ad1848_in(chip, 0x0b)); - printk(KERN_DEBUG " 0x04: AUXB left = 0x%02x ", snd_ad1848_in(chip, 0x04)); - printk(KERN_DEBUG " 0x0c: revision & mode = 0x%02x\n", snd_ad1848_in(chip, 0x0c)); - printk(KERN_DEBUG " 0x05: AUXB right = 0x%02x ", snd_ad1848_in(chip, 0x05)); - printk(KERN_DEBUG " 0x0d: loopback = 0x%02x\n", snd_ad1848_in(chip, 0x0d)); - printk(KERN_DEBUG " 0x06: left output = 0x%02x ", snd_ad1848_in(chip, 0x06)); - printk(KERN_DEBUG " 0x0e: data upr count = 0x%02x\n", snd_ad1848_in(chip, 0x0e)); - printk(KERN_DEBUG " 0x07: right output = 0x%02x ", snd_ad1848_in(chip, 0x07)); - printk(KERN_DEBUG " 0x0f: data lwr count = 0x%02x\n", snd_ad1848_in(chip, 0x0f)); -} - -#endif - -/* - * AD1848 detection / MCE routines - */ - -static void snd_ad1848_mce_up(struct snd_wss *chip) -{ - unsigned long flags; - int timeout; - - snd_ad1848_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); -#endif - spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit |= AD1848_MCE; - timeout = inb(chip->port + CS4231P(REGSEL)); - if (timeout == 0x80) - snd_printk(KERN_WARNING "mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); - if (!(timeout & AD1848_MCE)) - outb(chip->mce_bit | (timeout & 0x1f), - chip->port + CS4231P(REGSEL)); - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static void snd_ad1848_mce_down(struct snd_wss *chip) -{ - unsigned long flags, timeout; - int reg; - - spin_lock_irqsave(&chip->reg_lock, flags); - for (timeout = 5; timeout > 0; timeout--) - inb(chip->port + CS4231P(REGSEL)); - /* end of cleanup sequence */ - for (timeout = 12000; - timeout > 0 && (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT); - timeout--) - udelay(100); - - snd_printdd("(1) timeout = %ld\n", timeout); - -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING - "mce_down [0x%lx] - auto calibration time out (0)\n", - chip->port + CS4231P(REGSEL)); -#endif - - chip->mce_bit &= ~AD1848_MCE; - reg = inb(chip->port + CS4231P(REGSEL)); - outb(chip->mce_bit | (reg & 0x1f), chip->port + CS4231P(REGSEL)); - if (reg == 0x80) - snd_printk(KERN_WARNING "mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((reg & AD1848_MCE) == 0) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } - - /* - * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low. - * It may take up to 5 sample periods (at most 907 us @ 5.5125 kHz) for - * the process to _start_, so it is important to wait at least that long - * before checking. Otherwise we might think AC has finished when it - * has in fact not begun. It could take 128 (no AC) or 384 (AC) cycles - * for ACI to drop. This gives a wait of at most 70 ms with a more - * typical value of 3-9 ms. - */ - timeout = jiffies + msecs_to_jiffies(250); - do { - spin_unlock_irqrestore(&chip->reg_lock, flags); - msleep(1); - spin_lock_irqsave(&chip->reg_lock, flags); - reg = snd_ad1848_in(chip, AD1848_TEST_INIT) & - AD1848_CALIB_IN_PROGRESS; - } while (reg && time_before(jiffies, timeout)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (reg) - snd_printk(KERN_ERR - "mce_down - auto calibration time out (2)\n"); - - snd_printdd("(4) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", - inb(chip->port + CS4231P(REGSEL))); -} - -static irqreturn_t snd_ad1848_interrupt(int irq, void *dev_id) -{ - struct snd_wss *chip = dev_id; - - if ((chip->mode & WSS_MODE_PLAY) && chip->playback_substream) - snd_pcm_period_elapsed(chip->playback_substream); - if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) - snd_pcm_period_elapsed(chip->capture_substream); - outb(0, chip->port + CS4231P(STATUS)); /* clear global interrupt bit */ - return IRQ_HANDLED; -} - -/* - - */ - -static void snd_ad1848_thinkpad_twiddle(struct snd_wss *chip, int on) -{ - int tmp; - - if (!chip->thinkpad_flag) return; - - outb(0x1c, AD1848_THINKPAD_CTL_PORT1); - tmp = inb(AD1848_THINKPAD_CTL_PORT2); - - if (on) - /* turn it on */ - tmp |= AD1848_THINKPAD_CS4248_ENABLE_BIT; - else - /* turn it off */ - tmp &= ~AD1848_THINKPAD_CS4248_ENABLE_BIT; - - outb(tmp, AD1848_THINKPAD_CTL_PORT2); - -} - -#ifdef CONFIG_PM -static void snd_ad1848_suspend(struct snd_wss *chip) -{ - snd_pcm_suspend_all(chip->pcm); - if (chip->thinkpad_flag) - snd_ad1848_thinkpad_twiddle(chip, 0); -} - -static void snd_ad1848_resume(struct snd_wss *chip) -{ - int i; - - if (chip->thinkpad_flag) - snd_ad1848_thinkpad_twiddle(chip, 1); - - /* clear any pendings IRQ */ - inb(chip->port + CS4231P(STATUS)); - outb(0, chip->port + CS4231P(STATUS)); - mb(); - - snd_ad1848_mce_down(chip); - for (i = 0; i < 16; i++) - snd_ad1848_out(chip, i, chip->image[i]); - snd_ad1848_mce_up(chip); - snd_ad1848_mce_down(chip); -} -#endif /* CONFIG_PM */ - -static int snd_ad1848_probe(struct snd_wss *chip) -{ - unsigned long flags; - int i, id, rev, ad1847; - unsigned char *ptr; - -#if 0 - snd_ad1848_debug(chip); -#endif - id = ad1847 = 0; - for (i = 0; i < 1000; i++) { - mb(); - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - udelay(500); - else { - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_MISC_INFO, 0x00); - snd_ad1848_out(chip, AD1848_LEFT_INPUT, 0xaa); - snd_ad1848_out(chip, AD1848_RIGHT_INPUT, 0x45); - rev = snd_ad1848_in(chip, AD1848_RIGHT_INPUT); - if (rev == 0x65) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - ad1847 = 1; - break; - } - if (snd_ad1848_in(chip, AD1848_LEFT_INPUT) == 0xaa && rev == 0x45) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - break; - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - } - } - if (id != 1) - return -ENODEV; /* no valid device found */ - if (chip->hardware == WSS_HW_DETECT) { - if (ad1847) { - chip->hardware = WSS_HW_AD1847; - } else { - chip->hardware = WSS_HW_AD1848; - rev = snd_ad1848_in(chip, AD1848_MISC_INFO); - if (rev & 0x80) { - chip->hardware = WSS_HW_CS4248; - } else if ((rev & 0x0f) == 0x0a) { - snd_ad1848_out(chip, AD1848_MISC_INFO, 0x40); - for (i = 0; i < 16; ++i) { - if (snd_ad1848_in(chip, i) != snd_ad1848_in(chip, i + 16)) { - chip->hardware = WSS_HW_CMI8330; - break; - } - } - snd_ad1848_out(chip, AD1848_MISC_INFO, 0x00); - } - } - } - spin_lock_irqsave(&chip->reg_lock, flags); - inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ - outb(0, chip->port + CS4231P(STATUS)); - mb(); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->image[AD1848_MISC_INFO] = 0x00; - chip->image[AD1848_IFACE_CTRL] = - (chip->image[AD1848_IFACE_CTRL] & ~AD1848_SINGLE_DMA) | AD1848_SINGLE_DMA; - ptr = (unsigned char *) &chip->image; - snd_ad1848_mce_down(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - for (i = 0; i < 16; i++) /* ok.. fill all AD1848 registers */ - snd_ad1848_out(chip, i, *ptr++); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_up(chip); - /* init needed for WSS pcm */ - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | - AD1848_PLAYBACK_PIO | - AD1848_CAPTURE_ENABLE | - AD1848_CAPTURE_PIO | - AD1848_CALIB_MODE); - chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - return 0; /* all things are ok.. */ -} - -/* - - */ - -static int snd_ad1848_free(struct snd_wss *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->irq >= 0) - free_irq(chip->irq, (void *) chip); - if (chip->dma1 >= 0) { - snd_dma_disable(chip->dma1); - free_dma(chip->dma1); - } - kfree(chip); - return 0; -} - -static int snd_ad1848_dev_free(struct snd_device *device) -{ - struct snd_wss *chip = device->device_data; - return snd_ad1848_free(chip); -} - -int snd_ad1848_create(struct snd_card *card, - unsigned long port, - int irq, int dma, - unsigned short hardware, - struct snd_wss **rchip) -{ - static struct snd_device_ops ops = { - .dev_free = snd_ad1848_dev_free, - }; - struct snd_wss *chip; - int err; - - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - spin_lock_init(&chip->reg_lock); - chip->card = card; - chip->port = port; - chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->single_dma = 1; - chip->hardware = hardware; - memcpy(&chip->image, &snd_ad1848_original_image, sizeof(snd_ad1848_original_image)); - - if ((chip->res_port = request_region(port, 4, "AD1848")) == NULL) { - snd_printk(KERN_ERR "ad1848: can't grab port 0x%lx\n", port); - snd_ad1848_free(chip); - return -EBUSY; - } - if (request_irq(irq, snd_ad1848_interrupt, IRQF_DISABLED, "AD1848", (void *) chip)) { - snd_printk(KERN_ERR "ad1848: can't grab IRQ %d\n", irq); - snd_ad1848_free(chip); - return -EBUSY; - } - chip->irq = irq; - if (request_dma(dma, "AD1848")) { - snd_printk(KERN_ERR "ad1848: can't grab DMA %d\n", dma); - snd_ad1848_free(chip); - return -EBUSY; - } - chip->dma1 = dma; - chip->dma2 = dma; - - if (hardware == WSS_HW_THINKPAD) { - chip->thinkpad_flag = 1; - chip->hardware = WSS_HW_DETECT; /* reset */ - snd_ad1848_thinkpad_twiddle(chip, 1); - } - - if (snd_ad1848_probe(chip) < 0) { - snd_ad1848_free(chip); - return -ENODEV; - } - - /* Register device */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_ad1848_free(chip); - return err; - } - -#ifdef CONFIG_PM - chip->suspend = snd_ad1848_suspend; - chip->resume = snd_ad1848_resume; -#endif - - *rchip = chip; - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_create); - -/* - * INIT part - */ - -static int __init alsa_ad1848_init(void) -{ - return 0; -} - -static void __exit alsa_ad1848_exit(void) -{ -} - -module_init(alsa_ad1848_init) -module_exit(alsa_ad1848_exit) diff -urpN linux-ref/sound/isa/cmi8330.c linux-2.6.26/sound/isa/cmi8330.c --- linux-ref/sound/isa/cmi8330.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/cmi8330.c 2008-07-31 19:19:10.000000000 +0200 @@ -50,7 +50,7 @@ #include <linux/pnp.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/sb.h> #include <sound/initval.h> @@ -180,9 +180,9 @@ WSS_DOUBLE("Master Playback Volume", 0, WSS_SINGLE("Loud Playback Switch", 0, CMI8330_MUTEMUX, 6, 1, 1), WSS_DOUBLE("PCM Playback Switch", 0, - AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Playback Volume", 0, - AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), WSS_DOUBLE("Line Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, 4, 3, 1, 0), WSS_DOUBLE("Line Playback Volume", 0, @@ -489,12 +489,11 @@ static int __devinit snd_cmi8330_probe(s int i, err; acard = card->private_data; - if ((err = snd_ad1848_create(card, - wssport[dev] + 4, - wssirq[dev], - wssdma[dev], - WSS_HW_DETECT, - &acard->wss)) < 0) { + err = snd_wss_create(card, wssport[dev] + 4, -1, + wssirq[dev], + wssdma[dev], -1, + WSS_HW_DETECT, 0, &acard->wss); + if (err < 0) { snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } @@ -517,9 +516,10 @@ static int __devinit snd_cmi8330_probe(s return err; } - snd_ad1848_out(acard->wss, AD1848_MISC_INFO, 0x40); /* switch on MODE2 */ + snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */ for (i = CMI8330_RMUX3D; i <= CMI8330_CDINGAIN; i++) - snd_ad1848_out(acard->wss, i, snd_cmi8330_image[i - CMI8330_RMUX3D]); + snd_wss_out(acard->wss, i, + snd_cmi8330_image[i - CMI8330_RMUX3D]); if ((err = snd_cmi8330_mixer(card, acard)) < 0) { snd_printk(KERN_ERR PFX "failed to create mixers\n"); diff -urpN linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 19:27:21.000000000 +0200 @@ -33,11 +33,7 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#if defined(CS4231) || defined(OPTi93X) #include <sound/wss.h> -#else -#include <sound/ad1848.h> -#endif /* CS4231 */ #include <sound/mpu401.h> #include <sound/opl3.h> #ifndef OPTi93X @@ -148,9 +144,7 @@ struct snd_opti9xx { long wss_base; int irq; int dma1; -#if defined(CS4231) || defined(OPTi93X) int dma2; -#endif /* CS4231 || OPTi93X */ long fm_port; @@ -225,9 +219,7 @@ static int __devinit snd_opti9xx_init(st chip->wss_base = -1; chip->irq = -1; chip->dma1 = -1; -#if defined(CS4231) || defined (OPTi93X) chip->dma2 = -1; -#endif /* CS4231 || OPTi93X */ chip->fm_port = -1; chip->mpu_port = -1; chip->mpu_irq = -1; @@ -740,7 +732,6 @@ static int __devinit snd_opti9xx_probe(s if (error) return error; -#if defined(CS4231) || defined(OPTi93X) error = snd_wss_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, #ifdef CS4231 @@ -754,12 +745,6 @@ static int __devinit snd_opti9xx_probe(s #ifdef OPTi93X chip->codec = codec; #endif -#else - error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, - chip->dma1, WSS_HW_DETECT, &codec); - if (error < 0) - return error; -#endif error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; diff -urpN linux-ref/sound/isa/sc6000.c linux-2.6.26/sound/isa/sc6000.c --- linux-ref/sound/isa/sc6000.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/sc6000.c 2008-07-31 19:17:46.000000000 +0200 @@ -29,7 +29,7 @@ #include <linux/io.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/opl3.h> #include <sound/mpu401.h> #include <sound/control.h> @@ -548,8 +548,8 @@ static int __devinit snd_sc6000_probe(st if (err < 0) goto err_unmap2; - err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, - WSS_HW_DETECT, &chip); + err = snd_wss_create(card, mss_port[dev] + 4, -1, xirq, xdma, -1, + WSS_HW_DETECT, 0, &chip); if (err < 0) goto err_unmap2; card->private_data = chip; diff -urpN linux-ref/sound/isa/sgalaxy.c linux-2.6.26/sound/isa/sgalaxy.c --- linux-ref/sound/isa/sgalaxy.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/sgalaxy.c 2008-07-31 19:17:46.000000000 +0200 @@ -31,7 +31,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/sb.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/control.h> #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA @@ -267,9 +267,10 @@ static int __devinit snd_sgalaxy_probe(s if ((err = snd_sgalaxy_detect(dev, xirq, xdma1)) < 0) goto _err; - if ((err = snd_ad1848_create(card, wssport[dev] + 4, - xirq, xdma1, - WSS_HW_DETECT, &chip)) < 0) + err = snd_wss_create(card, wssport[dev] + 4, -1, + xirq, xdma1, -1, + WSS_HW_DETECT, 0, &chip); + if (err < 0) goto _err; card->private_data = chip; @@ -331,8 +332,8 @@ static int snd_sgalaxy_resume(struct dev struct snd_wss *chip = card->private_data; chip->resume(chip); - snd_ad1848_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); - snd_ad1848_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); + snd_wss_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); + snd_wss_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff -urpN linux-ref/sound/isa/wss/wss_lib.c linux-2.6.26/sound/isa/wss/wss_lib.c --- linux-ref/sound/isa/wss/wss_lib.c 2008-07-31 19:18:15.000000000 +0200 +++ linux-2.6.26/sound/isa/wss/wss_lib.c 2008-07-31 19:17:46.000000000 +0200 @@ -1073,7 +1073,11 @@ irqreturn_t snd_wss_interrupt(int irq, v struct snd_wss *chip = dev_id; unsigned char status; - status = snd_wss_in(chip, CS4231_IRQ_STATUS); + if (chip->hardware & WSS_HW_AD1848_MASK) + /* pretend it was the only possible irq for AD1848 */ + status = CS4231_PLAYBACK_IRQ; + else + status = snd_wss_in(chip, CS4231_IRQ_STATUS); if (status & CS4231_TIMER_IRQ) { if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); @@ -1105,7 +1109,11 @@ irqreturn_t snd_wss_interrupt(int irq, v } spin_lock(&chip->reg_lock); - snd_wss_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0); + status = ~CS4231_ALL_IRQS | ~status; + if (chip->hardware & WSS_HW_AD1848_MASK) + wss_outb(chip, CS4231P(STATUS), 0); + else + snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } @@ -1137,36 +1145,112 @@ static snd_pcm_uframes_t snd_wss_capture */ -static int snd_wss_probe(struct snd_wss *chip) +static int snd_ad1848_probe(struct snd_wss *chip) { unsigned long flags; - int i, id, rev; - unsigned char *ptr; - unsigned int hw; + int i, id, rev, ad1847; -#if 0 - snd_wss_debug(chip); -#endif id = 0; - for (i = 0; i < 50; i++) { + ad1847 = 0; + for (i = 0; i < 1000; i++) { mb(); - if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - udelay(2000); + if (inb(chip->port + CS4231P(REGSEL)) & CS4231_INIT) + msleep(1); else { spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - id = snd_wss_in(chip, CS4231_MISC_INFO) & 0x0f; + snd_wss_out(chip, CS4231_MISC_INFO, 0x00); + snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); + snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); + rev = snd_wss_in(chip, CS4231_RIGHT_INPUT); + if (rev == 0x65) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + id = 1; + ad1847 = 1; + break; + } + if (snd_wss_in(chip, CS4231_LEFT_INPUT) == 0xaa && + rev == 0x45) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + id = 1; + break; + } spin_unlock_irqrestore(&chip->reg_lock, flags); - if (id == 0x0a) - break; /* this is valid value */ } } - snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id); - if (id != 0x0a) + if (id != 1) return -ENODEV; /* no valid device found */ + id = 0; + if (chip->hardware == WSS_HW_DETECT) + id = ad1847 ? WSS_HW_AD1847 : WSS_HW_AD1848; + + spin_lock_irqsave(&chip->reg_lock, flags); + inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ + outb(0, chip->port + CS4231P(STATUS)); + mb(); + if (id == WSS_HW_AD1848) { + /* check if there are more than 16 registers */ + rev = snd_wss_in(chip, CS4231_MISC_INFO); + snd_wss_out(chip, CS4231_MISC_INFO, 0x40); + for (i = 0; i < 16; ++i) { + if (snd_wss_in(chip, i) != snd_wss_in(chip, i + 16)) { + id = WSS_HW_CMI8330; + break; + } + } + snd_wss_out(chip, CS4231_MISC_INFO, 0x00); + if (id != WSS_HW_CMI8330 && (rev & 0x80)) + id = WSS_HW_CS4248; + if (id == WSS_HW_CMI8330 && (rev & 0x0f) != 0x0a) + id = 0; + } + if (id == WSS_HW_CMI8330) { + /* verify it is not CS4231 by changing the version register */ + /* on CMI8330 it is volume control register and can be set 0 */ + snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + snd_wss_dout(chip, CS4231_VERSION, 0x00); + rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; + if (rev) + id = 0; + snd_wss_out(chip, CS4231_MISC_INFO, 0); + } + if (id) + chip->hardware = id; + + spin_unlock_irqrestore(&chip->reg_lock, flags); + return 0; /* all things are ok.. */ +} + +static int snd_wss_probe(struct snd_wss *chip) +{ + unsigned long flags; + int i, id, rev, regnum; + unsigned char *ptr; + unsigned int hw; + + id = snd_ad1848_probe(chip); + if (id < 0) + return id; hw = chip->hardware; if ((hw & WSS_HW_TYPE_MASK) == WSS_HW_DETECT) { + for (i = 0; i < 50; i++) { + mb(); + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + msleep(2); + else { + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_MISC_INFO, + CS4231_MODE2); + id = snd_wss_in(chip, CS4231_MISC_INFO) & 0x0f; + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (id == 0x0a) + break; /* this is valid value */ + } + } + snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id); + if (id != 0x0a) + return -ENODEV; /* no valid device found */ + rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev); if (rev == 0x80) { @@ -1197,7 +1281,8 @@ static int snd_wss_probe(struct snd_wss mb(); spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->image[CS4231_MISC_INFO] = CS4231_MODE2; + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + chip->image[CS4231_MISC_INFO] = CS4231_MODE2; switch (chip->hardware) { case WSS_HW_INTERWAVE: chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3; @@ -1223,9 +1308,10 @@ static int snd_wss_probe(struct snd_wss chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } ptr = (unsigned char *) &chip->image; + regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32; snd_wss_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); - for (i = 0; i < 32; i++) /* ok.. fill all CS4231 registers */ + for (i = 0; i < regnum; i++) /* ok.. fill all registers */ snd_wss_out(chip, i, *ptr++); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_up(chip); @@ -1635,6 +1721,10 @@ static int snd_wss_new(struct snd_card * else memcpy(&chip->image, &snd_wss_original_image, sizeof(snd_wss_original_image)); + if (chip->hardware & WSS_HW_AD1848_MASK) { + chip->image[CS4231_PIN_CTRL] = 0; + chip->image[CS4231_TEST_INIT] = 0; + } *rchip = chip; return 0; @@ -1662,7 +1752,7 @@ int snd_wss_create(struct snd_card *card chip->dma1 = -1; chip->dma2 = -1; - chip->res_port = request_region(port, 4, "CS4231"); + chip->res_port = request_region(port, 4, "WSS"); if (!chip->res_port) { snd_printk(KERN_ERR "wss: can't grab port 0x%lx\n", port); snd_wss_free(chip); @@ -1681,20 +1771,20 @@ int snd_wss_create(struct snd_card *card chip->cport = cport; if (!(hwshare & WSS_HWSHARE_IRQ)) if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED, - "CS4231", (void *) chip)) { + "WSS", (void *) chip)) { snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); snd_wss_free(chip); return -EBUSY; } chip->irq = irq; - if (!(hwshare & WSS_HWSHARE_DMA1) && request_dma(dma1, "CS4231 - 1")) { + if (!(hwshare & WSS_HWSHARE_DMA1) && request_dma(dma1, "WSS - 1")) { snd_printk(KERN_ERR "wss: can't grab DMA1 %d\n", dma1); snd_wss_free(chip); return -EBUSY; } chip->dma1 = dma1; if (!(hwshare & WSS_HWSHARE_DMA2) && dma1 != dma2 && - dma2 >= 0 && request_dma(dma2, "CS4231 - 2")) { + dma2 >= 0 && request_dma(dma2, "WSS - 2")) { snd_printk(KERN_ERR "wss: can't grab DMA2 %d\n", dma2); snd_wss_free(chip); return -EBUSY; @@ -1705,6 +1795,12 @@ int snd_wss_create(struct snd_card *card } else chip->dma2 = dma2; + if (hardware == WSS_HW_THINKPAD) { + chip->thinkpad_flag = 1; + chip->hardware = WSS_HW_DETECT; /* reset */ + snd_wss_thinkpad_twiddle(chip, 1); + } + /* global setup */ if (snd_wss_probe(chip) < 0) { snd_wss_free(chip); @@ -1775,12 +1871,6 @@ int snd_wss_pcm(struct snd_wss *chip, in snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); - /* temporary */ - if (chip->hardware & WSS_HW_AD1848_MASK) { - chip->rate_constraint = snd_wss_xrate; - chip->set_playback_format = snd_wss_playback_format; - chip->set_capture_format = snd_wss_capture_format; - } /* global setup */ pcm->private_data = chip; pcm->info_flags = 0; ---------------------------------------------------------------------- Partyjka w Chinczyka? Graj >>> http://link.interia.pl/f1e67

2 1

[alsa-devel] [PATCH 09/11] wss_lib: use wss pcm code instead of ad1848 one
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss pcm code and kill the ad1848 pcm code. The AD1848 chip is much slower than CS4231 chips so the waiting loop was increased 100x (10x is not enough). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Changes since previous version: 1. Condition in snd_wss_capture_open() fixed. 2. Few very small formatting improvements. diff -urp linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 2008-07-31 19:07:33.000000000 +0200 @@ -97,11 +97,6 @@ #define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ #define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ -/* IBM Thinkpad specific stuff */ -#define AD1848_THINKPAD_CTL_PORT1 0x15e8 -#define AD1848_THINKPAD_CTL_PORT2 0x15e9 -#define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02 - /* exported functions */ void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, @@ -113,7 +108,4 @@ int snd_ad1848_create(struct snd_card *c unsigned short hardware, struct snd_wss **chip); -int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); -const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); - #endif /* __SOUND_AD1848_H */ diff -urp linux-ref/include/sound/wss.h linux-2.6.26/include/sound/wss.h --- linux-ref/include/sound/wss.h 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/include/sound/wss.h 2008-07-31 19:07:33.000000000 +0200 @@ -71,6 +71,11 @@ #define WSS_HWSHARE_DMA1 (1<<1) #define WSS_HWSHARE_DMA2 (1<<2) +/* IBM Thinkpad specific stuff */ +#define AD1848_THINKPAD_CTL_PORT1 0x15e8 +#define AD1848_THINKPAD_CTL_PORT2 0x15e9 +#define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02 + struct snd_wss { unsigned long port; /* base i/o port */ struct resource *res_port; @@ -153,6 +158,8 @@ int snd_wss_pcm(struct snd_wss *chip, in int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer); int snd_wss_mixer(struct snd_wss *chip); +const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction); + int snd_cs4236_create(struct snd_card *card, unsigned long port, unsigned long cport, diff -urp linux-ref/sound/isa/Kconfig linux-2.6.26/sound/isa/Kconfig --- linux-ref/sound/isa/Kconfig 2008-07-31 17:28:06.000000000 +0200 +++ linux-2.6.26/sound/isa/Kconfig 2008-07-31 19:07:33.000000000 +0200 @@ -1,12 +1,13 @@ # ALSA ISA drivers -config SND_AD1848_LIB +config SND_WSS_LIB tristate select SND_PCM -config SND_WSS_LIB +config SND_AD1848_LIB tristate select SND_PCM + select SND_WSS_LIB config SND_SB_COMMON tristate diff -urp linux-ref/sound/isa/ad1848/ad1848.c linux-2.6.26/sound/isa/ad1848/ad1848.c --- linux-ref/sound/isa/ad1848/ad1848.c 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848.c 2008-07-31 19:07:33.000000000 +0200 @@ -102,7 +102,7 @@ static int __devinit snd_ad1848_probe(st card->private_data = chip; - error = snd_ad1848_pcm(chip, 0, &pcm); + error = snd_wss_pcm(chip, 0, &pcm); if (error < 0) goto out; diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 2008-07-31 19:08:10.000000000 +0200 @@ -46,34 +46,6 @@ MODULE_LICENSE("GPL"); * Some variables */ -static unsigned char freq_bits[14] = { - /* 5510 */ 0x00 | AD1848_XTAL2, - /* 6620 */ 0x0E | AD1848_XTAL2, - /* 8000 */ 0x00 | AD1848_XTAL1, - /* 9600 */ 0x0E | AD1848_XTAL1, - /* 11025 */ 0x02 | AD1848_XTAL2, - /* 16000 */ 0x02 | AD1848_XTAL1, - /* 18900 */ 0x04 | AD1848_XTAL2, - /* 22050 */ 0x06 | AD1848_XTAL2, - /* 27042 */ 0x04 | AD1848_XTAL1, - /* 32000 */ 0x06 | AD1848_XTAL1, - /* 33075 */ 0x0C | AD1848_XTAL2, - /* 37800 */ 0x08 | AD1848_XTAL2, - /* 44100 */ 0x0A | AD1848_XTAL2, - /* 48000 */ 0x0C | AD1848_XTAL1 -}; - -static unsigned int rates[14] = { - 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050, - 27042, 32000, 33075, 37800, 44100, 48000 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - static unsigned char snd_ad1848_original_image[16] = { 0x00, /* 00 - lic */ @@ -128,15 +100,6 @@ void snd_ad1848_out(struct snd_wss *chip EXPORT_SYMBOL(snd_ad1848_out); -static void snd_ad1848_dout(struct snd_wss *chip, - unsigned char reg, unsigned char value) -{ - snd_ad1848_wait(chip); - outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); - outb(value, chip->port + CS4231P(REG)); - mb(); -} - static unsigned char snd_ad1848_in(struct snd_wss *chip, unsigned char reg) { snd_ad1848_wait(chip); @@ -261,315 +224,6 @@ static void snd_ad1848_mce_down(struct s inb(chip->port + CS4231P(REGSEL))); } -static unsigned int snd_ad1848_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case AD1848_LINEAR_16: - size >>= 1; - break; - } - if (format & AD1848_STEREO) - size >>= 1; - return size; -} - -static int snd_ad1848_trigger(struct snd_wss *chip, unsigned char what, - int channel, int cmd) -{ - int result = 0; - -#if 0 - printk("codec trigger!!! - what = %i, enable = %i, status = 0x%x\n", what, enable, inb(AD1848P(card, STATUS))); -#endif - spin_lock(&chip->reg_lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - if (chip->image[AD1848_IFACE_CTRL] & what) { - spin_unlock(&chip->reg_lock); - return 0; - } - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] |= what); - } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { - if (!(chip->image[AD1848_IFACE_CTRL] & what)) { - spin_unlock(&chip->reg_lock); - return 0; - } - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] &= ~what); - } else { - result = -EINVAL; - } - spin_unlock(&chip->reg_lock); - return result; -} - -/* - * CODEC I/O - */ - -static unsigned char snd_ad1848_get_rate(unsigned int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(rates); i++) - if (rate == rates[i]) - return freq_bits[i]; - snd_BUG(); - return freq_bits[ARRAY_SIZE(rates) - 1]; -} - -static int snd_ad1848_ioctl(struct snd_pcm_substream *substream, - unsigned int cmd, void *arg) -{ - return snd_pcm_lib_ioctl(substream, cmd, arg); -} - -static unsigned char snd_ad1848_get_format(int format, int channels) -{ - unsigned char rformat; - - rformat = AD1848_LINEAR_8; - switch (format) { - case SNDRV_PCM_FORMAT_A_LAW: rformat = AD1848_ALAW_8; break; - case SNDRV_PCM_FORMAT_MU_LAW: rformat = AD1848_ULAW_8; break; - case SNDRV_PCM_FORMAT_S16_LE: rformat = AD1848_LINEAR_16; break; - } - if (channels > 1) - rformat |= AD1848_STEREO; -#if 0 - snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); -#endif - return rformat; -} - -static void snd_ad1848_calibrate_mute(struct snd_wss *chip, int mute) -{ - unsigned long flags; - - mute = mute ? 1 : 0; - spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->calibrate_mute == mute) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } - if (!mute) { - snd_ad1848_dout(chip, AD1848_LEFT_INPUT, chip->image[AD1848_LEFT_INPUT]); - snd_ad1848_dout(chip, AD1848_RIGHT_INPUT, chip->image[AD1848_RIGHT_INPUT]); - } - snd_ad1848_dout(chip, AD1848_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX1_LEFT_INPUT]); - snd_ad1848_dout(chip, AD1848_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX1_RIGHT_INPUT]); - snd_ad1848_dout(chip, AD1848_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX2_LEFT_INPUT]); - snd_ad1848_dout(chip, AD1848_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX2_RIGHT_INPUT]); - snd_ad1848_dout(chip, AD1848_LEFT_OUTPUT, mute ? 0x80 : chip->image[AD1848_LEFT_OUTPUT]); - snd_ad1848_dout(chip, AD1848_RIGHT_OUTPUT, mute ? 0x80 : chip->image[AD1848_RIGHT_OUTPUT]); - chip->calibrate_mute = mute; - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static void snd_ad1848_set_data_format(struct snd_wss *chip, - struct snd_pcm_hw_params *hw_params) -{ - if (hw_params == NULL) { - chip->image[AD1848_DATA_FORMAT] = 0x20; - } else { - chip->image[AD1848_DATA_FORMAT] = - snd_ad1848_get_format(params_format(hw_params), params_channels(hw_params)) | - snd_ad1848_get_rate(params_rate(hw_params)); - } - // snd_printk(">>> pmode = 0x%x, dfr = 0x%x\n", pstr->mode, chip->image[AD1848_DATA_FORMAT]); -} - -static int snd_ad1848_open(struct snd_wss *chip, unsigned int mode) -{ - unsigned long flags; - - if (chip->mode & WSS_MODE_OPEN) - return -EAGAIN; - - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (1)\n"); -#endif - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | - AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO | - AD1848_CALIB_MODE); - chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (2)\n"); -#endif - - snd_ad1848_set_data_format(chip, NULL); - - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (3)\n"); -#endif - - /* ok. now enable and ack CODEC IRQ */ - spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - chip->image[AD1848_PIN_CTRL] |= AD1848_IRQ_ENABLE; - snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->mode = mode; - - return 0; -} - -static void snd_ad1848_close(struct snd_wss *chip) -{ - unsigned long flags; - - if (!chip->mode) - return; - /* disable IRQ */ - spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - chip->image[AD1848_PIN_CTRL] &= ~AD1848_IRQ_ENABLE; - snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - /* now disable capture & playback */ - - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | - AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - - /* clear IRQ again */ - spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->mode = 0; -} - -/* - * ok.. exported functions.. - */ - -static int snd_ad1848_playback_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - return snd_ad1848_trigger(chip, AD1848_PLAYBACK_ENABLE, SNDRV_PCM_STREAM_PLAYBACK, cmd); -} - -static int snd_ad1848_capture_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - return snd_ad1848_trigger(chip, AD1848_CAPTURE_ENABLE, SNDRV_PCM_STREAM_CAPTURE, cmd); -} - -static int snd_ad1848_playback_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - unsigned long flags; - int err; - - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) - return err; - snd_ad1848_calibrate_mute(chip, 1); - snd_ad1848_set_data_format(chip, hw_params); - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - snd_ad1848_calibrate_mute(chip, 0); - return 0; -} - -static int snd_ad1848_playback_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int snd_ad1848_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); - - chip->p_dma_size = size; - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO); - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); - snd_ad1848_out(chip, AD1848_DATA_UPR_CNT, (unsigned char) (count >> 8)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static int snd_ad1848_capture_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - unsigned long flags; - int err; - - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) - return err; - snd_ad1848_calibrate_mute(chip, 1); - snd_ad1848_set_data_format(chip, hw_params); - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - snd_ad1848_calibrate_mute(chip, 0); - return 0; -} - -static int snd_ad1848_capture_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int snd_ad1848_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); - - chip->c_dma_size = size; - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); - snd_ad1848_out(chip, AD1848_DATA_UPR_CNT, (unsigned char) (count >> 8)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - static irqreturn_t snd_ad1848_interrupt(int irq, void *dev_id) { struct snd_wss *chip = dev_id; @@ -582,28 +236,6 @@ static irqreturn_t snd_ad1848_interrupt( return IRQ_HANDLED; } -static snd_pcm_uframes_t snd_ad1848_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) - return 0; - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_ad1848_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE)) - return 0; - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - /* */ @@ -728,6 +360,16 @@ static int snd_ad1848_probe(struct snd_w snd_ad1848_out(chip, i, *ptr++); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_ad1848_mce_up(chip); + /* init needed for WSS pcm */ + spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | + AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | + AD1848_CAPTURE_PIO | + AD1848_CALIB_MODE); + chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; + snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); + spin_unlock_irqrestore(&chip->reg_lock, flags); snd_ad1848_mce_down(chip); return 0; /* all things are ok.. */ } @@ -736,102 +378,6 @@ static int snd_ad1848_probe(struct snd_w */ -static struct snd_pcm_hardware snd_ad1848_playback = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5510, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_ad1848_capture = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5510, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -/* - - */ - -static int snd_ad1848_playback_open(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - err = snd_ad1848_open(chip, WSS_MODE_PLAY); - if (err < 0) - return err; - chip->playback_substream = substream; - runtime->hw = snd_ad1848_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return 0; -} - -static int snd_ad1848_capture_open(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - err = snd_ad1848_open(chip, WSS_MODE_RECORD); - if (err < 0) - return err; - chip->capture_substream = substream; - runtime->hw = snd_ad1848_capture; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return 0; -} - -static int snd_ad1848_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - - chip->mode &= ~WSS_MODE_PLAY; - chip->playback_substream = NULL; - snd_ad1848_close(chip); - return 0; -} - -static int snd_ad1848_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - - chip->mode &= ~WSS_MODE_RECORD; - chip->capture_substream = NULL; - snd_ad1848_close(chip); - return 0; -} - static int snd_ad1848_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); @@ -851,17 +397,6 @@ static int snd_ad1848_dev_free(struct sn return snd_ad1848_free(chip); } -static const char *snd_ad1848_chip_id(struct snd_wss *chip) -{ - switch (chip->hardware) { - case AD1848_HW_AD1847: return "AD1847"; - case AD1848_HW_AD1848: return "AD1848"; - case AD1848_HW_CS4248: return "CS4248"; - case AD1848_HW_CMI8330: return "CMI8330/C3D"; - default: return "???"; - } -} - int snd_ad1848_create(struct snd_card *card, unsigned long port, int irq, int dma, @@ -935,65 +470,6 @@ int snd_ad1848_create(struct snd_card *c EXPORT_SYMBOL(snd_ad1848_create); -static struct snd_pcm_ops snd_ad1848_playback_ops = { - .open = snd_ad1848_playback_open, - .close = snd_ad1848_playback_close, - .ioctl = snd_ad1848_ioctl, - .hw_params = snd_ad1848_playback_hw_params, - .hw_free = snd_ad1848_playback_hw_free, - .prepare = snd_ad1848_playback_prepare, - .trigger = snd_ad1848_playback_trigger, - .pointer = snd_ad1848_playback_pointer, -}; - -static struct snd_pcm_ops snd_ad1848_capture_ops = { - .open = snd_ad1848_capture_open, - .close = snd_ad1848_capture_close, - .ioctl = snd_ad1848_ioctl, - .hw_params = snd_ad1848_capture_hw_params, - .hw_free = snd_ad1848_capture_hw_free, - .prepare = snd_ad1848_capture_prepare, - .trigger = snd_ad1848_capture_trigger, - .pointer = snd_ad1848_capture_pointer, -}; - -int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(chip->card, "AD1848", device, 1, 1, &pcm)) < 0) - return err; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ad1848_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ad1848_capture_ops); - - pcm->private_data = chip; - pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - strcpy(pcm->name, snd_ad1848_chip_id(chip)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64 * 1024, - chip->dma1 > 3 ? - 128 * 1024 : 64 * 1024); - - chip->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_pcm); - -const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction) -{ - return direction == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_ad1848_playback_ops : &snd_ad1848_capture_ops; -} - -EXPORT_SYMBOL(snd_ad1848_get_pcm_ops); - /* * INIT part */ diff -urp linux-ref/sound/isa/cmi8330.c linux-2.6.26/sound/isa/cmi8330.c --- linux-ref/sound/isa/cmi8330.c 2008-07-31 18:59:20.000000000 +0200 +++ linux-2.6.26/sound/isa/cmi8330.c 2008-07-31 19:07:33.000000000 +0200 @@ -413,7 +413,7 @@ static int __devinit snd_cmi8330_pcm(str chip->streams[CMI_SB_STREAM].private_data = chip->sb; /* AD1848 */ - ops = snd_ad1848_get_pcm_ops(CMI_AD_STREAM); + ops = snd_wss_get_pcm_ops(CMI_AD_STREAM); chip->streams[CMI_AD_STREAM].ops = *ops; chip->streams[CMI_AD_STREAM].open = ops->open; chip->streams[CMI_AD_STREAM].ops.open = cmi_open_callbacks[CMI_AD_STREAM]; diff -urp linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 19:07:33.000000000 +0200 @@ -754,18 +754,15 @@ static int __devinit snd_opti9xx_probe(s #ifdef OPTi93X chip->codec = codec; #endif - error = snd_wss_pcm(codec, 0, &pcm); - if (error < 0) - return error; #else error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, WSS_HW_DETECT, &codec); if (error < 0) return error; - error = snd_ad1848_pcm(codec, 0, &pcm); +#endif + error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; -#endif error = snd_wss_mixer(codec); if (error < 0) return error; diff -urp linux-ref/sound/isa/sc6000.c linux-2.6.26/sound/isa/sc6000.c --- linux-ref/sound/isa/sc6000.c 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/sound/isa/sc6000.c 2008-07-31 19:07:33.000000000 +0200 @@ -554,10 +554,10 @@ static int __devinit snd_sc6000_probe(st goto err_unmap2; card->private_data = chip; - err = snd_ad1848_pcm(chip, 0, NULL); + err = snd_wss_pcm(chip, 0, NULL); if (err < 0) { snd_printk(KERN_ERR PFX - "error creating new ad1848 PCM device\n"); + "error creating new WSS PCM device\n"); goto err_unmap2; } err = snd_wss_mixer(chip); diff -urp linux-ref/sound/isa/sgalaxy.c linux-2.6.26/sound/isa/sgalaxy.c --- linux-ref/sound/isa/sgalaxy.c 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/sound/isa/sgalaxy.c 2008-07-31 19:07:33.000000000 +0200 @@ -273,8 +273,9 @@ static int __devinit snd_sgalaxy_probe(s goto _err; card->private_data = chip; - if ((err = snd_ad1848_pcm(chip, 0, NULL)) < 0) { - snd_printdd(PFX "error creating new ad1848 PCM device\n"); + err = snd_wss_pcm(chip, 0, NULL); + if (err < 0) { + snd_printdd(PFX "error creating new WSS PCM device\n"); goto _err; } err = snd_wss_mixer(chip); diff -urp linux-ref/sound/isa/wss/wss_lib.c linux-2.6.26/sound/isa/wss/wss_lib.c --- linux-ref/sound/isa/wss/wss_lib.c 2008-07-31 19:07:26.000000000 +0200 +++ linux-2.6.26/sound/isa/wss/wss_lib.c 2008-07-31 19:16:22.000000000 +0200 @@ -380,7 +380,7 @@ static void snd_wss_busy_wait(struct snd for (timeout = 5; timeout > 0; timeout--) wss_inb(chip, CS4231P(REGSEL)); /* end of cleanup sequence */ - for (timeout = 250; + for (timeout = 25000; timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(10); @@ -413,6 +413,7 @@ void snd_wss_mce_down(struct snd_wss *ch unsigned long flags; unsigned long end_time; int timeout; + int hw_mask = WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK | WSS_HW_AD1848; snd_wss_busy_wait(chip); @@ -427,10 +428,8 @@ void snd_wss_mce_down(struct snd_wss *ch spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0 || - !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { + if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask)) return; - } /* * Wait for (possible -- during init auto-calibration may not be set) @@ -601,12 +600,14 @@ static void snd_wss_calibrate_mute(struc mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); - snd_wss_dout(chip, CS4231_LEFT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); - snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); - snd_wss_dout(chip, CS4231_MONO_CTRL, - mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) { + snd_wss_dout(chip, CS4231_LEFT_LINE_IN, + mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, + mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + snd_wss_dout(chip, CS4231_MONO_CTRL, + mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); + } if (chip->hardware == WSS_HW_INTERWAVE) { snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); @@ -706,7 +707,10 @@ static void snd_wss_capture_format(struc snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); } - snd_wss_out(chip, CS4231_REC_FORMAT, cdfr); + if (chip->hardware & WSS_HW_AD1848_MASK) + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, cdfr); + else + snd_wss_out(chip, CS4231_REC_FORMAT, cdfr); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); } @@ -818,7 +822,9 @@ static void snd_wss_init(struct snd_wss snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + snd_wss_out(chip, CS4231_REC_FORMAT, + chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); @@ -844,20 +850,24 @@ static int snd_wss_open(struct snd_wss * } /* ok. now enable and ack CODEC IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IRQ_STATUS, - CS4231_PLAYBACK_IRQ | - CS4231_RECORD_IRQ | - CS4231_TIMER_IRQ); - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) { + snd_wss_out(chip, CS4231_IRQ_STATUS, + CS4231_PLAYBACK_IRQ | + CS4231_RECORD_IRQ | + CS4231_TIMER_IRQ); + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + } wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE; snd_wss_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); - snd_wss_out(chip, CS4231_IRQ_STATUS, - CS4231_PLAYBACK_IRQ | - CS4231_RECORD_IRQ | - CS4231_TIMER_IRQ); - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) { + snd_wss_out(chip, CS4231_IRQ_STATUS, + CS4231_PLAYBACK_IRQ | + CS4231_RECORD_IRQ | + CS4231_TIMER_IRQ); + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + } spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; @@ -879,7 +889,8 @@ static void snd_wss_close(struct snd_wss /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE; @@ -902,7 +913,8 @@ static void snd_wss_close(struct snd_wss } /* clear IRQ again */ - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1023,7 +1035,13 @@ static int snd_wss_capture_prepare(struc chip->c_dma_size = size; chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); - count = snd_wss_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; + if (chip->hardware & WSS_HW_AD1848_MASK) + count = snd_wss_get_count(chip->image[CS4231_PLAYBK_FORMAT], + count); + else + count = snd_wss_get_count(chip->image[CS4231_REC_FORMAT], + count); + count--; if (chip->single_dma && chip->hardware != WSS_HW_INTERWAVE) { snd_wss_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); snd_wss_out(chip, CS4231_PLY_UPR_CNT, @@ -1341,6 +1359,11 @@ static int snd_wss_playback_open(struct runtime->hw = snd_wss_playback; + /* hardware limitation of older chipsets */ + if (chip->hardware & WSS_HW_AD1848_MASK) + runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_S16_BE); + /* hardware bug in InterWave chipset */ if (chip->hardware == WSS_HW_INTERWAVE && chip->dma1 > 3) runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_MU_LAW; @@ -1379,6 +1402,11 @@ static int snd_wss_capture_open(struct s runtime->hw = snd_wss_capture; + /* hardware limitation of older chipsets */ + if (chip->hardware & WSS_HW_AD1848_MASK) + runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_S16_BE); + /* hardware limitation of cheap chips */ if (chip->hardware == WSS_HW_CS4235 || chip->hardware == WSS_HW_CS4239) @@ -1423,6 +1451,26 @@ static int snd_wss_capture_close(struct return 0; } +static void snd_wss_thinkpad_twiddle(struct snd_wss *chip, int on) +{ + int tmp; + + if (!chip->thinkpad_flag) + return; + + outb(0x1c, AD1848_THINKPAD_CTL_PORT1); + tmp = inb(AD1848_THINKPAD_CTL_PORT2); + + if (on) + /* turn it on */ + tmp |= AD1848_THINKPAD_CS4248_ENABLE_BIT; + else + /* turn it off */ + tmp &= ~AD1848_THINKPAD_CS4248_ENABLE_BIT; + + outb(tmp, AD1848_THINKPAD_CTL_PORT2); +} + #ifdef CONFIG_PM /* lowlevel suspend callback for CS4231 */ @@ -1436,6 +1484,8 @@ static void snd_wss_suspend(struct snd_w for (reg = 0; reg < 32; reg++) chip->image[reg] = snd_wss_in(chip, reg); spin_unlock_irqrestore(&chip->reg_lock, flags); + if (chip->thinkpad_flag) + snd_wss_thinkpad_twiddle(chip, 0); } /* lowlevel resume callback for CS4231 */ @@ -1445,6 +1495,8 @@ static void snd_wss_resume(struct snd_ws unsigned long flags; /* int timeout; */ + if (chip->thinkpad_flag) + snd_wss_thinkpad_twiddle(chip, 1); snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); for (reg = 0; reg < 32; reg++) { @@ -1542,6 +1594,14 @@ const char *snd_wss_chip_id(struct snd_w return "AD1845"; case WSS_HW_OPTI93X: return "OPTi 93x"; + case WSS_HW_AD1847: + return "AD1847"; + case WSS_HW_AD1848: + return "AD1848"; + case WSS_HW_CS4248: + return "CS4248"; + case WSS_HW_CMI8330: + return "CMI8330/C3D"; default: return "???"; } @@ -1704,7 +1764,8 @@ int snd_wss_pcm(struct snd_wss *chip, in struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "CS4231", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(chip->card, "WSS", device, 1, 1, &pcm); + if (err < 0) return err; spin_lock_init(&chip->reg_lock); @@ -1714,6 +1775,12 @@ int snd_wss_pcm(struct snd_wss *chip, in snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); + /* temporary */ + if (chip->hardware & WSS_HW_AD1848_MASK) { + chip->rate_constraint = snd_wss_xrate; + chip->set_playback_format = snd_wss_playback_format; + chip->set_capture_format = snd_wss_capture_format; + } /* global setup */ pcm->private_data = chip; pcm->info_flags = 0; @@ -2134,6 +2201,13 @@ int snd_wss_mixer(struct snd_wss *chip) } EXPORT_SYMBOL(snd_wss_mixer); +const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction) +{ + return direction == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_wss_playback_ops : &snd_wss_capture_ops; +} +EXPORT_SYMBOL(snd_wss_get_pcm_ops); + /* * INIT part */ ---------------------------------------------------------------------- Rowerem po Roztoczu. Zobacz relacje >>> http://link.interia.pl/f1e65

2 1

[alsa-devel] [PATCH 08/11] wss_lib: use wss mixer code instead of ad1848 one
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss mixer code and kill the ad1848 mixer code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Changes since previous version: 1. Nicer formatting of WSS_SINGLE/WSS_DOUBLE and TLV macros. 2. TLV macro definitions moved to the wss.h directly. diff -urp linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-31 19:03:47.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 2008-07-31 19:03:53.000000000 +0200 @@ -115,6 +115,5 @@ int snd_ad1848_create(struct snd_card *c int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); -int snd_ad1848_mixer(struct snd_wss *chip); #endif /* __SOUND_AD1848_H */ diff -urp linux-ref/include/sound/wss.h linux-2.6.26/include/sound/wss.h --- linux-ref/include/sound/wss.h 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/include/sound/wss.h 2008-07-31 19:03:53.000000000 +0200 @@ -193,6 +193,31 @@ int snd_wss_put_single(struct snd_kcontr .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ (shift_right << 19) | (mask << 24) | (invert << 22) } +#define WSS_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, \ + .index = xindex, \ + .info = snd_wss_info_single, \ + .get = snd_wss_get_single, \ + .put = snd_wss_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } + +#define WSS_DOUBLE_TLV(xname, xindex, left_reg, right_reg, \ + shift_left, shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, \ + .index = xindex, \ + .info = snd_wss_info_double, \ + .get = snd_wss_get_double, \ + .put = snd_wss_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + + int snd_wss_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_wss_get_double(struct snd_kcontrol *kcontrol, diff -urp linux-ref/sound/isa/ad1848/ad1848.c linux-2.6.26/sound/isa/ad1848/ad1848.c --- linux-ref/sound/isa/ad1848/ad1848.c 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848.c 2008-07-31 19:03:53.000000000 +0200 @@ -106,7 +106,7 @@ static int __devinit snd_ad1848_probe(st if (error < 0) goto out; - error = snd_ad1848_mixer(chip); + error = snd_wss_mixer(chip); if (error < 0) goto out; diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-31 19:03:47.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 2008-07-31 19:05:04.000000000 +0200 @@ -995,267 +995,6 @@ const struct snd_pcm_ops *snd_ad1848_get EXPORT_SYMBOL(snd_ad1848_get_pcm_ops); /* - * MIXER part - */ - -static int snd_ad1848_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_ad1848_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[AD1848_LEFT_INPUT] & AD1848_MIXS_ALL) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[AD1848_RIGHT_INPUT] & AD1848_MIXS_ALL) >> 6; - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static int snd_ad1848_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->reg_lock, flags); - left = (chip->image[AD1848_LEFT_INPUT] & ~AD1848_MIXS_ALL) | left; - right = (chip->image[AD1848_RIGHT_INPUT] & ~AD1848_MIXS_ALL) | right; - change = left != chip->image[AD1848_LEFT_INPUT] || - right != chip->image[AD1848_RIGHT_INPUT]; - snd_ad1848_out(chip, AD1848_LEFT_INPUT, left); - snd_ad1848_out(chip, AD1848_RIGHT_INPUT, right); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -static int snd_ad1848_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_ad1848_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_ad1848_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->reg_lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_ad1848_out(chip, reg, val); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -static int snd_ad1848_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_ad1848_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_ad1848_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->reg_lock, flags); - if (left_reg != right_reg) { - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_ad1848_out(chip, left_reg, val1); - snd_ad1848_out(chip, right_reg, val2); - } else { - val1 = (chip->image[left_reg] & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2; - change = val1 != chip->image[left_reg]; - snd_ad1848_out(chip, left_reg, val1); - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -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); - -#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ - .name = xname, \ - .index = xindex, \ - .info = snd_ad1848_info_single, \ - .get = snd_ad1848_get_single, \ - .put = snd_ad1848_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ - .tlv = { .p = (xtlv) } } - -#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ - .name = xname, \ - .index = xindex, \ - .info = snd_ad1848_info_double, \ - .get = snd_ad1848_get_double, \ - .put = snd_ad1848_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ - (shift_right << 19) | (mask << 24) | (invert << 22), \ - .tlv = { .p = (xtlv) } } - -static struct snd_kcontrol_new snd_ad1848_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, - AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("PCM Playback Volume", 0, - AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), -WSS_DOUBLE("Aux Playback Switch", 0, - AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 0, - AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, - AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 1, - AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -AD1848_DOUBLE_TLV("Capture Volume", 0, - AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0, - db_scale_rec_gain), -{ - .name = "Capture Source", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_mux, - .get = snd_ad1848_get_mux, - .put = snd_ad1848_put_mux, -}, -WSS_SINGLE("Loopback Capture Switch", 0, - AD1848_LOOPBACK, 0, 1, 0), -AD1848_SINGLE_TLV("Loopback Capture Volume", 0, - AD1848_LOOPBACK, 1, 63, 0, - db_scale_6bit), -}; - -int snd_ad1848_mixer(struct snd_wss *chip) -{ - struct snd_card *card; - struct snd_pcm *pcm; - unsigned int idx; - int err; - - snd_assert(chip != NULL && chip->pcm != NULL, return -EINVAL); - - pcm = chip->pcm; - card = chip->card; - - strcpy(card->mixername, pcm->name); - - 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; - } - - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_mixer); - -/* * INIT part */ diff -urp linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 19:03:59.000000000 +0200 @@ -757,6 +757,15 @@ static int __devinit snd_opti9xx_probe(s error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; +#else + error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, + chip->dma1, WSS_HW_DETECT, &codec); + if (error < 0) + return error; + error = snd_ad1848_pcm(codec, 0, &pcm); + if (error < 0) + return error; +#endif error = snd_wss_mixer(codec); if (error < 0) return error; @@ -764,7 +773,8 @@ static int __devinit snd_opti9xx_probe(s error = snd_wss_timer(codec, 0, &timer); if (error < 0) return error; -#else /* OPTI93X */ +#endif +#ifdef OPTi93X error = request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); if (error < 0) { @@ -772,16 +782,6 @@ static int __devinit snd_opti9xx_probe(s return error; } #endif -#else - if ((error = snd_ad1848_create(card, chip->wss_base + 4, - chip->irq, chip->dma1, - WSS_HW_DETECT, &codec)) < 0) - return error; - if ((error = snd_ad1848_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_ad1848_mixer(codec)) < 0) - return error; -#endif strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) diff -urp linux-ref/sound/isa/sc6000.c linux-2.6.26/sound/isa/sc6000.c --- linux-ref/sound/isa/sc6000.c 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/sound/isa/sc6000.c 2008-07-31 19:03:59.000000000 +0200 @@ -560,9 +560,9 @@ static int __devinit snd_sc6000_probe(st "error creating new ad1848 PCM device\n"); goto err_unmap2; } - err = snd_ad1848_mixer(chip); + err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR PFX "error creating new ad1848 mixer\n"); + snd_printk(KERN_ERR PFX "error creating new WSS mixer\n"); goto err_unmap2; } err = snd_sc6000_mixer(chip); diff -urp linux-ref/sound/isa/sgalaxy.c linux-2.6.26/sound/isa/sgalaxy.c --- linux-ref/sound/isa/sgalaxy.c 2008-07-31 18:59:20.000000000 +0200 +++ linux-2.6.26/sound/isa/sgalaxy.c 2008-07-31 19:03:59.000000000 +0200 @@ -277,8 +277,9 @@ static int __devinit snd_sgalaxy_probe(s snd_printdd(PFX "error creating new ad1848 PCM device\n"); goto _err; } - if ((err = snd_ad1848_mixer(chip)) < 0) { - snd_printdd(PFX "error creating new ad1848 mixer\n"); + err = snd_wss_mixer(chip); + if (err < 0) { + snd_printdd(PFX "error creating new WSS mixer\n"); goto _err; } if ((err = snd_sgalaxy_mixer(chip)) < 0) { diff -urp linux-ref/sound/isa/wss/wss_lib.c linux-2.6.26/sound/isa/wss/wss_lib.c --- linux-ref/sound/isa/wss/wss_lib.c 2008-07-31 18:40:55.000000000 +0200 +++ linux-2.6.26/sound/isa/wss/wss_lib.c 2008-07-31 19:03:59.000000000 +0200 @@ -33,6 +33,7 @@ #include <sound/core.h> #include <sound/wss.h> #include <sound/pcm_params.h> +#include <sound/tlv.h> #include <asm/io.h> #include <asm/dma.h> @@ -1957,16 +1958,58 @@ int snd_wss_put_double(struct snd_kcontr val1 <<= shift_left; val2 <<= shift_right; spin_lock_irqsave(&chip->reg_lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_wss_out(chip, left_reg, val1); - snd_wss_out(chip, right_reg, val2); + if (left_reg != right_reg) { + val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->image[left_reg] || + val2 != chip->image[right_reg]; + snd_wss_out(chip, left_reg, val1); + snd_wss_out(chip, right_reg, val2); + } else { + mask = (mask << shift_left) | (mask << shift_right); + val1 = (chip->image[left_reg] & ~mask) | val1 | val2; + change = val1 != chip->image[left_reg]; + snd_wss_out(chip, left_reg, val1); + } spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } 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 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_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, + 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), @@ -2071,6 +2114,14 @@ 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++) { err = snd_ctl_add(card, ---------------------------------------------------------------------- Galeria absurdow. zobacz >>> http://link.interia.pl/f1e5e

2 1

[alsa-devel] [PATCH 07/11] wss_lib: use CS4231P instead of AD1848P (kill the AD1848P)
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use CS4231P instead of AD1848P (kill the AD1848P). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Changes since previous version: 1. No reformatting of the snd_ad1848_debug() as the file is going to be killed. 1. Nicer formatting of WSS_SINGLE and WSS_DOUBLE macros. diff -urp linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-31 18:59:20.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 2008-07-31 18:59:06.000000000 +0200 @@ -27,15 +27,6 @@ #include "wss.h" /* temporary till the driver is removed */ -/* IO ports */ - -#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) - -#define c_d_c_AD1848REGSEL 0 -#define c_d_c_AD1848REG 1 -#define c_d_c_AD1848STATUS 2 -#define c_d_c_AD1848PIO 3 - /* codec registers */ #define AD1848_LEFT_INPUT 0x00 /* left input control */ diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-31 18:59:20.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 2008-07-31 19:01:10.000000000 +0200 @@ -103,7 +103,7 @@ static void snd_ad1848_wait(struct snd_w int timeout; for (timeout = 250; timeout > 0; timeout--) { - if ((inb(AD1848P(chip, REGSEL)) & AD1848_INIT) == 0) + if ((inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) == 0) break; udelay(100); } @@ -115,12 +115,12 @@ void snd_ad1848_out(struct snd_wss *chip { snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "auto calibration time out - " "reg = 0x%x, value = 0x%x\n", reg, value); #endif - outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); - outb(chip->image[reg] = value, AD1848P(chip, REG)); + outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); + outb(chip->image[reg] = value, chip->port + CS4231P(REG)); mb(); snd_printdd("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); @@ -132,8 +132,8 @@ static void snd_ad1848_dout(struct snd_w unsigned char reg, unsigned char value) { snd_ad1848_wait(chip); - outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); - outb(value, AD1848P(chip, REG)); + outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); + outb(value, chip->port + CS4231P(REG)); mb(); } @@ -141,37 +141,37 @@ static unsigned char snd_ad1848_in(struc { snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "auto calibration time out - " "reg = 0x%x\n", reg); #endif - outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); + outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); mb(); - return inb(AD1848P(chip, REG)); + return inb(chip->port + CS4231P(REG)); } #if 0 static void snd_ad1848_debug(struct snd_wss *chip) { - printk("AD1848 REGS: INDEX = 0x%02x ", inb(AD1848P(chip, REGSEL))); - printk(" STATUS = 0x%02x\n", inb(AD1848P(chip, STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_ad1848_in(chip, 0x00)); - printk(" 0x08: playback format = 0x%02x\n", snd_ad1848_in(chip, 0x08)); - printk(" 0x01: right input = 0x%02x ", snd_ad1848_in(chip, 0x01)); - printk(" 0x09: iface (CFIG 1) = 0x%02x\n", snd_ad1848_in(chip, 0x09)); - printk(" 0x02: AUXA left = 0x%02x ", snd_ad1848_in(chip, 0x02)); - printk(" 0x0a: pin control = 0x%02x\n", snd_ad1848_in(chip, 0x0a)); - printk(" 0x03: AUXA right = 0x%02x ", snd_ad1848_in(chip, 0x03)); - printk(" 0x0b: init & status = 0x%02x\n", snd_ad1848_in(chip, 0x0b)); - printk(" 0x04: AUXB left = 0x%02x ", snd_ad1848_in(chip, 0x04)); - printk(" 0x0c: revision & mode = 0x%02x\n", snd_ad1848_in(chip, 0x0c)); - printk(" 0x05: AUXB right = 0x%02x ", snd_ad1848_in(chip, 0x05)); - printk(" 0x0d: loopback = 0x%02x\n", snd_ad1848_in(chip, 0x0d)); - printk(" 0x06: left output = 0x%02x ", snd_ad1848_in(chip, 0x06)); - printk(" 0x0e: data upr count = 0x%02x\n", snd_ad1848_in(chip, 0x0e)); - printk(" 0x07: right output = 0x%02x ", snd_ad1848_in(chip, 0x07)); - printk(" 0x0f: data lwr count = 0x%02x\n", snd_ad1848_in(chip, 0x0f)); + printk(KERN_DEBUG "AD1848 REGS: INDEX = 0x%02x ", inb(chip->port + CS4231P(REGSEL))); + printk(KERN_DEBUG " STATUS = 0x%02x\n", inb(chip->port + CS4231P(STATUS))); + printk(KERN_DEBUG " 0x00: left input = 0x%02x ", snd_ad1848_in(chip, 0x00)); + printk(KERN_DEBUG " 0x08: playback format = 0x%02x\n", snd_ad1848_in(chip, 0x08)); + printk(KERN_DEBUG " 0x01: right input = 0x%02x ", snd_ad1848_in(chip, 0x01)); + printk(KERN_DEBUG " 0x09: iface (CFIG 1) = 0x%02x\n", snd_ad1848_in(chip, 0x09)); + printk(KERN_DEBUG " 0x02: AUXA left = 0x%02x ", snd_ad1848_in(chip, 0x02)); + printk(KERN_DEBUG " 0x0a: pin control = 0x%02x\n", snd_ad1848_in(chip, 0x0a)); + printk(KERN_DEBUG " 0x03: AUXA right = 0x%02x ", snd_ad1848_in(chip, 0x03)); + printk(KERN_DEBUG " 0x0b: init & status = 0x%02x\n", snd_ad1848_in(chip, 0x0b)); + printk(KERN_DEBUG " 0x04: AUXB left = 0x%02x ", snd_ad1848_in(chip, 0x04)); + printk(KERN_DEBUG " 0x0c: revision & mode = 0x%02x\n", snd_ad1848_in(chip, 0x0c)); + printk(KERN_DEBUG " 0x05: AUXB right = 0x%02x ", snd_ad1848_in(chip, 0x05)); + printk(KERN_DEBUG " 0x0d: loopback = 0x%02x\n", snd_ad1848_in(chip, 0x0d)); + printk(KERN_DEBUG " 0x06: left output = 0x%02x ", snd_ad1848_in(chip, 0x06)); + printk(KERN_DEBUG " 0x0e: data upr count = 0x%02x\n", snd_ad1848_in(chip, 0x0e)); + printk(KERN_DEBUG " 0x07: right output = 0x%02x ", snd_ad1848_in(chip, 0x07)); + printk(KERN_DEBUG " 0x0f: data lwr count = 0x%02x\n", snd_ad1848_in(chip, 0x0f)); } #endif @@ -187,16 +187,17 @@ static void snd_ad1848_mce_up(struct snd snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= AD1848_MCE; - timeout = inb(AD1848P(chip, REGSEL)); + timeout = inb(chip->port + CS4231P(REGSEL)); if (timeout == 0x80) snd_printk(KERN_WARNING "mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); if (!(timeout & AD1848_MCE)) - outb(chip->mce_bit | (timeout & 0x1f), AD1848P(chip, REGSEL)); + outb(chip->mce_bit | (timeout & 0x1f), + chip->port + CS4231P(REGSEL)); spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -207,21 +208,25 @@ static void snd_ad1848_mce_down(struct s spin_lock_irqsave(&chip->reg_lock, flags); for (timeout = 5; timeout > 0; timeout--) - inb(AD1848P(chip, REGSEL)); + inb(chip->port + CS4231P(REGSEL)); /* end of cleanup sequence */ - for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) + for (timeout = 12000; + timeout > 0 && (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT); + timeout--) udelay(100); snd_printdd("(1) timeout = %ld\n", timeout); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "mce_down [0x%lx] - auto calibration time out (0)\n", AD1848P(chip, REGSEL)); + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) + snd_printk(KERN_WARNING + "mce_down [0x%lx] - auto calibration time out (0)\n", + chip->port + CS4231P(REGSEL)); #endif chip->mce_bit &= ~AD1848_MCE; - reg = inb(AD1848P(chip, REGSEL)); - outb(chip->mce_bit | (reg & 0x1f), AD1848P(chip, REGSEL)); + reg = inb(chip->port + CS4231P(REGSEL)); + outb(chip->mce_bit | (reg & 0x1f), chip->port + CS4231P(REGSEL)); if (reg == 0x80) snd_printk(KERN_WARNING "mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); if ((reg & AD1848_MCE) == 0) { @@ -252,7 +257,8 @@ static void snd_ad1848_mce_down(struct s "mce_down - auto calibration time out (2)\n"); snd_printdd("(4) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); + snd_printd("mce_down - exit = 0x%x\n", + inb(chip->port + CS4231P(REGSEL))); } static unsigned int snd_ad1848_get_count(unsigned char format, @@ -412,8 +418,8 @@ static int snd_ad1848_open(struct snd_ws /* ok. now enable and ack CODEC IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ chip->image[AD1848_PIN_CTRL] |= AD1848_IRQ_ENABLE; snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -431,8 +437,8 @@ static void snd_ad1848_close(struct snd_ return; /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ chip->image[AD1848_PIN_CTRL] &= ~AD1848_IRQ_ENABLE; snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -449,8 +455,8 @@ static void snd_ad1848_close(struct snd_ /* clear IRQ again */ spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = 0; @@ -572,7 +578,7 @@ static irqreturn_t snd_ad1848_interrupt( snd_pcm_period_elapsed(chip->playback_substream); if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) snd_pcm_period_elapsed(chip->capture_substream); - outb(0, AD1848P(chip, STATUS)); /* clear global interrupt bit */ + outb(0, chip->port + CS4231P(STATUS)); /* clear global interrupt bit */ return IRQ_HANDLED; } @@ -638,8 +644,8 @@ static void snd_ad1848_resume(struct snd snd_ad1848_thinkpad_twiddle(chip, 1); /* clear any pendings IRQ */ - inb(AD1848P(chip, STATUS)); - outb(0, AD1848P(chip, STATUS)); + inb(chip->port + CS4231P(STATUS)); + outb(0, chip->port + CS4231P(STATUS)); mb(); snd_ad1848_mce_down(chip); @@ -662,7 +668,7 @@ static int snd_ad1848_probe(struct snd_w id = ad1847 = 0; for (i = 0; i < 1000; i++) { mb(); - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) udelay(500); else { spin_lock_irqsave(&chip->reg_lock, flags); @@ -707,8 +713,8 @@ static int snd_ad1848_probe(struct snd_w } } spin_lock_irqsave(&chip->reg_lock, flags); - inb(AD1848P(chip, STATUS)); /* clear any pendings IRQ */ - outb(0, AD1848P(chip, STATUS)); + inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ + outb(0, chip->port + CS4231P(STATUS)); mb(); spin_unlock_irqrestore(&chip->reg_lock, flags); ---------------------------------------------------------------------- Tanie rozmowy! Sprawdz >>> http://link.interia.pl/f1e91

2 1

[alsa-devel] [PATCH 06/11] wss_lib: replace ad1848 mixer element macros with wss ones
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss macros instead of ad1848 ones. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Changes since previous version: 1. Nicer formatting of the TLV macros. 1. Nicer formatting of the WSS_SINGLE/WSS_DOUBLE macros. diff -urp linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 2008-07-31 18:51:14.000000000 +0200 @@ -126,36 +126,4 @@ int snd_ad1848_pcm(struct snd_wss *chip, const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); int snd_ad1848_mixer(struct snd_wss *chip); -/* exported mixer stuffs */ -enum { AD1848_MIX_SINGLE, AD1848_MIX_DOUBLE, AD1848_MIX_CAPTURE }; - -#define AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) \ - ((reg) | ((shift) << 8) | ((mask) << 16) | ((invert) << 24)) -#define AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) \ - ((left_reg) | ((right_reg) << 8) | ((shift_left) << 16) | ((shift_right) << 19) | ((mask) << 24) | ((invert) << 22)) - -/* for ease of use */ -struct ad1848_mix_elem { - const char *name; - int index; - int type; - unsigned long private_value; - const unsigned int *tlv; -}; - -#define AD1848_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) } - -#define AD1848_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) } - -int snd_ad1848_add_ctl_elem(struct snd_wss *chip, - const struct ad1848_mix_elem *c); - #endif /* __SOUND_AD1848_H */ diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 2008-07-31 18:53:32.000000000 +0200 @@ -1163,88 +1163,64 @@ static int snd_ad1848_put_double(struct return change; } -/* - */ -int snd_ad1848_add_ctl_elem(struct snd_wss *chip, - const struct ad1848_mix_elem *c) -{ - static struct snd_kcontrol_new newctls[] = { - [AD1848_MIX_SINGLE] = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_single, - .get = snd_ad1848_get_single, - .put = snd_ad1848_put_single, - }, - [AD1848_MIX_DOUBLE] = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_double, - .get = snd_ad1848_get_double, - .put = snd_ad1848_put_double, - }, - [AD1848_MIX_CAPTURE] = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_mux, - .get = snd_ad1848_get_mux, - .put = snd_ad1848_put_mux, - }, - }; - struct snd_kcontrol *ctl; - int err; - - ctl = snd_ctl_new1(&newctls[c->type], chip); - if (! ctl) - return -ENOMEM; - strlcpy(ctl->id.name, c->name, sizeof(ctl->id.name)); - ctl->id.index = c->index; - ctl->private_value = c->private_value; - if (c->tlv) { - ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - ctl->tlv.p = c->tlv; - } - if ((err = snd_ctl_add(chip->card, ctl)) < 0) - return err; - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_add_ctl_elem); - 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); #define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .name = xname, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, \ .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ - .tlv = xtlv } + .info = snd_ad1848_info_single, \ + .get = snd_ad1848_get_single, \ + .put = snd_ad1848_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } #define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .name = xname, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, \ .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ - .tlv = xtlv } - -static struct ad1848_mix_elem snd_ad1848_controls[] = { -AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), -AD1848_DOUBLE("Aux Playback Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -AD1848_DOUBLE("Aux Playback Switch", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -AD1848_DOUBLE_TLV("Capture Volume", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0, - db_scale_rec_gain), + .info = snd_ad1848_info_double, \ + .get = snd_ad1848_get_double, \ + .put = snd_ad1848_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + +static struct snd_kcontrol_new snd_ad1848_controls[] = { +WSS_DOUBLE("PCM Playback Switch", 0, + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), +AD1848_DOUBLE_TLV("PCM Playback Volume", 0, + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), +WSS_DOUBLE("Aux Playback Switch", 0, + AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +AD1848_DOUBLE_TLV("Aux Playback Volume", 0, + AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_DOUBLE("Aux Playback Switch", 1, + AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +AD1848_DOUBLE_TLV("Aux Playback Volume", 1, + AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), +AD1848_DOUBLE_TLV("Capture Volume", 0, + AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0, + db_scale_rec_gain), { .name = "Capture Source", - .type = AD1848_MIX_CAPTURE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_ad1848_info_mux, + .get = snd_ad1848_get_mux, + .put = snd_ad1848_put_mux, }, -AD1848_SINGLE("Loopback Capture Switch", 0, AD1848_LOOPBACK, 0, 1, 0), -AD1848_SINGLE_TLV("Loopback Capture Volume", 0, AD1848_LOOPBACK, 1, 63, 0, - db_scale_6bit), +WSS_SINGLE("Loopback Capture Switch", 0, + AD1848_LOOPBACK, 0, 1, 0), +AD1848_SINGLE_TLV("Loopback Capture Volume", 0, + AD1848_LOOPBACK, 1, 63, 0, + db_scale_6bit), }; int snd_ad1848_mixer(struct snd_wss *chip) @@ -1261,9 +1237,12 @@ int snd_ad1848_mixer(struct snd_wss *chi strcpy(card->mixername, pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) - if ((err = snd_ad1848_add_ctl_elem(chip, &snd_ad1848_controls[idx])) < 0) + 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; + } return 0; } diff -urp linux-ref/sound/isa/cmi8330.c linux-2.6.26/sound/isa/cmi8330.c --- linux-ref/sound/isa/cmi8330.c 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/sound/isa/cmi8330.c 2008-07-31 18:56:09.000000000 +0200 @@ -174,32 +174,57 @@ MODULE_DEVICE_TABLE(pnp_card, snd_cmi833 #endif -static struct ad1848_mix_elem snd_cmi8330_controls[] __devinitdata = { -AD1848_DOUBLE("Master Playback Volume", 0, CMI8330_MASTVOL, CMI8330_MASTVOL, 4, 0, 15, 0), -AD1848_SINGLE("Loud Playback Switch", 0, CMI8330_MUTEMUX, 6, 1, 1), -AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), -AD1848_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), -AD1848_DOUBLE("Line Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, 4, 3, 1, 0), -AD1848_DOUBLE("Line Playback Volume", 0, CMI8330_LINVOL, CMI8330_LINVOL, 4, 0, 15, 0), -AD1848_DOUBLE("Line Capture Switch", 0, CMI8330_RMUX3D, CMI8330_RMUX3D, 2, 1, 1, 0), -AD1848_DOUBLE("Line Capture Volume", 0, CMI8330_LINGAIN, CMI8330_LINGAIN, 4, 0, 15, 0), -AD1848_DOUBLE("CD Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, 2, 1, 1, 0), -AD1848_DOUBLE("CD Capture Switch", 0, CMI8330_RMUX3D, CMI8330_RMUX3D, 4, 3, 1, 0), -AD1848_DOUBLE("CD Playback Volume", 0, CMI8330_CDINVOL, CMI8330_CDINVOL, 4, 0, 15, 0), -AD1848_DOUBLE("CD Capture Volume", 0, CMI8330_CDINGAIN, CMI8330_CDINGAIN, 4, 0, 15, 0), -AD1848_SINGLE("Mic Playback Switch", 0, CMI8330_MUTEMUX, 0, 1, 0), -AD1848_SINGLE("Mic Playback Volume", 0, CMI8330_OUTPUTVOL, 0, 7, 0), -AD1848_SINGLE("Mic Capture Switch", 0, CMI8330_RMUX3D, 0, 1, 0), -AD1848_SINGLE("Mic Capture Volume", 0, CMI8330_OUTPUTVOL, 5, 7, 0), -AD1848_DOUBLE("Wavetable Playback Switch", 0, CMI8330_RECMUX, CMI8330_RECMUX, 1, 0, 1, 0), -AD1848_DOUBLE("Wavetable Playback Volume", 0, CMI8330_WAVVOL, CMI8330_WAVVOL, 4, 0, 15, 0), -AD1848_DOUBLE("Wavetable Capture Switch", 0, CMI8330_RECMUX, CMI8330_RECMUX, 5, 4, 1, 0), -AD1848_DOUBLE("Wavetable Capture Volume", 0, CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0), -AD1848_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), -AD1848_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), -AD1848_SINGLE("FM Playback Switch", 0, CMI8330_RECMUX, 3, 1, 1), -AD1848_SINGLE(SNDRV_CTL_NAME_IEC958("Input ",CAPTURE,SWITCH), 0, CMI8330_RMUX3D, 7, 1, 1), -AD1848_SINGLE(SNDRV_CTL_NAME_IEC958("Input ",PLAYBACK,SWITCH), 0, CMI8330_MUTEMUX, 7, 1, 1), +static struct snd_kcontrol_new snd_cmi8330_controls[] __devinitdata = { +WSS_DOUBLE("Master Playback Volume", 0, + CMI8330_MASTVOL, CMI8330_MASTVOL, 4, 0, 15, 0), +WSS_SINGLE("Loud Playback Switch", 0, + CMI8330_MUTEMUX, 6, 1, 1), +WSS_DOUBLE("PCM Playback Switch", 0, + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Playback Volume", 0, + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE("Line Playback Switch", 0, + CMI8330_MUTEMUX, CMI8330_MUTEMUX, 4, 3, 1, 0), +WSS_DOUBLE("Line Playback Volume", 0, + CMI8330_LINVOL, CMI8330_LINVOL, 4, 0, 15, 0), +WSS_DOUBLE("Line Capture Switch", 0, + CMI8330_RMUX3D, CMI8330_RMUX3D, 2, 1, 1, 0), +WSS_DOUBLE("Line Capture Volume", 0, + CMI8330_LINGAIN, CMI8330_LINGAIN, 4, 0, 15, 0), +WSS_DOUBLE("CD Playback Switch", 0, + CMI8330_MUTEMUX, CMI8330_MUTEMUX, 2, 1, 1, 0), +WSS_DOUBLE("CD Capture Switch", 0, + CMI8330_RMUX3D, CMI8330_RMUX3D, 4, 3, 1, 0), +WSS_DOUBLE("CD Playback Volume", 0, + CMI8330_CDINVOL, CMI8330_CDINVOL, 4, 0, 15, 0), +WSS_DOUBLE("CD Capture Volume", 0, + CMI8330_CDINGAIN, CMI8330_CDINGAIN, 4, 0, 15, 0), +WSS_SINGLE("Mic Playback Switch", 0, + CMI8330_MUTEMUX, 0, 1, 0), +WSS_SINGLE("Mic Playback Volume", 0, + CMI8330_OUTPUTVOL, 0, 7, 0), +WSS_SINGLE("Mic Capture Switch", 0, + CMI8330_RMUX3D, 0, 1, 0), +WSS_SINGLE("Mic Capture Volume", 0, + CMI8330_OUTPUTVOL, 5, 7, 0), +WSS_DOUBLE("Wavetable Playback Switch", 0, + CMI8330_RECMUX, CMI8330_RECMUX, 1, 0, 1, 0), +WSS_DOUBLE("Wavetable Playback Volume", 0, + CMI8330_WAVVOL, CMI8330_WAVVOL, 4, 0, 15, 0), +WSS_DOUBLE("Wavetable Capture Switch", 0, + CMI8330_RECMUX, CMI8330_RECMUX, 5, 4, 1, 0), +WSS_DOUBLE("Wavetable Capture Volume", 0, + CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0), +WSS_SINGLE("3D Control - Switch", 0, + CMI8330_RMUX3D, 5, 1, 1), +WSS_SINGLE("PC Speaker Playback Volume", 0, + CMI8330_OUTPUTVOL, 3, 3, 0), +WSS_SINGLE("FM Playback Switch", 0, + CMI8330_RECMUX, 3, 1, 1), +WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0, + CMI8330_RMUX3D, 7, 1, 1), +WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0, + CMI8330_MUTEMUX, 7, 1, 1), }; #ifdef ENABLE_SB_MIXER @@ -268,7 +293,10 @@ static int __devinit snd_cmi8330_mixer(s strcpy(card->mixername, "CMI8330/C3D"); for (idx = 0; idx < ARRAY_SIZE(snd_cmi8330_controls); idx++) { - if ((err = snd_ad1848_add_ctl_elem(acard->wss, &snd_cmi8330_controls[idx])) < 0) + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cmi8330_controls[idx], + acard->wss)); + if (err < 0) return err; } diff -urp linux-ref/sound/isa/sgalaxy.c linux-2.6.26/sound/isa/sgalaxy.c --- linux-ref/sound/isa/sgalaxy.c 2008-07-31 18:51:25.000000000 +0200 +++ linux-2.6.26/sound/isa/sgalaxy.c 2008-07-31 18:57:31.000000000 +0200 @@ -175,9 +175,11 @@ static int __devinit snd_sgalaxy_detect( return snd_sgalaxy_setup_wss(wssport[dev], irq, dma); } -static struct ad1848_mix_elem snd_sgalaxy_controls[] = { -AD1848_DOUBLE("Aux Playback Switch", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), -AD1848_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) +static struct snd_kcontrol_new snd_sgalaxy_controls[] = { +WSS_DOUBLE("Aux Playback Switch", 0, + SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 0, + SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) }; static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip) @@ -210,7 +212,9 @@ static int __devinit snd_sgalaxy_mixer(s return err; /* build AUX2 input */ for (idx = 0; idx < ARRAY_SIZE(snd_sgalaxy_controls); idx++) { - if ((err = snd_ad1848_add_ctl_elem(chip, &snd_sgalaxy_controls[idx])) < 0) + err = snd_ctl_add(card, + snd_ctl_new1(&snd_sgalaxy_controls[idx], chip)); + if (err < 0) return err; } return 0; ---------------------------------------------------------------------- Tylko dla detektywow! 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2 1

[alsa-devel] [PATCH 05/11] wss_lib: use wss constants instead of ad1848 ones
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use wss constants for mode. Move ad1848 hardware constants to the wss.h. Move mixer tlv macros into the ad1848_lib.c from the ad1848.h. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Changes since previous version: 1. No reformatting of switch in the ad1848_lib.c in the snd_ad1848_chip_id() as the file is going to be killed anyway. diff -urp linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-31 18:46:34.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 2008-07-31 18:46:41.000000000 +0200 @@ -106,24 +106,6 @@ #define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ #define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ -/* defines for codec.mode */ - -#define AD1848_MODE_NONE 0x0000 -#define AD1848_MODE_PLAY 0x0001 -#define AD1848_MODE_CAPTURE 0x0002 -#define AD1848_MODE_TIMER 0x0004 -#define AD1848_MODE_OPEN (AD1848_MODE_PLAY|AD1848_MODE_CAPTURE|AD1848_MODE_TIMER) -#define AD1848_MODE_RUNNING 0x0010 - -/* defines for codec.hardware */ - -#define AD1848_HW_DETECT 0x0000 /* let AD1848 driver detect chip */ -#define AD1848_HW_AD1847 0x0001 /* AD1847 chip */ -#define AD1848_HW_AD1848 0x0002 /* AD1848 chip */ -#define AD1848_HW_CS4248 0x0003 /* CS4248 chip */ -#define AD1848_HW_CMI8330 0x0004 /* CMI8330 chip */ -#define AD1848_HW_THINKPAD 0x0005 /* Thinkpad 360/750/755 */ - /* IBM Thinkpad specific stuff */ #define AD1848_THINKPAD_CTL_PORT1 0x15e8 #define AD1848_THINKPAD_CTL_PORT2 0x15e9 @@ -167,26 +149,12 @@ struct ad1848_mix_elem { .type = AD1848_MIX_SINGLE, \ .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) } -#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ - .tlv = xtlv } - #define AD1848_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ { .name = xname, \ .index = xindex, \ .type = AD1848_MIX_DOUBLE, \ .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) } -#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ - .tlv = xtlv } - int snd_ad1848_add_ctl_elem(struct snd_wss *chip, const struct ad1848_mix_elem *c); diff -urp linux-ref/include/sound/wss.h linux-2.6.26/include/sound/wss.h --- linux-ref/include/sound/wss.h 2008-07-31 18:46:34.000000000 +0200 +++ linux-2.6.26/include/sound/wss.h 2008-07-31 18:46:41.000000000 +0200 @@ -55,6 +55,12 @@ #define WSS_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ #define WSS_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ #define WSS_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ +#define WSS_HW_AD1848_MASK 0x0800 /* AD1848 serie (half duplex) */ +#define WSS_HW_AD1847 0x0801 /* AD1847 chip */ +#define WSS_HW_AD1848 0x0802 /* AD1848 chip */ +#define WSS_HW_CS4248 0x0803 /* CS4248 chip */ +#define WSS_HW_CMI8330 0x0804 /* CMI8330 chip */ +#define WSS_HW_THINKPAD 0x0805 /* Thinkpad 360/750/755 */ /* compatible, but clones */ #define WSS_HW_INTERWAVE 0x1000 /* InterWave chip */ #define WSS_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ diff -urp linux-ref/sound/isa/ad1848/ad1848.c linux-2.6.26/sound/isa/ad1848/ad1848.c --- linux-ref/sound/isa/ad1848/ad1848.c 2008-07-31 18:46:34.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848.c 2008-07-31 18:46:41.000000000 +0200 @@ -96,7 +96,7 @@ static int __devinit snd_ad1848_probe(st return -EINVAL; error = snd_ad1848_create(card, port[n], irq[n], dma1[n], - thinkpad[n] ? AD1848_HW_THINKPAD : AD1848_HW_DETECT, &chip); + thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, &chip); if (error < 0) goto out; diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-31 18:46:35.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 2008-07-31 18:48:41.000000000 +0200 @@ -283,14 +283,12 @@ static int snd_ad1848_trigger(struct snd return 0; } snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] |= what); - chip->mode |= AD1848_MODE_RUNNING; } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { if (!(chip->image[AD1848_IFACE_CTRL] & what)) { spin_unlock(&chip->reg_lock); return 0; } snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] &= ~what); - chip->mode &= ~AD1848_MODE_RUNNING; } else { result = -EINVAL; } @@ -378,7 +376,7 @@ static int snd_ad1848_open(struct snd_ws { unsigned long flags; - if (chip->mode & AD1848_MODE_OPEN) + if (chip->mode & WSS_MODE_OPEN) return -EAGAIN; snd_ad1848_mce_down(chip); @@ -570,11 +568,9 @@ static irqreturn_t snd_ad1848_interrupt( { struct snd_wss *chip = dev_id; - if ((chip->mode & AD1848_MODE_PLAY) && chip->playback_substream && - (chip->mode & AD1848_MODE_RUNNING)) + if ((chip->mode & WSS_MODE_PLAY) && chip->playback_substream) snd_pcm_period_elapsed(chip->playback_substream); - if ((chip->mode & AD1848_MODE_CAPTURE) && chip->capture_substream && - (chip->mode & AD1848_MODE_RUNNING)) + if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) snd_pcm_period_elapsed(chip->capture_substream); outb(0, AD1848P(chip, STATUS)); /* clear global interrupt bit */ return IRQ_HANDLED; @@ -690,19 +686,19 @@ static int snd_ad1848_probe(struct snd_w } if (id != 1) return -ENODEV; /* no valid device found */ - if (chip->hardware == AD1848_HW_DETECT) { + if (chip->hardware == WSS_HW_DETECT) { if (ad1847) { - chip->hardware = AD1848_HW_AD1847; + chip->hardware = WSS_HW_AD1847; } else { - chip->hardware = AD1848_HW_AD1848; + chip->hardware = WSS_HW_AD1848; rev = snd_ad1848_in(chip, AD1848_MISC_INFO); if (rev & 0x80) { - chip->hardware = AD1848_HW_CS4248; + chip->hardware = WSS_HW_CS4248; } else if ((rev & 0x0f) == 0x0a) { snd_ad1848_out(chip, AD1848_MISC_INFO, 0x40); for (i = 0; i < 16; ++i) { if (snd_ad1848_in(chip, i) != snd_ad1848_in(chip, i + 16)) { - chip->hardware = AD1848_HW_CMI8330; + chip->hardware = WSS_HW_CMI8330; break; } } @@ -782,7 +778,8 @@ static int snd_ad1848_playback_open(stru struct snd_pcm_runtime *runtime = substream->runtime; int err; - if ((err = snd_ad1848_open(chip, AD1848_MODE_PLAY)) < 0) + err = snd_ad1848_open(chip, WSS_MODE_PLAY); + if (err < 0) return err; chip->playback_substream = substream; runtime->hw = snd_ad1848_playback; @@ -798,7 +795,8 @@ static int snd_ad1848_capture_open(struc struct snd_pcm_runtime *runtime = substream->runtime; int err; - if ((err = snd_ad1848_open(chip, AD1848_MODE_CAPTURE)) < 0) + err = snd_ad1848_open(chip, WSS_MODE_RECORD); + if (err < 0) return err; chip->capture_substream = substream; runtime->hw = snd_ad1848_capture; @@ -812,7 +810,7 @@ static int snd_ad1848_playback_close(str { struct snd_wss *chip = snd_pcm_substream_chip(substream); - chip->mode &= ~AD1848_MODE_PLAY; + chip->mode &= ~WSS_MODE_PLAY; chip->playback_substream = NULL; snd_ad1848_close(chip); return 0; @@ -822,7 +820,7 @@ static int snd_ad1848_capture_close(stru { struct snd_wss *chip = snd_pcm_substream_chip(substream); - chip->mode &= ~AD1848_MODE_CAPTURE; + chip->mode &= ~WSS_MODE_RECORD; chip->capture_substream = NULL; snd_ad1848_close(chip); return 0; @@ -903,9 +901,9 @@ int snd_ad1848_create(struct snd_card *c chip->dma1 = dma; chip->dma2 = dma; - if (hardware == AD1848_HW_THINKPAD) { + if (hardware == WSS_HW_THINKPAD) { chip->thinkpad_flag = 1; - chip->hardware = AD1848_HW_DETECT; /* reset */ + chip->hardware = WSS_HW_DETECT; /* reset */ snd_ad1848_thinkpad_twiddle(chip, 1); } @@ -1214,6 +1212,20 @@ static const DECLARE_TLV_DB_SCALE(db_sca 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); +#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .name = xname, \ + .index = xindex, \ + .type = AD1848_MIX_SINGLE, \ + .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ + .tlv = xtlv } + +#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ +{ .name = xname, \ + .index = xindex, \ + .type = AD1848_MIX_DOUBLE, \ + .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ + .tlv = xtlv } + static struct ad1848_mix_elem snd_ad1848_controls[] = { AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), AD1848_DOUBLE_TLV("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, diff -urp linux-ref/sound/isa/cmi8330.c linux-2.6.26/sound/isa/cmi8330.c --- linux-ref/sound/isa/cmi8330.c 2008-07-31 18:46:35.000000000 +0200 +++ linux-2.6.26/sound/isa/cmi8330.c 2008-07-31 18:46:41.000000000 +0200 @@ -465,12 +465,12 @@ static int __devinit snd_cmi8330_probe(s wssport[dev] + 4, wssirq[dev], wssdma[dev], - AD1848_HW_DETECT, + WSS_HW_DETECT, &acard->wss)) < 0) { snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } - if (acard->wss->hardware != AD1848_HW_CMI8330) { + if (acard->wss->hardware != WSS_HW_CMI8330) { snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); return -ENODEV; } diff -urp linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 18:46:35.000000000 +0200 +++ linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 18:46:41.000000000 +0200 @@ -775,7 +775,7 @@ static int __devinit snd_opti9xx_probe(s #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, - AD1848_HW_DETECT, &codec)) < 0) + WSS_HW_DETECT, &codec)) < 0) return error; if ((error = snd_ad1848_pcm(codec, 0, &pcm)) < 0) return error; diff -urp linux-ref/sound/isa/sc6000.c linux-2.6.26/sound/isa/sc6000.c --- linux-ref/sound/isa/sc6000.c 2008-07-31 18:46:35.000000000 +0200 +++ linux-2.6.26/sound/isa/sc6000.c 2008-07-31 18:46:41.000000000 +0200 @@ -549,7 +549,7 @@ static int __devinit snd_sc6000_probe(st goto err_unmap2; err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, - AD1848_HW_DETECT, &chip); + WSS_HW_DETECT, &chip); if (err < 0) goto err_unmap2; card->private_data = chip; diff -urp linux-ref/sound/isa/sgalaxy.c linux-2.6.26/sound/isa/sgalaxy.c --- linux-ref/sound/isa/sgalaxy.c 2008-07-31 18:46:35.000000000 +0200 +++ linux-2.6.26/sound/isa/sgalaxy.c 2008-07-31 18:46:41.000000000 +0200 @@ -265,7 +265,7 @@ static int __devinit snd_sgalaxy_probe(s if ((err = snd_ad1848_create(card, wssport[dev] + 4, xirq, xdma1, - AD1848_HW_DETECT, &chip)) < 0) + WSS_HW_DETECT, &chip)) < 0) goto _err; card->private_data = chip; ---------------------------------------------------------------------- Najciekawsze miejsca w Polsce i na swiecie! 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2 1

[alsa-devel] [PATCH 04/11] wss_lib: use struct snd_wss instead of snd_ad1848
by Krzysztof Helt 05 Aug '08

05 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The snd_wss is superset of the snd_ad1848 so kill the latter and replace it with the snd_wss. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Only small formatting changes in the header comments. diff -urp linux-ref/include/sound/ad1848.h linux-2.6.26/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/include/sound/ad1848.h 2008-07-31 18:41:12.000000000 +0200 @@ -25,6 +25,8 @@ #include "pcm.h" #include <linux/interrupt.h> +#include "wss.h" /* temporary till the driver is removed */ + /* IO ports */ #define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) @@ -127,48 +129,20 @@ #define AD1848_THINKPAD_CTL_PORT2 0x15e9 #define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02 -struct snd_ad1848 { - unsigned long port; /* i/o port */ - struct resource *res_port; - int irq; /* IRQ line */ - int dma; /* data DMA */ - unsigned short version; /* version of CODEC chip */ - unsigned short mode; /* see to AD1848_MODE_XXXX */ - unsigned short hardware; /* see to AD1848_HW_XXXX */ - unsigned short single_dma:1; /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ - - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - struct snd_card *card; - - unsigned char image[32]; /* SGalaxy needs an access to extended registers */ - int mce_bit; - int calibrate_mute; - int dma_size; - int thinkpad_flag; /* Thinkpad CS4248 needs some extra help */ - -#ifdef CONFIG_PM - void (*suspend)(struct snd_ad1848 *chip); - void (*resume)(struct snd_ad1848 *chip); -#endif - - spinlock_t reg_lock; -}; - /* exported functions */ -void snd_ad1848_out(struct snd_ad1848 *chip, unsigned char reg, unsigned char value); +void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, + unsigned char value); int snd_ad1848_create(struct snd_card *card, unsigned long port, int irq, int dma, unsigned short hardware, - struct snd_ad1848 ** chip); + struct snd_wss **chip); -int snd_ad1848_pcm(struct snd_ad1848 * chip, int device, struct snd_pcm **rpcm); +int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); -int snd_ad1848_mixer(struct snd_ad1848 * chip); +int snd_ad1848_mixer(struct snd_wss *chip); /* exported mixer stuffs */ enum { AD1848_MIX_SINGLE, AD1848_MIX_DOUBLE, AD1848_MIX_CAPTURE }; @@ -213,6 +187,7 @@ struct ad1848_mix_elem { .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ .tlv = xtlv } -int snd_ad1848_add_ctl_elem(struct snd_ad1848 *chip, const struct ad1848_mix_elem *c); +int snd_ad1848_add_ctl_elem(struct snd_wss *chip, + const struct ad1848_mix_elem *c); #endif /* __SOUND_AD1848_H */ diff -urp linux-ref/include/sound/wss.h linux-2.6.26/include/sound/wss.h --- linux-ref/include/sound/wss.h 2008-07-31 18:40:55.000000000 +0200 +++ linux-2.6.26/include/sound/wss.h 2008-07-31 18:41:12.000000000 +0200 @@ -77,8 +77,10 @@ struct snd_wss { unsigned short mode; /* see to WSS_MODE_XXXX */ unsigned short hardware; /* see to WSS_HW_XXXX */ unsigned short hwshare; /* shared resources */ - unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ - ebus_flag:1; /* SPARC: EBUS present */ + unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit */ + /* daughter board) or dma1 == dma2 */ + ebus_flag:1, /* SPARC: EBUS present */ + thinkpad_flag:1; /* Thinkpad CS4248 needs extra help */ struct snd_card *card; struct snd_pcm *pcm; diff -urp linux-ref/sound/isa/ad1848/ad1848.c linux-2.6.26/sound/isa/ad1848/ad1848.c --- linux-ref/sound/isa/ad1848/ad1848.c 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848.c 2008-07-31 18:41:12.000000000 +0200 @@ -87,7 +87,7 @@ static int __devinit snd_ad1848_match(st static int __devinit snd_ad1848_probe(struct device *dev, unsigned int n) { struct snd_card *card; - struct snd_ad1848 *chip; + struct snd_wss *chip; struct snd_pcm *pcm; int error; @@ -142,7 +142,7 @@ static int __devexit snd_ad1848_remove(s static int snd_ad1848_suspend(struct device *dev, unsigned int n, pm_message_t state) { struct snd_card *card = dev_get_drvdata(dev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->suspend(chip); @@ -152,7 +152,7 @@ static int snd_ad1848_suspend(struct dev static int snd_ad1848_resume(struct device *dev, unsigned int n) { struct snd_card *card = dev_get_drvdata(dev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; chip->resume(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.26/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/ad1848/ad1848_lib.c 2008-07-31 18:41:12.000000000 +0200 @@ -98,7 +98,7 @@ static unsigned char snd_ad1848_original * Basic I/O functions */ -static void snd_ad1848_wait(struct snd_ad1848 *chip) +static void snd_ad1848_wait(struct snd_wss *chip) { int timeout; @@ -109,7 +109,7 @@ static void snd_ad1848_wait(struct snd_a } } -void snd_ad1848_out(struct snd_ad1848 *chip, +void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, unsigned char value) { @@ -128,7 +128,7 @@ void snd_ad1848_out(struct snd_ad1848 *c EXPORT_SYMBOL(snd_ad1848_out); -static void snd_ad1848_dout(struct snd_ad1848 *chip, +static void snd_ad1848_dout(struct snd_wss *chip, unsigned char reg, unsigned char value) { snd_ad1848_wait(chip); @@ -137,7 +137,7 @@ static void snd_ad1848_dout(struct snd_a mb(); } -static unsigned char snd_ad1848_in(struct snd_ad1848 *chip, unsigned char reg) +static unsigned char snd_ad1848_in(struct snd_wss *chip, unsigned char reg) { snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG @@ -152,7 +152,7 @@ static unsigned char snd_ad1848_in(struc #if 0 -static void snd_ad1848_debug(struct snd_ad1848 *chip) +static void snd_ad1848_debug(struct snd_wss *chip) { printk("AD1848 REGS: INDEX = 0x%02x ", inb(AD1848P(chip, REGSEL))); printk(" STATUS = 0x%02x\n", inb(AD1848P(chip, STATUS))); @@ -180,7 +180,7 @@ static void snd_ad1848_debug(struct snd_ * AD1848 detection / MCE routines */ -static void snd_ad1848_mce_up(struct snd_ad1848 *chip) +static void snd_ad1848_mce_up(struct snd_wss *chip) { unsigned long flags; int timeout; @@ -200,7 +200,7 @@ static void snd_ad1848_mce_up(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_ad1848_mce_down(struct snd_ad1848 *chip) +static void snd_ad1848_mce_down(struct snd_wss *chip) { unsigned long flags, timeout; int reg; @@ -268,7 +268,7 @@ static unsigned int snd_ad1848_get_count return size; } -static int snd_ad1848_trigger(struct snd_ad1848 *chip, unsigned char what, +static int snd_ad1848_trigger(struct snd_wss *chip, unsigned char what, int channel, int cmd) { int result = 0; @@ -337,7 +337,7 @@ static unsigned char snd_ad1848_get_form return rformat; } -static void snd_ad1848_calibrate_mute(struct snd_ad1848 *chip, int mute) +static void snd_ad1848_calibrate_mute(struct snd_wss *chip, int mute) { unsigned long flags; @@ -361,7 +361,8 @@ static void snd_ad1848_calibrate_mute(st spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_ad1848_set_data_format(struct snd_ad1848 *chip, struct snd_pcm_hw_params *hw_params) +static void snd_ad1848_set_data_format(struct snd_wss *chip, + struct snd_pcm_hw_params *hw_params) { if (hw_params == NULL) { chip->image[AD1848_DATA_FORMAT] = 0x20; @@ -373,7 +374,7 @@ static void snd_ad1848_set_data_format(s // snd_printk(">>> pmode = 0x%x, dfr = 0x%x\n", pstr->mode, chip->image[AD1848_DATA_FORMAT]); } -static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode) +static int snd_ad1848_open(struct snd_wss *chip, unsigned int mode) { unsigned long flags; @@ -424,7 +425,7 @@ static int snd_ad1848_open(struct snd_ad return 0; } -static void snd_ad1848_close(struct snd_ad1848 *chip) +static void snd_ad1848_close(struct snd_wss *chip) { unsigned long flags; @@ -464,21 +465,21 @@ static void snd_ad1848_close(struct snd_ static int snd_ad1848_playback_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); return snd_ad1848_trigger(chip, AD1848_PLAYBACK_ENABLE, SNDRV_PCM_STREAM_PLAYBACK, cmd); } static int snd_ad1848_capture_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); return snd_ad1848_trigger(chip, AD1848_CAPTURE_ENABLE, SNDRV_PCM_STREAM_CAPTURE, cmd); } static int snd_ad1848_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); unsigned long flags; int err; @@ -502,15 +503,16 @@ static int snd_ad1848_playback_hw_free(s static int snd_ad1848_playback_prepare(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma_size = size; + chip->p_dma_size = size; chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO); - snd_dma_program(chip->dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); + snd_dma_program(chip->dma1, runtime->dma_addr, size, + DMA_MODE_WRITE | DMA_AUTOINIT); count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; spin_lock_irqsave(&chip->reg_lock, flags); snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); @@ -522,7 +524,7 @@ static int snd_ad1848_playback_prepare(s static int snd_ad1848_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); unsigned long flags; int err; @@ -546,15 +548,16 @@ static int snd_ad1848_capture_hw_free(st static int snd_ad1848_capture_prepare(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma_size = size; + chip->c_dma_size = size; chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); - snd_dma_program(chip->dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); + snd_dma_program(chip->dma2, runtime->dma_addr, size, + DMA_MODE_READ | DMA_AUTOINIT); count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; spin_lock_irqsave(&chip->reg_lock, flags); snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); @@ -565,7 +568,7 @@ static int snd_ad1848_capture_prepare(st static irqreturn_t snd_ad1848_interrupt(int irq, void *dev_id) { - struct snd_ad1848 *chip = dev_id; + struct snd_wss *chip = dev_id; if ((chip->mode & AD1848_MODE_PLAY) && chip->playback_substream && (chip->mode & AD1848_MODE_RUNNING)) @@ -579,23 +582,23 @@ static irqreturn_t snd_ad1848_interrupt( static snd_pcm_uframes_t snd_ad1848_playback_pointer(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); size_t ptr; if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) return 0; - ptr = snd_dma_pointer(chip->dma, chip->dma_size); + ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } static snd_pcm_uframes_t snd_ad1848_capture_pointer(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); size_t ptr; if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE)) return 0; - ptr = snd_dma_pointer(chip->dma, chip->dma_size); + ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -603,8 +606,8 @@ static snd_pcm_uframes_t snd_ad1848_capt */ -static void snd_ad1848_thinkpad_twiddle(struct snd_ad1848 *chip, int on) { - +static void snd_ad1848_thinkpad_twiddle(struct snd_wss *chip, int on) +{ int tmp; if (!chip->thinkpad_flag) return; @@ -624,14 +627,14 @@ static void snd_ad1848_thinkpad_twiddle( } #ifdef CONFIG_PM -static void snd_ad1848_suspend(struct snd_ad1848 *chip) +static void snd_ad1848_suspend(struct snd_wss *chip) { snd_pcm_suspend_all(chip->pcm); if (chip->thinkpad_flag) snd_ad1848_thinkpad_twiddle(chip, 0); } -static void snd_ad1848_resume(struct snd_ad1848 *chip) +static void snd_ad1848_resume(struct snd_wss *chip) { int i; @@ -651,7 +654,7 @@ static void snd_ad1848_resume(struct snd } #endif /* CONFIG_PM */ -static int snd_ad1848_probe(struct snd_ad1848 * chip) +static int snd_ad1848_probe(struct snd_wss *chip) { unsigned long flags; int i, id, rev, ad1847; @@ -775,7 +778,7 @@ static struct snd_pcm_hardware snd_ad184 static int snd_ad1848_playback_open(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -783,15 +786,15 @@ static int snd_ad1848_playback_open(stru return err; chip->playback_substream = substream; runtime->hw = snd_ad1848_playback; - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.period_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); return 0; } static int snd_ad1848_capture_open(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -799,15 +802,15 @@ static int snd_ad1848_capture_open(struc return err; chip->capture_substream = substream; runtime->hw = snd_ad1848_capture; - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.period_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); return 0; } static int snd_ad1848_playback_close(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); chip->mode &= ~AD1848_MODE_PLAY; chip->playback_substream = NULL; @@ -817,7 +820,7 @@ static int snd_ad1848_playback_close(str static int snd_ad1848_capture_close(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); chip->mode &= ~AD1848_MODE_CAPTURE; chip->capture_substream = NULL; @@ -825,14 +828,14 @@ static int snd_ad1848_capture_close(stru return 0; } -static int snd_ad1848_free(struct snd_ad1848 *chip) +static int snd_ad1848_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); - if (chip->dma >= 0) { - snd_dma_disable(chip->dma); - free_dma(chip->dma); + if (chip->dma1 >= 0) { + snd_dma_disable(chip->dma1); + free_dma(chip->dma1); } kfree(chip); return 0; @@ -840,11 +843,11 @@ static int snd_ad1848_free(struct snd_ad static int snd_ad1848_dev_free(struct snd_device *device) { - struct snd_ad1848 *chip = device->device_data; + struct snd_wss *chip = device->device_data; return snd_ad1848_free(chip); } -static const char *snd_ad1848_chip_id(struct snd_ad1848 *chip) +static const char *snd_ad1848_chip_id(struct snd_wss *chip) { switch (chip->hardware) { case AD1848_HW_AD1847: return "AD1847"; @@ -859,12 +862,12 @@ int snd_ad1848_create(struct snd_card *c unsigned long port, int irq, int dma, unsigned short hardware, - struct snd_ad1848 ** rchip) + struct snd_wss **rchip) { static struct snd_device_ops ops = { .dev_free = snd_ad1848_dev_free, }; - struct snd_ad1848 *chip; + struct snd_wss *chip; int err; *rchip = NULL; @@ -875,7 +878,9 @@ int snd_ad1848_create(struct snd_card *c chip->card = card; chip->port = port; chip->irq = -1; - chip->dma = -1; + chip->dma1 = -1; + chip->dma2 = -1; + chip->single_dma = 1; chip->hardware = hardware; memcpy(&chip->image, &snd_ad1848_original_image, sizeof(snd_ad1848_original_image)); @@ -895,7 +900,8 @@ int snd_ad1848_create(struct snd_card *c snd_ad1848_free(chip); return -EBUSY; } - chip->dma = dma; + chip->dma1 = dma; + chip->dma2 = dma; if (hardware == AD1848_HW_THINKPAD) { chip->thinkpad_flag = 1; @@ -947,7 +953,7 @@ static struct snd_pcm_ops snd_ad1848_cap .pointer = snd_ad1848_capture_pointer, }; -int snd_ad1848_pcm(struct snd_ad1848 *chip, int device, struct snd_pcm **rpcm) +int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -964,7 +970,9 @@ int snd_ad1848_pcm(struct snd_ad1848 *ch snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, chip->dma > 3 ? 128*1024 : 64*1024); + 64 * 1024, + chip->dma1 > 3 ? + 128 * 1024 : 64 * 1024); chip->pcm = pcm; if (rpcm) @@ -1003,7 +1011,7 @@ static int snd_ad1848_info_mux(struct sn static int snd_ad1848_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); @@ -1015,7 +1023,7 @@ static int snd_ad1848_get_mux(struct snd static int snd_ad1848_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; unsigned short left, right; int change; @@ -1049,7 +1057,7 @@ static int snd_ad1848_info_single(struct static int snd_ad1848_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -1066,7 +1074,7 @@ static int snd_ad1848_get_single(struct static int snd_ad1848_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -1100,7 +1108,7 @@ static int snd_ad1848_info_double(struct static int snd_ad1848_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -1122,7 +1130,7 @@ static int snd_ad1848_get_double(struct static int snd_ad1848_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -1159,7 +1167,7 @@ static int snd_ad1848_put_double(struct /* */ -int snd_ad1848_add_ctl_elem(struct snd_ad1848 *chip, +int snd_ad1848_add_ctl_elem(struct snd_wss *chip, const struct ad1848_mix_elem *c) { static struct snd_kcontrol_new newctls[] = { @@ -1227,7 +1235,7 @@ AD1848_SINGLE_TLV("Loopback Capture Volu db_scale_6bit), }; -int snd_ad1848_mixer(struct snd_ad1848 *chip) +int snd_ad1848_mixer(struct snd_wss *chip) { struct snd_card *card; struct snd_pcm *pcm; diff -urp linux-ref/sound/isa/cmi8330.c linux-2.6.26/sound/isa/cmi8330.c --- linux-ref/sound/isa/cmi8330.c 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/cmi8330.c 2008-07-31 18:41:12.000000000 +0200 @@ -151,7 +151,7 @@ struct snd_cmi8330 { struct pnp_dev *play; #endif struct snd_card *card; - struct snd_ad1848 *wss; + struct snd_wss *wss; struct snd_sb *sb; struct snd_pcm *pcm; diff -urp linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 18:40:55.000000000 +0200 +++ linux-2.6.26/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-31 18:41:12.000000000 +0200 @@ -706,14 +706,10 @@ static int __devinit snd_opti9xx_probe(s static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(CS4231) || defined(OPTi93X) struct snd_wss *codec; #ifdef CS4231 struct snd_timer *timer; #endif -#else - struct snd_ad1848 *codec; -#endif struct snd_pcm *pcm; struct snd_rawmidi *rmidi; struct snd_hwdep *synth; diff -urp linux-ref/sound/isa/sc6000.c linux-2.6.26/sound/isa/sc6000.c --- linux-ref/sound/isa/sc6000.c 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/sc6000.c 2008-07-31 18:41:12.000000000 +0200 @@ -397,7 +397,7 @@ static int __devinit sc6000_init_board(c return 0; } -static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +static int __devinit snd_sc6000_mixer(struct snd_wss *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -483,7 +483,7 @@ static int __devinit snd_sc6000_probe(st int xirq = irq[dev]; int xdma = dma[dev]; struct snd_card *card; - struct snd_ad1848 *chip; + struct snd_wss *chip; struct snd_opl3 *opl3; char __iomem *vport; char __iomem *vmss_port; diff -urp linux-ref/sound/isa/sgalaxy.c linux-2.6.26/sound/isa/sgalaxy.c --- linux-ref/sound/isa/sgalaxy.c 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/sgalaxy.c 2008-07-31 18:41:12.000000000 +0200 @@ -180,7 +180,7 @@ AD1848_DOUBLE("Aux Playback Switch", 0, AD1848_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) }; -static int __devinit snd_sgalaxy_mixer(struct snd_ad1848 *chip) +static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -237,7 +237,7 @@ static int __devinit snd_sgalaxy_probe(s static int possible_dmas[] = {1, 3, 0, -1}; int err, xirq, xdma1; struct snd_card *card; - struct snd_ad1848 *chip; + struct snd_wss *chip; card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) @@ -312,7 +312,7 @@ static int snd_sgalaxy_suspend(struct de pm_message_t state) { struct snd_card *card = dev_get_drvdata(pdev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->suspend(chip); @@ -322,7 +322,7 @@ static int snd_sgalaxy_suspend(struct de static int snd_sgalaxy_resume(struct device *pdev, unsigned int n) { struct snd_card *card = dev_get_drvdata(pdev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; chip->resume(chip); snd_ad1848_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); ---------------------------------------------------------------------- Najciekawsze miejsca w Polsce i na swiecie! 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2 1

[alsa-devel] [PATCH] ALSA: ASoC V2: restrict sample rate and size in Freescale MPC8610 sound drivers
by Timur Tabi 04 Aug '08

04 Aug '08

The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. The spinlock used to protect SCR was also removed, because it just made the code more complicated. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for ASoC V2. A similar patch for ASoC V1 has already been posted and applied. sound/soc/fsl/fsl_ssi.c | 122 +++++++++++++++++++++++++++++++---------------- sound/soc/fsl/fsl_ssi.h | 3 +- 2 files changed, 82 insertions(+), 43 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bcb1880..4858ff1 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -241,18 +241,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * SSI needs to be disabled before updating the registers we set * here. */ - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + out_be32(&ssi->scr, 0); /* * Program the SSI into I2S Slave Non-Network Synchronous mode. - * Also enable the transmit and receive FIFO. Make sure TE and - * RE are disabled. - * - * FIXME: Little-endian samples require a different shift dir + * Also enable the transmit and receive FIFO. */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | - CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE, - CCSR_SSI_SCR_TFR_CLK_DIS | + setbits32(&ssi->scr, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); out_be32(&ssi->stcr, @@ -286,8 +281,6 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, out_be32(&ssi->sfcsr, CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); - spin_unlock(&ssi_info->lock); - /* * We keep the SSI disabled because if we enable it, then the * DMA controller will start. It's not supposed to start until @@ -299,6 +292,45 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ } + if (!ssi_info->master_stream) + ssi_info->master_stream = substream; + else { + /* This is the slave stream open, so we need to impose sample + * rate and sample size constraints. Note that this can cause a + * race condition if the slave stream is opened before the + * master stream is fully initialized. + * + * We provide some protection by checking to make sure the + * master stream is initialized, but it's not perfect. ALSA + * sometimes re-initializes the driver with a different sample + * rate or size. If the slave stream is opened before the + * master stream has received its final parameters, then the + * slave stream may be constrained to the wrong sample rate or + * size. + */ + struct snd_pcm_runtime *first_runtime = + ssi_info->master_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); + + ssi_info->slave_stream = substream; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ssi_info->playback = 1; else @@ -325,38 +357,20 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, static int fsl_ssi_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_ssi_info *ssi_info = cpu_dai->private_data; - struct ccsr_ssi __iomem *ssi = ssi_info->ssi; - uint32_t wl; - u32 scr; - u32 sxccr; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - - spin_lock(&ssi_info->lock); - - scr = in_be32(&ssi->scr); + if (substream == ssi_info->master_stream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct ccsr_ssi __iomem *ssi = ssi_info->ssi; + u32 wl; - /* If the SSI is currently enabled, then we want to keep it disabled for - * as short as time as possible, so we pre-calculate all the values for - * the appropriate registers, and then program them rapidly. - */ + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sxccr = (in_be32(&ssi->stccr) & ~CCSR_SSI_SxCCR_WL_MASK) | wl; - out_be32(&ssi->scr, scr & ~CCSR_SSI_SCR_SSIEN); - out_be32(&ssi->stccr, sxccr); - out_be32(&ssi->scr, scr); - } else { - sxccr = (in_be32(&ssi->srccr) & ~CCSR_SSI_SxCCR_WL_MASK) | wl; - out_be32(&ssi->scr, scr & ~CCSR_SSI_SCR_SSIEN); - out_be32(&ssi->srccr, sxccr); - out_be32(&ssi->scr, scr); + /* In synchronous mode, the SSI uses STCCR for capture */ + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); } - spin_unlock(&ssi_info->lock); - return 0; } @@ -368,6 +382,23 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream, * * The DMA channel is in external master start and pause mode, which * means the SSI completely controls the flow of data. + * + * The SSI is disabled and re-enabled to reset its connection to the DMA + * controller. The SSI communicates with the DMA controller when it is + * enabled, even if the SCR.TE and SCR.RE bits are zero. This means that + * the DMA controller must be programmed (at least partially) for playback + * *and* capture, before the SSI is enabled. + * + * This is a problem for simultaneous playback and capture, because the + * streams are not created at the same time. If playback is started first, + * then the DMA controller will not be programmed correctly for capture when + * the SSI is enabled for playback. + * + * The solution is to temporary disable the SSI when starting playback or + * capture, even if capture or playback is already running. When the SSI is + * re-enabled, the first stream will continue, so it will still be + * programmed correctly. The SSI will also tell the DMA controller to start + * operations for the second stream. */ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) @@ -380,12 +411,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - spin_lock(&ssi_info->lock); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - else { + } else { + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); @@ -396,8 +427,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, */ mdelay(1); } - - spin_unlock(&ssi_info->lock); break; case SNDRV_PCM_TRIGGER_STOP: @@ -431,6 +460,15 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, else ssi_info->capture = 0; + /* If we're closing the master stream, then the slave stream becomes + * the new master. If there is no slave stream, then slave_stream is + NULL, so this code still works. */ + if (ssi_info->master_stream == substream) + ssi_info->master_stream = ssi_info->slave_stream; + + /* Either way, there is no slave stream any more. */ + ssi_info->slave_stream = NULL; + /* * If this is the last active substream, disable the SSI and release * the IRQ. diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 91af81c..160b8bc 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -226,7 +226,8 @@ struct fsl_ssi_info { unsigned int irq; unsigned int playback; unsigned int capture; - spinlock_t lock; + struct snd_pcm_substream *master_stream; + struct snd_pcm_substream *slave_stream; struct snd_soc_dai *dai; struct device_attribute dev_attr; u32 pmuxcr; -- 1.5.5

2 2

[alsa-devel] [PATCH] ASoC: Add WM8900 CODEC driver
by Mark Brown 04 Aug '08

04 Aug '08

The WM8900 is designed for portable multimedia applications requiring low power consumption, high performance audio and a compact form factor providing: - 24 bit stereo ADC and DAC - Microphone and line inputs - Line outputs - Class G headphone amplifier Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8900.c | 1542 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8900.h | 64 ++ 4 files changed, 1612 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8900.c create mode 100644 sound/soc/codecs/wm8900.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c93423..d7bacf6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8731 select SND_SOC_WM8750 select SND_SOC_WM8753 + select SND_SOC_WM8900 select SND_SOC_WM8990 select SND_SOC_CS4270 select SND_SOC_TLV320AIC26 @@ -46,6 +47,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8900 + tristate + config SND_SOC_WM8990 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 409e4dd..98808d9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8900-objs := wm8900.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c new file mode 100644 index 0000000..0b8c6d3 --- /dev/null +++ b/sound/soc/codecs/wm8900.c @@ -0,0 +1,1542 @@ +/* + * wm8900.c -- WM8900 ALSA Soc Audio driver + * + * Copyright 2007, 2008 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + * + * TODO: + * - Tristating. + * - TDM. + * - Jack detect. + * - FLL source configuration, currently only MCLK is supported. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.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 "wm8900.h" + +/* WM8900 register space */ +#define WM8900_REG_RESET 0x0 +#define WM8900_REG_ID 0x0 +#define WM8900_REG_POWER1 0x1 +#define WM8900_REG_POWER2 0x2 +#define WM8900_REG_POWER3 0x3 +#define WM8900_REG_AUDIO1 0x4 +#define WM8900_REG_AUDIO2 0x5 +#define WM8900_REG_CLOCKING1 0x6 +#define WM8900_REG_CLOCKING2 0x7 +#define WM8900_REG_AUDIO3 0x8 +#define WM8900_REG_AUDIO4 0x9 +#define WM8900_REG_DACCTRL 0xa +#define WM8900_REG_LDAC_DV 0xb +#define WM8900_REG_RDAC_DV 0xc +#define WM8900_REG_SIDETONE 0xd +#define WM8900_REG_ADCCTRL 0xe +#define WM8900_REG_LADC_DV 0xf +#define WM8900_REG_RADC_DV 0x10 +#define WM8900_REG_GPIO 0x12 +#define WM8900_REG_INCTL 0x15 +#define WM8900_REG_LINVOL 0x16 +#define WM8900_REG_RINVOL 0x17 +#define WM8900_REG_INBOOSTMIX1 0x18 +#define WM8900_REG_INBOOSTMIX2 0x19 +#define WM8900_REG_ADCPATH 0x1a +#define WM8900_REG_AUXBOOST 0x1b +#define WM8900_REG_ADDCTL 0x1e +#define WM8900_REG_FLLCTL1 0x24 +#define WM8900_REG_FLLCTL2 0x25 +#define WM8900_REG_FLLCTL3 0x26 +#define WM8900_REG_FLLCTL4 0x27 +#define WM8900_REG_FLLCTL5 0x28 +#define WM8900_REG_FLLCTL6 0x29 +#define WM8900_REG_LOUTMIXCTL1 0x2c +#define WM8900_REG_ROUTMIXCTL1 0x2d +#define WM8900_REG_BYPASS1 0x2e +#define WM8900_REG_BYPASS2 0x2f +#define WM8900_REG_AUXOUT_CTL 0x30 +#define WM8900_REG_LOUT1CTL 0x33 +#define WM8900_REG_ROUT1CTL 0x34 +#define WM8900_REG_LOUT2CTL 0x35 +#define WM8900_REG_ROUT2CTL 0x36 +#define WM8900_REG_HPCTL1 0x3a +#define WM8900_REG_OUTBIASCTL 0x73 + +#define WM8900_MAXREG 0x80 + +#define WM8900_REG_ADDCTL_OUT1_DIS 0x80 +#define WM8900_REG_ADDCTL_OUT2_DIS 0x40 +#define WM8900_REG_ADDCTL_VMID_DIS 0x20 +#define WM8900_REG_ADDCTL_BIAS_SRC 0x10 +#define WM8900_REG_ADDCTL_VMID_SOFTST 0x04 +#define WM8900_REG_ADDCTL_TEMP_SD 0x02 + +#define WM8900_REG_GPIO_TEMP_ENA 0x2 + +#define WM8900_REG_POWER1_STARTUP_BIAS_ENA 0x0100 +#define WM8900_REG_POWER1_BIAS_ENA 0x0008 +#define WM8900_REG_POWER1_VMID_BUF_ENA 0x0004 +#define WM8900_REG_POWER1_FLL_ENA 0x0040 + +#define WM8900_REG_POWER2_SYSCLK_ENA 0x8000 +#define WM8900_REG_POWER2_ADCL_ENA 0x0002 +#define WM8900_REG_POWER2_ADCR_ENA 0x0001 + +#define WM8900_REG_POWER3_DACL_ENA 0x0002 +#define WM8900_REG_POWER3_DACR_ENA 0x0001 + +#define WM8900_REG_AUDIO1_AIF_FMT_MASK 0x0018 +#define WM8900_REG_AUDIO1_LRCLK_INV 0x0080 +#define WM8900_REG_AUDIO1_BCLK_INV 0x0100 + +#define WM8900_REG_CLOCKING1_BCLK_DIR 0x1 +#define WM8900_REG_CLOCKING1_MCLK_SRC 0x100 +#define WM8900_REG_CLOCKING1_BCLK_MASK (~0x01e) +#define WM8900_REG_CLOCKING1_OPCLK_MASK (~0x7000) + +#define WM8900_REG_CLOCKING2_ADC_CLKDIV 0xe0 +#define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c + +#define WM8900_REG_DACCTRL_MUTE 0x004 +#define WM8900_REG_DACCTRL_AIF_LRCLKRATE 0x400 + +#define WM8900_REG_AUDIO3_ADCLRC_DIR 0x0800 + +#define WM8900_REG_AUDIO4_DACLRC_DIR 0x0800 + +#define WM8900_REG_FLLCTL1_OSC_ENA 0x100 + +#define WM8900_REG_FLLCTL6_FLL_SLOW_LOCK_REF 0x100 + +#define WM8900_REG_HPCTL1_HP_IPSTAGE_ENA 0x80 +#define WM8900_REG_HPCTL1_HP_OPSTAGE_ENA 0x40 +#define WM8900_REG_HPCTL1_HP_CLAMP_IP 0x20 +#define WM8900_REG_HPCTL1_HP_CLAMP_OP 0x10 +#define WM8900_REG_HPCTL1_HP_SHORT 0x08 +#define WM8900_REG_HPCTL1_HP_SHORT2 0x04 + +#define WM8900_LRC_MASK 0xfc00 + +struct snd_soc_codec_device soc_codec_dev_wm8900; + +struct wm8900_priv { + u32 fll_in; /* FLL input frequency */ + u32 fll_out; /* FLL output frequency */ +}; + +/* + * wm8900 register cache. We can't read the entire register space and we + * have slow control buses so we cache the registers. + */ +static const u16 wm8900_reg_defaults[WM8900_MAXREG] = { + 0x8900, 0x0000, + 0xc000, 0x0000, + 0x4050, 0x4000, + 0x0008, 0x0000, + 0x0040, 0x0040, + 0x1004, 0x00c0, + 0x00c0, 0x0000, + 0x0100, 0x00c0, + 0x00c0, 0x0000, + 0xb001, 0x0000, + 0x0000, 0x0044, + 0x004c, 0x004c, + 0x0044, 0x0044, + 0x0000, 0x0044, + 0x0000, 0x0000, + 0x0002, 0x0000, + 0x0000, 0x0000, + 0x0000, 0x0000, + 0x0008, 0x0000, + 0x0000, 0x0008, + 0x0097, 0x0100, + 0x0000, 0x0000, + 0x0050, 0x0050, + 0x0055, 0x0055, + 0x0055, 0x0000, + 0x0000, 0x0079, + 0x0079, 0x0079, + 0x0079, 0x0000, + /* Remaining registers all zero */ +}; + +/* + * read wm8900 register cache + */ +static inline unsigned int wm8900_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + BUG_ON(reg >= WM8900_MAXREG); + + if (reg == WM8900_REG_ID) + return 0; + + return cache[reg]; +} + +/* + * write wm8900 register cache + */ +static inline void wm8900_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + + BUG_ON(reg >= WM8900_MAXREG); + + cache[reg] = value; +} + +/* + * write to the WM8900 register space + */ +static int wm8900_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + if (value == wm8900_read_reg_cache(codec, reg)) + return 0; + + /* data is + * D15..D9 WM8900 register offset + * D8...D0 register data + */ + data[0] = reg; + data[1] = value >> 8; + data[2] = value & 0x00ff; + + wm8900_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 3) == 3) + return 0; + else + return -EIO; +} + +/* + * Read from the wm8900. + */ +static unsigned int wm8900_chip_read(struct snd_soc_codec *codec, u8 reg) +{ + struct i2c_msg xfer[2]; + u16 data; + int ret; + struct i2c_client *client = codec->control_data; + + BUG_ON(reg != WM8900_REG_ID && reg != WM8900_REG_POWER1); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + printk(KERN_CRIT "i2c_transfer returned %d\n", ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} + +/* + * Read from the WM8900 register space. Most registers can't be read + * and are therefore supplied from cache. + */ +static unsigned int wm8900_read(struct snd_soc_codec *codec, unsigned int reg) +{ + switch (reg) { + case WM8900_REG_ID: + return wm8900_chip_read(codec, reg); + default: + return wm8900_read_reg_cache(codec, reg); + } +} + +static void wm8900_reset(struct snd_soc_codec *codec) +{ + wm8900_write(codec, WM8900_REG_RESET, 0); + + memcpy(codec->reg_cache, wm8900_reg_defaults, + sizeof(codec->reg_cache)); +} + +static int wm8900_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 hpctl1 = wm8900_read(codec, WM8900_REG_HPCTL1); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Clamp headphone outputs */ + hpctl1 = WM8900_REG_HPCTL1_HP_CLAMP_IP | + WM8900_REG_HPCTL1_HP_CLAMP_OP; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + break; + + case SND_SOC_DAPM_POST_PMU: + /* Enable the input stage */ + hpctl1 &= ~WM8900_REG_HPCTL1_HP_CLAMP_IP; + hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT | + WM8900_REG_HPCTL1_HP_SHORT2 | + WM8900_REG_HPCTL1_HP_IPSTAGE_ENA; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + + msleep(400); + + /* Enable the output stage */ + hpctl1 &= ~WM8900_REG_HPCTL1_HP_CLAMP_OP; + hpctl1 |= WM8900_REG_HPCTL1_HP_OPSTAGE_ENA; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + + /* Remove the shorts */ + hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT2; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* Short the output */ + hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + + /* Disable the output stage */ + hpctl1 &= ~WM8900_REG_HPCTL1_HP_OPSTAGE_ENA; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + + /* Clamp the outputs and power down input */ + hpctl1 |= WM8900_REG_HPCTL1_HP_CLAMP_IP | + WM8900_REG_HPCTL1_HP_CLAMP_OP; + hpctl1 &= ~WM8900_REG_HPCTL1_HP_IPSTAGE_ENA; + wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + break; + + case SND_SOC_DAPM_POST_PMD: + /* Disable everything */ + wm8900_write(codec, WM8900_REG_HPCTL1, 0); + break; + + default: + BUG(); + } + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -5700, 100, 0); + +static const DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 0); + +static const DECLARE_TLV_DB_SCALE(in_boost_tlv, -1200, 600, 0); + +static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0); + +static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1); + +static const DECLARE_TLV_DB_SCALE(adc_svol_tlv, -3600, 300, 0); + +static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); + +static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" }; + +static const struct soc_enum mic_bias_level = +SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt); + +static const char *dac_mute_rate_txt[] = { "Fast", "Slow" }; + +static const struct soc_enum dac_mute_rate = +SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt); + +static const char *dac_deemphasis_txt[] = { + "Disabled", "32kHz", "44.1kHz", "48kHz" +}; + +static const struct soc_enum dac_deemphasis = +SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt); + +static const char *adc_hpf_cut_txt[] = { + "Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3" +}; + +static const struct soc_enum adc_hpf_cut = +SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt); + +static const char *lr_txt[] = { + "Left", "Right" +}; + +static const struct soc_enum aifl_src = +SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt); + +static const struct soc_enum aifr_src = +SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt); + +static const struct soc_enum dacl_src = +SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt); + +static const struct soc_enum dacr_src = +SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt); + +static const char *sidetone_txt[] = { + "Disabled", "Left ADC", "Right ADC" +}; + +static const struct soc_enum dacl_sidetone = +SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt); + +static const struct soc_enum dacr_sidetone = +SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt); + +static const struct snd_kcontrol_new wm8900_snd_controls[] = { +SOC_ENUM("Mic Bias Level", mic_bias_level), + +SOC_SINGLE_TLV("Left Input PGA Volume", WM8900_REG_LINVOL, 0, 31, 0, + in_pga_tlv), +SOC_SINGLE("Left Input PGA Switch", WM8900_REG_LINVOL, 6, 1, 1), +SOC_SINGLE("Left Input PGA ZC Switch", WM8900_REG_LINVOL, 7, 1, 0), + +SOC_SINGLE_TLV("Right Input PGA Volume", WM8900_REG_RINVOL, 0, 31, 0, + in_pga_tlv), +SOC_SINGLE("Right Input PGA Switch", WM8900_REG_RINVOL, 6, 1, 1), +SOC_SINGLE("Right Input PGA ZC Switch", WM8900_REG_RINVOL, 7, 1, 0), + +SOC_SINGLE("DAC Soft Mute Switch", WM8900_REG_DACCTRL, 6, 1, 1), +SOC_ENUM("DAC Mute Rate", dac_mute_rate), +SOC_SINGLE("DAC Mono Switch", WM8900_REG_DACCTRL, 9, 1, 0), +SOC_ENUM("DAC Deemphasis", dac_deemphasis), +SOC_SINGLE("DAC Sloping Stopband Filter Switch", WM8900_REG_DACCTRL, 8, 1, 0), +SOC_SINGLE("DAC Sigma-Delta Modulator Clock Switch", WM8900_REG_DACCTRL, + 12, 1, 0), + +SOC_SINGLE("ADC HPF Switch", WM8900_REG_ADCCTRL, 8, 1, 0), +SOC_ENUM("ADC HPF Cut-Off", adc_hpf_cut), +SOC_DOUBLE("ADC Invert Switch", WM8900_REG_ADCCTRL, 1, 0, 1, 0), +SOC_SINGLE_TLV("Left ADC Sidetone Volume", WM8900_REG_SIDETONE, 9, 12, 0, + adc_svol_tlv), +SOC_SINGLE_TLV("Right ADC Sidetone Volume", WM8900_REG_SIDETONE, 5, 12, 0, + adc_svol_tlv), +SOC_ENUM("Left Digital Audio Source", aifl_src), +SOC_ENUM("Right Digital Audio Source", aifr_src), + +SOC_SINGLE_TLV("DAC Input Boost Volume", WM8900_REG_AUDIO2, 10, 4, 0, + dac_boost_tlv), +SOC_ENUM("Left DAC Source", dacl_src), +SOC_ENUM("Right DAC Source", dacr_src), +SOC_ENUM("Left DAC Sidetone", dacl_sidetone), +SOC_ENUM("Right DAC Sidetone", dacr_sidetone), +SOC_DOUBLE("DAC Invert Switch", WM8900_REG_DACCTRL, 1, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Digital Playback Volume", + WM8900_REG_LDAC_DV, WM8900_REG_RDAC_DV, + 1, 96, 0, dac_tlv), +SOC_DOUBLE_R_TLV("Digital Capture Volume", + WM8900_REG_LADC_DV, WM8900_REG_RADC_DV, 1, 119, 0, adc_tlv), + +SOC_SINGLE_TLV("LINPUT3 Bypass Volume", WM8900_REG_LOUTMIXCTL1, 4, 7, 0, + out_mix_tlv), +SOC_SINGLE_TLV("RINPUT3 Bypass Volume", WM8900_REG_ROUTMIXCTL1, 4, 7, 0, + out_mix_tlv), +SOC_SINGLE_TLV("Left AUX Bypass Volume", WM8900_REG_AUXOUT_CTL, 4, 7, 0, + out_mix_tlv), +SOC_SINGLE_TLV("Right AUX Bypass Volume", WM8900_REG_AUXOUT_CTL, 0, 7, 0, + out_mix_tlv), + +SOC_SINGLE_TLV("LeftIn to RightOut Mixer Volume", WM8900_REG_BYPASS1, 0, 7, 0, + out_mix_tlv), +SOC_SINGLE_TLV("LeftIn to LeftOut Mixer Volume", WM8900_REG_BYPASS1, 4, 7, 0, + out_mix_tlv), +SOC_SINGLE_TLV("RightIn to LeftOut Mixer Volume", WM8900_REG_BYPASS2, 0, 7, 0, + out_mix_tlv), +SOC_SINGLE_TLV("RightIn to RightOut Mixer Volume", WM8900_REG_BYPASS2, 4, 7, 0, + out_mix_tlv), + +SOC_SINGLE_TLV("IN2L Boost Volume", WM8900_REG_INBOOSTMIX1, 0, 3, 0, + in_boost_tlv), +SOC_SINGLE_TLV("IN3L Boost Volume", WM8900_REG_INBOOSTMIX1, 4, 3, 0, + in_boost_tlv), +SOC_SINGLE_TLV("IN2R Boost Volume", WM8900_REG_INBOOSTMIX2, 0, 3, 0, + in_boost_tlv), +SOC_SINGLE_TLV("IN3R Boost Volume", WM8900_REG_INBOOSTMIX2, 4, 3, 0, + in_boost_tlv), +SOC_SINGLE_TLV("Left AUX Boost Volume", WM8900_REG_AUXBOOST, 4, 3, 0, + in_boost_tlv), +SOC_SINGLE_TLV("Right AUX Boost Volume", WM8900_REG_AUXBOOST, 0, 3, 0, + in_boost_tlv), + +SOC_DOUBLE_R_TLV("LINEOUT1 Volume", WM8900_REG_LOUT1CTL, WM8900_REG_ROUT1CTL, + 0, 63, 0, out_pga_tlv), +SOC_DOUBLE_R("LINEOUT1 Switch", WM8900_REG_LOUT1CTL, WM8900_REG_ROUT1CTL, + 6, 1, 1), +SOC_DOUBLE_R("LINEOUT1 ZC Switch", WM8900_REG_LOUT1CTL, WM8900_REG_ROUT1CTL, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("LINEOUT2 Volume", + WM8900_REG_LOUT2CTL, WM8900_REG_ROUT2CTL, + 0, 63, 0, out_pga_tlv), +SOC_DOUBLE_R("LINEOUT2 Switch", + WM8900_REG_LOUT2CTL, WM8900_REG_ROUT2CTL, 6, 1, 1), +SOC_DOUBLE_R("LINEOUT2 ZC Switch", + WM8900_REG_LOUT2CTL, WM8900_REG_ROUT2CTL, 7, 1, 0), +SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1, + 0, 1, 1), + +}; + +/* add non dapm controls */ +static int wm8900_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8900_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static const struct snd_kcontrol_new wm8900_dapm_loutput2_control = +SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0); + +static const struct snd_kcontrol_new wm8900_dapm_routput2_control = +SOC_DAPM_SINGLE("LINEOUT2R Switch", WM8900_REG_POWER3, 5, 1, 0); + +static const struct snd_kcontrol_new wm8900_loutmix_controls[] = { +SOC_DAPM_SINGLE("LINPUT3 Bypass Switch", WM8900_REG_LOUTMIXCTL1, 7, 1, 0), +SOC_DAPM_SINGLE("AUX Bypass Switch", WM8900_REG_AUXOUT_CTL, 7, 1, 0), +SOC_DAPM_SINGLE("Left Input Mixer Switch", WM8900_REG_BYPASS1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Input Mixer Switch", WM8900_REG_BYPASS2, 3, 1, 0), +SOC_DAPM_SINGLE("DACL Switch", WM8900_REG_LOUTMIXCTL1, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8900_routmix_controls[] = { +SOC_DAPM_SINGLE("RINPUT3 Bypass Switch", WM8900_REG_ROUTMIXCTL1, 7, 1, 0), +SOC_DAPM_SINGLE("AUX Bypass Switch", WM8900_REG_AUXOUT_CTL, 3, 1, 0), +SOC_DAPM_SINGLE("Left Input Mixer Switch", WM8900_REG_BYPASS1, 3, 1, 0), +SOC_DAPM_SINGLE("Right Input Mixer Switch", WM8900_REG_BYPASS2, 7, 1, 0), +SOC_DAPM_SINGLE("DACR Switch", WM8900_REG_ROUTMIXCTL1, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8900_linmix_controls[] = { +SOC_DAPM_SINGLE("LINPUT2 Switch", WM8900_REG_INBOOSTMIX1, 2, 1, 1), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8900_REG_INBOOSTMIX1, 6, 1, 1), +SOC_DAPM_SINGLE("AUX Switch", WM8900_REG_AUXBOOST, 6, 1, 1), +SOC_DAPM_SINGLE("Input PGA Switch", WM8900_REG_ADCPATH, 6, 1, 0), +}; + +static const struct snd_kcontrol_new wm8900_rinmix_controls[] = { +SOC_DAPM_SINGLE("RINPUT2 Switch", WM8900_REG_INBOOSTMIX2, 2, 1, 1), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INBOOSTMIX2, 6, 1, 1), +SOC_DAPM_SINGLE("AUX Switch", WM8900_REG_AUXBOOST, 2, 1, 1), +SOC_DAPM_SINGLE("Input PGA Switch", WM8900_REG_ADCPATH, 2, 1, 0), +}; + +static const struct snd_kcontrol_new wm8900_linpga_controls[] = { +SOC_DAPM_SINGLE("LINPUT1 Switch", WM8900_REG_INCTL, 6, 1, 0), +SOC_DAPM_SINGLE("LINPUT2 Switch", WM8900_REG_INCTL, 5, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8900_REG_INCTL, 4, 1, 0), +}; + +static const struct snd_kcontrol_new wm8900_rinpga_controls[] = { +SOC_DAPM_SINGLE("RINPUT1 Switch", WM8900_REG_INCTL, 2, 1, 0), +SOC_DAPM_SINGLE("RINPUT2 Switch", WM8900_REG_INCTL, 1, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0), +}; + +static const char *wm9700_lp_mux[] = { "Disabled", "Enabled" }; + +static const struct soc_enum wm8900_lineout2_lp_mux = +SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm9700_lp_mux); + +static const struct snd_kcontrol_new wm8900_lineout2_lp = +SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux); + +static const struct snd_soc_dapm_widget wm8900_dapm_widgets[] = { + +/* Externally visible pins */ +SND_SOC_DAPM_OUTPUT("LINEOUT1L"), +SND_SOC_DAPM_OUTPUT("LINEOUT1R"), +SND_SOC_DAPM_OUTPUT("LINEOUT2L"), +SND_SOC_DAPM_OUTPUT("LINEOUT2R"), +SND_SOC_DAPM_OUTPUT("HP_L"), +SND_SOC_DAPM_OUTPUT("HP_R"), + +SND_SOC_DAPM_INPUT("RINPUT1"), +SND_SOC_DAPM_INPUT("LINPUT1"), +SND_SOC_DAPM_INPUT("RINPUT2"), +SND_SOC_DAPM_INPUT("LINPUT2"), +SND_SOC_DAPM_INPUT("RINPUT3"), +SND_SOC_DAPM_INPUT("LINPUT3"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_VMID("VMID"), + +/* Input */ +SND_SOC_DAPM_MIXER("Left Input PGA", WM8900_REG_POWER2, 3, 0, + wm8900_linpga_controls, + ARRAY_SIZE(wm8900_linpga_controls)), +SND_SOC_DAPM_MIXER("Right Input PGA", WM8900_REG_POWER2, 2, 0, + wm8900_rinpga_controls, + ARRAY_SIZE(wm8900_rinpga_controls)), + +SND_SOC_DAPM_MIXER("Left Input Mixer", WM8900_REG_POWER2, 5, 0, + wm8900_linmix_controls, + ARRAY_SIZE(wm8900_linmix_controls)), +SND_SOC_DAPM_MIXER("Right Input Mixer", WM8900_REG_POWER2, 4, 0, + wm8900_rinmix_controls, + ARRAY_SIZE(wm8900_rinmix_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8900_REG_POWER1, 4, 0), + +SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8900_REG_POWER2, 1, 0), +SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8900_REG_POWER2, 0, 0), + +/* Output */ +SND_SOC_DAPM_DAC("DACL", "Left HiFi Playback", WM8900_REG_POWER3, 1, 0), +SND_SOC_DAPM_DAC("DACR", "Right HiFi Playback", WM8900_REG_POWER3, 0, 0), + +SND_SOC_DAPM_PGA_E("Headphone Amplifier", WM8900_REG_POWER3, 7, 0, NULL, 0, + wm8900_hp_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_PGA("LINEOUT1L PGA", WM8900_REG_POWER2, 8, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINEOUT1R PGA", WM8900_REG_POWER2, 7, 0, NULL, 0), + +SND_SOC_DAPM_MUX("LINEOUT2 LP", SND_SOC_NOPM, 0, 0, &wm8900_lineout2_lp), +SND_SOC_DAPM_PGA("LINEOUT2L PGA", WM8900_REG_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINEOUT2R PGA", WM8900_REG_POWER3, 5, 0, NULL, 0), + +SND_SOC_DAPM_MIXER("Left Output Mixer", WM8900_REG_POWER3, 3, 0, + wm8900_loutmix_controls, + ARRAY_SIZE(wm8900_loutmix_controls)), +SND_SOC_DAPM_MIXER("Right Output Mixer", WM8900_REG_POWER3, 2, 0, + wm8900_routmix_controls, + ARRAY_SIZE(wm8900_routmix_controls)), +}; + +/* Target, Path, Source */ +static const struct snd_soc_dapm_route audio_map[] = { +/* Inputs */ +{"Left Input PGA", "LINPUT1 Switch", "LINPUT1"}, +{"Left Input PGA", "LINPUT2 Switch", "LINPUT2"}, +{"Left Input PGA", "LINPUT3 Switch", "LINPUT3"}, + +{"Right Input PGA", "RINPUT1 Switch", "RINPUT1"}, +{"Right Input PGA", "RINPUT2 Switch", "RINPUT2"}, +{"Right Input PGA", "RINPUT3 Switch", "RINPUT3"}, + +{"Left Input Mixer", "LINPUT2 Switch", "LINPUT2"}, +{"Left Input Mixer", "LINPUT3 Switch", "LINPUT3"}, +{"Left Input Mixer", "AUX Switch", "AUX"}, +{"Left Input Mixer", "Input PGA Switch", "Left Input PGA"}, + +{"Right Input Mixer", "RINPUT2 Switch", "RINPUT2"}, +{"Right Input Mixer", "RINPUT3 Switch", "RINPUT3"}, +{"Right Input Mixer", "AUX Switch", "AUX"}, +{"Right Input Mixer", "Input PGA Switch", "Right Input PGA"}, + +{"ADCL", NULL, "Left Input Mixer"}, +{"ADCR", NULL, "Right Input Mixer"}, + +/* Outputs */ +{"LINEOUT1L", NULL, "LINEOUT1L PGA"}, +{"LINEOUT1L PGA", NULL, "Left Output Mixer"}, +{"LINEOUT1R", NULL, "LINEOUT1R PGA"}, +{"LINEOUT1R PGA", NULL, "Right Output Mixer"}, + +{"LINEOUT2L PGA", NULL, "Left Output Mixer"}, +{"LINEOUT2 LP", "Disabled", "LINEOUT2L PGA"}, +{"LINEOUT2 LP", "Enabled", "Left Output Mixer"}, +{"LINEOUT2L", NULL, "LINEOUT2 LP"}, + +{"LINEOUT2R PGA", NULL, "Right Output Mixer"}, +{"LINEOUT2 LP", "Disabled", "LINEOUT2R PGA"}, +{"LINEOUT2 LP", "Enabled", "Right Output Mixer"}, +{"LINEOUT2R", NULL, "LINEOUT2 LP"}, + +{"Left Output Mixer", "LINPUT3 Bypass Switch", "LINPUT3"}, +{"Left Output Mixer", "AUX Bypass Switch", "AUX"}, +{"Left Output Mixer", "Left Input Mixer Switch", "Left Input Mixer"}, +{"Left Output Mixer", "Right Input Mixer Switch", "Right Input Mixer"}, +{"Left Output Mixer", "DACL Switch", "DACL"}, + +{"Right Output Mixer", "RINPUT3 Bypass Switch", "RINPUT3"}, +{"Right Output Mixer", "AUX Bypass Switch", "AUX"}, +{"Right Output Mixer", "Left Input Mixer Switch", "Left Input Mixer"}, +{"Right Output Mixer", "Right Input Mixer Switch", "Right Input Mixer"}, +{"Right Output Mixer", "DACR Switch", "DACR"}, + +/* Note that the headphone output stage needs to be connected + * externally to LINEOUT2 via DC blocking capacitors. Other + * configurations are not supported. + * + * Note also that left and right headphone paths are treated as a + * mono path. + */ +{"Headphone Amplifier", NULL, "LINEOUT2 LP"}, +{"Headphone Amplifier", NULL, "LINEOUT2 LP"}, +{"HP_L", NULL, "Headphone Amplifier"}, +{"HP_R", NULL, "Headphone Amplifier"}, +}; + +static int wm8900_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8900_dapm_widgets, + ARRAY_SIZE(wm8900_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int wm8900_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 reg; + + reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 0x60; + break; + default: + return -EINVAL; + } + + wm8900_write(codec, WM8900_REG_AUDIO1, reg); + + return 0; +} + +/* FLL divisors */ +struct _fll_div { + u16 fll_ratio; + u16 fllclk_div; + u16 fll_slow_lock_ref; + u16 n; + u16 k; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + unsigned int div; + + BUG_ON(!Fout); + + /* The FLL must run at 90-100MHz which is then scaled down to + * the output value by FLLCLK_DIV. */ + target = Fout; + div = 1; + while (target < 90000000) { + div *= 2; + target *= 2; + } + + if (target > 100000000) + printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d" + " Fout=%d\n", target, Fref, Fout); + if (div > 32) { + printk(KERN_ERR "wm8900: Invalid FLL division rate %u, " + "Fref=%d, Fout=%d, target=%d\n", + div, Fref, Fout, target); + return -EINVAL; + } + + fll_div->fllclk_div = div >> 2; + + if (Fref < 48000) + fll_div->fll_slow_lock_ref = 1; + else + fll_div->fll_slow_lock_ref = 0; + + Ndiv = target / Fref; + + if (Fref < 1000000) + fll_div->fll_ratio = 8; + else + fll_div->fll_ratio = 1; + + fll_div->n = Ndiv / fll_div->fll_ratio; + Nmod = (target / fll_div->fll_ratio) % Fref; + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll_div->k = K / 10; + + BUG_ON(target != Fout * (fll_div->fllclk_div << 2)); + BUG_ON(!K && target != Fref * fll_div->fll_ratio * fll_div->n); + + return 0; +} + +static int wm8900_set_fll(struct snd_soc_codec *codec, + int fll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct wm8900_priv *wm8900 = codec->private_data; + struct _fll_div fll_div; + unsigned int reg; + + if (wm8900->fll_in == freq_in && wm8900->fll_out == freq_out) + return 0; + + /* The digital side should be disabled during any change. */ + reg = wm8900_read(codec, WM8900_REG_POWER1); + wm8900_write(codec, WM8900_REG_POWER1, + reg & (~WM8900_REG_POWER1_FLL_ENA)); + + /* Disable the FLL? */ + if (!freq_in || !freq_out) { + reg = wm8900_read(codec, WM8900_REG_CLOCKING1); + wm8900_write(codec, WM8900_REG_CLOCKING1, + reg & (~WM8900_REG_CLOCKING1_MCLK_SRC)); + + reg = wm8900_read(codec, WM8900_REG_FLLCTL1); + wm8900_write(codec, WM8900_REG_FLLCTL1, + reg & (~WM8900_REG_FLLCTL1_OSC_ENA)); + + wm8900->fll_in = freq_in; + wm8900->fll_out = freq_out; + + return 0; + } + + if (fll_factors(&fll_div, freq_in, freq_out) != 0) + goto reenable; + + wm8900->fll_in = freq_in; + wm8900->fll_out = freq_out; + + /* The osclilator *MUST* be enabled before we enable the + * digital circuit. */ + wm8900_write(codec, WM8900_REG_FLLCTL1, + fll_div.fll_ratio | WM8900_REG_FLLCTL1_OSC_ENA); + + wm8900_write(codec, WM8900_REG_FLLCTL4, fll_div.n >> 5); + wm8900_write(codec, WM8900_REG_FLLCTL5, + (fll_div.fllclk_div << 6) | (fll_div.n & 0x1f)); + + if (fll_div.k) { + wm8900_write(codec, WM8900_REG_FLLCTL2, + (fll_div.k >> 8) | 0x100); + wm8900_write(codec, WM8900_REG_FLLCTL3, fll_div.k & 0xff); + } else + wm8900_write(codec, WM8900_REG_FLLCTL2, 0); + + if (fll_div.fll_slow_lock_ref) + wm8900_write(codec, WM8900_REG_FLLCTL6, + WM8900_REG_FLLCTL6_FLL_SLOW_LOCK_REF); + else + wm8900_write(codec, WM8900_REG_FLLCTL6, 0); + + reg = wm8900_read(codec, WM8900_REG_POWER1); + wm8900_write(codec, WM8900_REG_POWER1, + reg | WM8900_REG_POWER1_FLL_ENA); + +reenable: + reg = wm8900_read(codec, WM8900_REG_CLOCKING1); + wm8900_write(codec, WM8900_REG_CLOCKING1, + reg | WM8900_REG_CLOCKING1_MCLK_SRC); + + return 0; +} + +static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out); +} + +static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg; + + switch (div_id) { + case WM8900_BCLK_DIV: + reg = wm8900_read(codec, WM8900_REG_CLOCKING1); + wm8900_write(codec, WM8900_REG_CLOCKING1, + div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK)); + break; + case WM8900_OPCLK_DIV: + reg = wm8900_read(codec, WM8900_REG_CLOCKING1); + wm8900_write(codec, WM8900_REG_CLOCKING1, + div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK)); + break; + case WM8900_DAC_LRCLK: + reg = wm8900_read(codec, WM8900_REG_AUDIO4); + wm8900_write(codec, WM8900_REG_AUDIO4, + div | (reg & WM8900_LRC_MASK)); + break; + case WM8900_ADC_LRCLK: + reg = wm8900_read(codec, WM8900_REG_AUDIO3); + wm8900_write(codec, WM8900_REG_AUDIO3, + div | (reg & WM8900_LRC_MASK)); + break; + case WM8900_DAC_CLKDIV: + reg = wm8900_read(codec, WM8900_REG_CLOCKING2); + wm8900_write(codec, WM8900_REG_CLOCKING2, + div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV)); + break; + case WM8900_ADC_CLKDIV: + reg = wm8900_read(codec, WM8900_REG_CLOCKING2); + wm8900_write(codec, WM8900_REG_CLOCKING2, + div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV)); + break; + case WM8900_LRCLK_MODE: + reg = wm8900_read(codec, WM8900_REG_DACCTRL); + wm8900_write(codec, WM8900_REG_DACCTRL, + div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE)); + break; + default: + return -EINVAL; + } + + return 0; +} + + +static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int clocking1, aif1, aif3, aif4; + + clocking1 = wm8900_read(codec, WM8900_REG_CLOCKING1); + aif1 = wm8900_read(codec, WM8900_REG_AUDIO1); + aif3 = wm8900_read(codec, WM8900_REG_AUDIO3); + aif4 = wm8900_read(codec, WM8900_REG_AUDIO4); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + clocking1 &= ~WM8900_REG_CLOCKING1_BCLK_DIR; + aif3 &= ~WM8900_REG_AUDIO3_ADCLRC_DIR; + aif4 &= ~WM8900_REG_AUDIO4_DACLRC_DIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + clocking1 &= ~WM8900_REG_CLOCKING1_BCLK_DIR; + aif3 |= WM8900_REG_AUDIO3_ADCLRC_DIR; + aif4 |= WM8900_REG_AUDIO4_DACLRC_DIR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + clocking1 |= WM8900_REG_CLOCKING1_BCLK_DIR; + aif3 |= WM8900_REG_AUDIO3_ADCLRC_DIR; + aif4 |= WM8900_REG_AUDIO4_DACLRC_DIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + clocking1 |= WM8900_REG_CLOCKING1_BCLK_DIR; + aif3 &= ~WM8900_REG_AUDIO3_ADCLRC_DIR; + aif4 &= ~WM8900_REG_AUDIO4_DACLRC_DIR; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + aif1 |= WM8900_REG_AUDIO1_AIF_FMT_MASK; + aif1 &= ~WM8900_REG_AUDIO1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + aif1 |= WM8900_REG_AUDIO1_AIF_FMT_MASK; + aif1 |= WM8900_REG_AUDIO1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_I2S: + aif1 &= ~WM8900_REG_AUDIO1_AIF_FMT_MASK; + aif1 |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif1 &= ~WM8900_REG_AUDIO1_AIF_FMT_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif1 &= ~WM8900_REG_AUDIO1_AIF_FMT_MASK; + aif1 |= 0x8; + break; + default: + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + aif1 &= ~WM8900_REG_AUDIO1_BCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8900_REG_AUDIO1_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + aif1 &= ~WM8900_REG_AUDIO1_BCLK_INV; + aif1 &= ~WM8900_REG_AUDIO1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + aif1 |= WM8900_REG_AUDIO1_BCLK_INV; + aif1 |= WM8900_REG_AUDIO1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8900_REG_AUDIO1_BCLK_INV; + aif1 &= ~WM8900_REG_AUDIO1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif1 &= ~WM8900_REG_AUDIO1_BCLK_INV; + aif1 |= WM8900_REG_AUDIO1_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + wm8900_write(codec, WM8900_REG_CLOCKING1, clocking1); + wm8900_write(codec, WM8900_REG_AUDIO1, aif1); + wm8900_write(codec, WM8900_REG_AUDIO3, aif3); + wm8900_write(codec, WM8900_REG_AUDIO4, aif4); + + return 0; +} + +static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + reg = wm8900_read(codec, WM8900_REG_DACCTRL); + + if (mute) + reg |= WM8900_REG_DACCTRL_MUTE; + else + reg &= ~WM8900_REG_DACCTRL_MUTE; + + wm8900_write(codec, WM8900_REG_DACCTRL, reg); + + return 0; +} + +#define WM8900_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8900_PCM_FORMATS \ + (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_dai wm8900_dai = { + .name = "WM8900 HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8900_RATES, + .formats = WM8900_PCM_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8900_RATES, + .formats = WM8900_PCM_FORMATS, + }, + .ops = { + .hw_params = wm8900_hw_params, + }, + .dai_ops = { + .set_clkdiv = wm8900_set_dai_clkdiv, + .set_pll = wm8900_set_dai_pll, + .set_fmt = wm8900_set_dai_fmt, + .digital_mute = wm8900_digital_mute, + }, +}; +EXPORT_SYMBOL_GPL(wm8900_dai); + +static int wm8900_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + /* Enable thermal shutdown */ + reg = wm8900_read(codec, WM8900_REG_GPIO); + wm8900_write(codec, WM8900_REG_GPIO, + reg | WM8900_REG_GPIO_TEMP_ENA); + reg = wm8900_read(codec, WM8900_REG_ADDCTL); + wm8900_write(codec, WM8900_REG_ADDCTL, + reg | WM8900_REG_ADDCTL_TEMP_SD); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + /* Charge capacitors if initial power up */ + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* STARTUP_BIAS_ENA on */ + wm8900_write(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_STARTUP_BIAS_ENA); + + /* Startup bias mode */ + wm8900_write(codec, WM8900_REG_ADDCTL, + WM8900_REG_ADDCTL_BIAS_SRC | + WM8900_REG_ADDCTL_VMID_SOFTST); + + /* VMID 2x50k */ + wm8900_write(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_STARTUP_BIAS_ENA | 0x1); + + /* Allow capacitors to charge */ + schedule_timeout_interruptible(msecs_to_jiffies(400)); + + /* Enable bias */ + wm8900_write(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_STARTUP_BIAS_ENA | + WM8900_REG_POWER1_BIAS_ENA | 0x1); + + wm8900_write(codec, WM8900_REG_ADDCTL, 0); + + wm8900_write(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_BIAS_ENA | 0x1); + } + + reg = wm8900_read(codec, WM8900_REG_POWER1); + wm8900_write(codec, WM8900_REG_POWER1, + (reg & WM8900_REG_POWER1_FLL_ENA) | + WM8900_REG_POWER1_BIAS_ENA | 0x1); + wm8900_write(codec, WM8900_REG_POWER2, + WM8900_REG_POWER2_SYSCLK_ENA); + wm8900_write(codec, WM8900_REG_POWER3, 0); + break; + + case SND_SOC_BIAS_OFF: + /* Startup bias enable */ + reg = wm8900_read(codec, WM8900_REG_POWER1); + wm8900_write(codec, WM8900_REG_POWER1, + reg & WM8900_REG_POWER1_STARTUP_BIAS_ENA); + wm8900_write(codec, WM8900_REG_ADDCTL, + WM8900_REG_ADDCTL_BIAS_SRC | + WM8900_REG_ADDCTL_VMID_SOFTST); + + /* Discharge caps */ + wm8900_write(codec, WM8900_REG_POWER1, + WM8900_REG_POWER1_STARTUP_BIAS_ENA); + schedule_timeout_interruptible(msecs_to_jiffies(500)); + + /* Remove clamp */ + wm8900_write(codec, WM8900_REG_HPCTL1, 0); + + /* Power down */ + wm8900_write(codec, WM8900_REG_ADDCTL, 0); + wm8900_write(codec, WM8900_REG_POWER1, 0); + wm8900_write(codec, WM8900_REG_POWER2, 0); + wm8900_write(codec, WM8900_REG_POWER3, 0); + + /* Need to let things settle before stopping the clock + * to ensure that restart works, see "Stopping the + * master clock" in the datasheet. */ + schedule_timeout_interruptible(msecs_to_jiffies(1)); + wm8900_write(codec, WM8900_REG_POWER2, + WM8900_REG_POWER2_SYSCLK_ENA); + break; + } + codec->bias_level = level; + return 0; +} + +static int wm8900_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm8900_priv *wm8900 = codec->private_data; + int fll_out = wm8900->fll_out; + int fll_in = wm8900->fll_in; + int ret; + + /* Stop the FLL in an orderly fashion */ + ret = wm8900_set_fll(codec, 0, 0, 0); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to stop FLL\n"); + return ret; + } + + wm8900->fll_out = fll_out; + wm8900->fll_in = fll_in; + + wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8900_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm8900_priv *wm8900 = codec->private_data; + u16 *cache; + int i, ret; + + cache = kmemdup(codec->reg_cache, sizeof(wm8900_reg_defaults), + GFP_KERNEL); + + wm8900_reset(codec); + wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restart the FLL? */ + if (wm8900->fll_out) { + int fll_out = wm8900->fll_out; + int fll_in = wm8900->fll_in; + + wm8900->fll_in = 0; + wm8900->fll_out = 0; + + ret = wm8900_set_fll(codec, 0, fll_in, fll_out); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to restart FLL\n"); + return ret; + } + } + + if (cache) { + for (i = 0; i < WM8900_MAXREG; i++) + wm8900_write(codec, i, cache[i]); + kfree(cache); + } else + dev_err(&pdev->dev, "Unable to allocate register cache\n"); + + return 0; +} + +/* + * initialise the WM8900 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8900_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + unsigned int reg; + struct i2c_client *i2c_client = socdev->codec->control_data; + + codec->name = "WM8900"; + codec->owner = THIS_MODULE; + codec->read = wm8900_read; + codec->write = wm8900_write; + codec->dai = &wm8900_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8900_MAXREG; + codec->reg_cache = kmemdup(wm8900_reg_defaults, + sizeof(wm8900_reg_defaults), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + reg = wm8900_read(codec, WM8900_REG_ID); + if (reg != 0x8900) { + dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n", + reg); + return -ENODEV; + } + + codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + if (codec->private_data == NULL) { + ret = -ENOMEM; + goto priv_err; + } + + /* Read back from the chip */ + reg = wm8900_chip_read(codec, WM8900_REG_POWER1); + reg = (reg >> 12) & 0xf; + dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg); + + wm8900_reset(codec); + + /* Latch the volume update bits */ + wm8900_write(codec, WM8900_REG_LINVOL, + wm8900_read(codec, WM8900_REG_LINVOL) | 0x100); + wm8900_write(codec, WM8900_REG_RINVOL, + wm8900_read(codec, WM8900_REG_RINVOL) | 0x100); + wm8900_write(codec, WM8900_REG_LOUT1CTL, + wm8900_read(codec, WM8900_REG_LOUT1CTL) | 0x100); + wm8900_write(codec, WM8900_REG_ROUT1CTL, + wm8900_read(codec, WM8900_REG_ROUT1CTL) | 0x100); + wm8900_write(codec, WM8900_REG_LOUT2CTL, + wm8900_read(codec, WM8900_REG_LOUT2CTL) | 0x100); + wm8900_write(codec, WM8900_REG_ROUT2CTL, + wm8900_read(codec, WM8900_REG_ROUT2CTL) | 0x100); + wm8900_write(codec, WM8900_REG_LDAC_DV, + wm8900_read(codec, WM8900_REG_LDAC_DV) | 0x100); + wm8900_write(codec, WM8900_REG_RDAC_DV, + wm8900_read(codec, WM8900_REG_RDAC_DV) | 0x100); + wm8900_write(codec, WM8900_REG_LADC_DV, + wm8900_read(codec, WM8900_REG_LADC_DV) | 0x100); + wm8900_write(codec, WM8900_REG_RADC_DV, + wm8900_read(codec, WM8900_REG_RADC_DV) | 0x100); + + /* Set the DAC and mixer output bias */ + wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81); + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to register new PCMs\n"); + goto pcm_err; + } + + /* Turn the chip on */ + codec->bias_level = SND_SOC_BIAS_OFF; + wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8900_add_controls(codec); + wm8900_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +priv_err: + kfree(codec->private_data); + return ret; +} + +static struct snd_soc_device *wm8900_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8900_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8900_socdev; + struct wm8900_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + dev_err(&adap->dev, "Probe on %x\n", addr); + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + dev_err(&adap->dev, + "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8900_init(socdev); + if (ret < 0) { + dev_err(&adap->dev, "failed to initialise WM8900\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8900_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8900_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8900_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8900_i2c_driver = { + .driver = { + .name = "WM8900 I2C codec", + .owner = THIS_MODULE, + }, + .attach_adapter = wm8900_i2c_attach, + .detach_client = wm8900_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8900", + .driver = &wm8900_i2c_driver, +}; +#endif + +static int wm8900_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8900_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + dev_info(&pdev->dev, "WM8900 Audio Codec\n"); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + socdev->codec = codec; + + codec->set_bias_level = wm8900_set_bias_level; + + wm8900_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8900_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else +#error Non-I2C interfaces not yet supported +#endif + return ret; +} + +/* power down chip */ +static int wm8900_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8900_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8900 = { + .probe = wm8900_probe, + .remove = wm8900_remove, + .suspend = wm8900_suspend, + .resume = wm8900_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900); + +MODULE_DESCRIPTION("ASoC WM8900 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h new file mode 100644 index 0000000..ba450d9 --- /dev/null +++ b/sound/soc/codecs/wm8900.h @@ -0,0 +1,64 @@ +/* + * wm8900.h -- WM890 Soc Audio driver + * + * 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 _WM8900_H +#define _WM8900_H + +#define WM8900_FLL 1 + +#define WM8900_BCLK_DIV 1 +#define WM8900_ADC_CLKDIV 2 +#define WM8900_DAC_CLKDIV 3 +#define WM8900_ADC_LRCLK 4 +#define WM8900_DAC_LRCLK 5 +#define WM8900_OPCLK_DIV 6 +#define WM8900_LRCLK_MODE 7 + +#define WM8900_BCLK_DIV_1 0x00 +#define WM8900_BCLK_DIV_1_5 0x02 +#define WM8900_BCLK_DIV_2 0x04 +#define WM8900_BCLK_DIV_3 0x06 +#define WM8900_BCLK_DIV_4 0x08 +#define WM8900_BCLK_DIV_5_5 0x0a +#define WM8900_BCLK_DIV_6 0x0c +#define WM8900_BCLK_DIV_8 0x0e +#define WM8900_BCLK_DIV_11 0x10 +#define WM8900_BCLK_DIV_12 0x12 +#define WM8900_BCLK_DIV_16 0x14 +#define WM8900_BCLK_DIV_22 0x16 +#define WM8900_BCLK_DIV_24 0x18 +#define WM8900_BCLK_DIV_32 0x1a +#define WM8900_BCLK_DIV_44 0x1c +#define WM8900_BCLK_DIV_48 0x1e + +#define WM8900_ADC_CLKDIV_1 0x00 +#define WM8900_ADC_CLKDIV_1_5 0x20 +#define WM8900_ADC_CLKDIV_2 0x40 +#define WM8900_ADC_CLKDIV_3 0x60 +#define WM8900_ADC_CLKDIV_4 0x80 +#define WM8900_ADC_CLKDIV_5_5 0xa0 +#define WM8900_ADC_CLKDIV_6 0xc0 + +#define WM8900_DAC_CLKDIV_1 0x00 +#define WM8900_DAC_CLKDIV_1_5 0x04 +#define WM8900_DAC_CLKDIV_2 0x08 +#define WM8900_DAC_CLKDIV_3 0x0c +#define WM8900_DAC_CLKDIV_4 0x10 +#define WM8900_DAC_CLKDIV_5_5 0x14 +#define WM8900_DAC_CLKDIV_6 0x18 + +#define WM8900_ + +struct wm8900_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8900_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8900; + +#endif -- 1.5.6.3

2 1

[alsa-devel] [PATCH] wss_lib: remove second mutexes initialization
by Krzysztof Helt 04 Aug '08

04 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove initializations of spinlock and mutexes which are done earlier in snd_wss_new(). The snd_wss_new() is called from snd_wss_create(). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-2.6.26/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-31 20:39:59.000000000 +0200 +++ linux-2.6.26/sound/isa/wss/wss_lib.c 2008-08-03 16:54:01.293267081 +0200 @@ -1866,10 +1866,6 @@ int snd_wss_pcm(struct snd_wss *chip, in if (err < 0) return err; - spin_lock_init(&chip->reg_lock); - mutex_init(&chip->mce_mutex); - mutex_init(&chip->open_mutex); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); ---------------------------------------------------------------------- Tylko dla detektywow! Konkurs na Smaker.pl Kliknij >>> http://link.interia.pl/f1eb1

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[alsa-devel] [PATCH] ALSA: wss_lib: opti92x-ad1848 WSS_HW_DETECT fix
by Rene Herman 04 Aug '08

04 Aug '08

I wrote: > Firstly, snd-opti92x-ad1848 doesn't work for anything anymore. The > chip is each time detected as "OPTi 93x" instead of the actual AD/CS > chip. Probably minor buglet -- I did not look, just mechanically > switched cards and tested. The attached seems to fix things. Rene.

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[alsa-devel] [PATCH] ALSA: snd-cs4236: add Techmakers MF-4236PW PnP card ID
by Rene Herman 04 Aug '08

04 Aug '08

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[alsa-devel] Multichannel sound distortions with Realtek ALC889a
by Matthias Dahl 04 Aug '08

04 Aug '08

Hello. Earlier this month I already posted about my problems with the ALC889a chip. I've gathered some more informations that might help or inform others: 1) the ALC889a is a custom job for Gigabyte and Realtek is not allowed to release its data sheet to anyone but Gigabyte. So no data sheet on this one. 2) There seem to be two revisions of this chip. I was unable to confirm this with Realtek but my revision id is 0x100101 whereas ALSA expects 0x100103. Modifing patch_realtek.c appropriately to recognize my revision id and use patch_alc883 instead of patch_alc882, doesn't fix the problem but seems to help my initial volume problems. 3) Sound distortions only appear on the center channel while playing back multi channel content. They sound like voltage sizzling noise. Hard to explain. Muting all channels but the center channel makes those distortions disappear. A test under Vista revealed no distortions or anything like it. So a faulty chip is actually pretty unlikely. Since three weeks I am in contact with Realtek about this problem but there is not much to report on that front yet except for a rather friendly and helpful contact. Unfortunately due to time constraints and other issues, the problem hasn't been confirmed nor denied yet by Realtek. They received a similar mb from Gigabyte and are planing on trying to reproduce that problem there. To make a long story short, I'd appreciate if anyone with an ALC889a chip and a 5.1 setup could try to confirm this bug and post about it. If anyone knows a way how I could diagnose this further, I am open to suggestions... Currently I tend to get myself an Asus Xonar D2X or similar and keep this as a reminder that one should never buy hardware that's not in some way or another open. :-( To defend myself, when I bought my board a few weeks back I was under the impression that _all_ HDA codecs work well with ALSA and at least there I wouldn't have to worry. Well... far from it. :( Thanks a lot in advance for any suggestions or help. Have a nice day, Matthias Dahl

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[alsa-devel] [PATCH 10/10] wss_lib: use wss detection code instead of ad1848 one
by Krzysztof Helt 04 Aug '08

04 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss detection code and kill the ad1848 library. The library is fully assimilated into the new wss library. This patch fixes also the issue with AD1848 codec MCE timeout - a waiting loop is now 10 times longer. I have tested it on following cards: Gallant SC-6600 (codec: AD1848, driver: snd-sc6600] SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape] SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy] Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x] Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x] Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236] Sound playback and recording works on all these cards. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the last patch which completely removes the ad1848_lib.c and the ad1848.h header. The detection code was modified comparing to the original ad1848 code to correctly recognize pre cs4231 chips (16 codec registers) and post cs4231 chips (32 codec registers). One issue for Rene: Opti93x cards supports mu/a-law playback but no recording. If these formats are selected only noise is recorded (I suspect ADPCM and big endian formats cannot be recorded as well). The mu-law recording works ok on CS4231 and AD1845 chips. I tested only on them to check if it is a hardware or driver problem. Some wss code improvement patches may follow. Regards, Krzysztof diff -urpN linux-alsa/include/sound/ad1848.h linux-mm/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-18 07:51:49.694754585 +0200 +++ linux-mm/include/sound/ad1848.h 1970-01-01 01:00:00.000000000 +0100 @@ -1,111 +0,0 @@ -#ifndef __SOUND_AD1848_H -#define __SOUND_AD1848_H - -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Definitions for AD1847/AD1848/CS4248 chips - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include "pcm.h" -#include <linux/interrupt.h> - -/* codec registers */ - -#define AD1848_LEFT_INPUT 0x00 /* left input control */ -#define AD1848_RIGHT_INPUT 0x01 /* right input control */ -#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ -#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ -#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ -#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define AD1848_PIN_CTRL 0x0a /* pin control */ -#define AD1848_TEST_INIT 0x0b /* test and initialization */ -#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ -#define AD1848_LOOPBACK 0x0d /* loopback control */ -#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ -#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define AD1848_INIT 0x80 /* CODEC is initializing */ -#define AD1848_MCE 0x40 /* mode change enable */ -#define AD1848_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ - -#define AD1848_ENABLE_MIC_GAIN 0x20 - -#define AD1848_MIXS_LINE1 0x00 -#define AD1848_MIXS_AUX1 0x40 -#define AD1848_MIXS_LINE2 0x80 -#define AD1848_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ - -#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ -#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ -#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define AD1848_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define AD1848_XTAL1 0x00 /* 24.576 crystal */ -#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - AD1848_IFACE_CTRL */ - -#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ -#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ -#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ -#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ -#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ -#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - AD1848_PIN_CTRL */ - -#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ -#define AD1848_XCTL1 0x40 /* external control #1 */ -#define AD1848_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - AD1848_TEST_INIT */ - -#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ - -#include "wss.h" - -/* exported functions */ - -void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, - unsigned char value); - -int snd_ad1848_create(struct snd_card *card, - unsigned long port, - int irq, int dma, - unsigned short hardware, - struct snd_wss **chip); - -#endif /* __SOUND_AD1848_H */ diff -urpN linux-alsa/sound/isa/Kconfig linux-mm/sound/isa/Kconfig --- linux-alsa/sound/isa/Kconfig 2008-07-18 07:51:49.750768622 +0200 +++ linux-mm/sound/isa/Kconfig 2008-07-18 13:34:54.698303387 +0200 @@ -4,11 +4,6 @@ config SND_WSS_LIB tristate select SND_PCM -config SND_AD1848_LIB - tristate - select SND_PCM - select SND_WSS_LIB - config SND_SB_COMMON tristate @@ -56,7 +51,7 @@ config SND_AD1816A config SND_AD1848 tristate "Generic AD1848/CS4248 driver" - select SND_AD1848_LIB + select SND_WSS_LIB help Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. @@ -97,7 +92,7 @@ config SND_AZT2320 config SND_CMI8330 tristate "C-Media CMI8330" - select SND_AD1848_LIB + select SND_WSS_LIB select SND_SB16_DSP help Say Y here to include support for soundcards based on the @@ -193,7 +188,7 @@ config SND_ES18XX config SND_SC6000 tristate "Gallant SC-6000, Audio Excel DSP 16" depends on HAS_IOPORT - select SND_AD1848_LIB + select SND_WSS_LIB select SND_OPL3_LIB select SND_MPU401_UART help @@ -280,7 +275,7 @@ config SND_OPTI92X_AD1848 select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART - select SND_AD1848_LIB + select SND_WSS_LIB help Say Y here to include support for soundcards based on Opti 82C92x or OTI-601 chips and using an AD1848 codec. @@ -382,7 +377,7 @@ config SND_SB16_CSP_FIRMWARE_IN_KERNEL config SND_SGALAXY tristate "Aztech Sound Galaxy" - select SND_AD1848_LIB + select SND_WSS_LIB help Say Y here to include support for Aztech Sound Galaxy soundcards. diff -urpN linux-alsa/sound/isa/ad1848/Makefile linux-mm/sound/isa/ad1848/Makefile --- linux-alsa/sound/isa/ad1848/Makefile 2008-04-17 04:49:44.000000000 +0200 +++ linux-mm/sound/isa/ad1848/Makefile 2008-07-18 13:34:54.698303387 +0200 @@ -3,10 +3,8 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex(a)perex.cz> # -snd-ad1848-lib-objs := ad1848_lib.o snd-ad1848-objs := ad1848.o # Toplevel Module Dependency obj-$(CONFIG_SND_AD1848) += snd-ad1848.o -obj-$(CONFIG_SND_AD1848_LIB) += snd-ad1848-lib.o diff -urpN linux-alsa/sound/isa/ad1848/ad1848.c linux-mm/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-07-18 07:51:49.786753998 +0200 +++ linux-mm/sound/isa/ad1848/ad1848.c 2008-07-18 13:34:54.702304924 +0200 @@ -28,7 +28,7 @@ #include <linux/wait.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/initval.h> #define CRD_NAME "Generic AD1848/AD1847/CS4248" @@ -95,8 +95,9 @@ static int __devinit snd_ad1848_probe(st if (!card) return -EINVAL; - error = snd_ad1848_create(card, port[n], irq[n], dma1[n], - thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, &chip); + error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], -1, + thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, + 0, &chip); if (error < 0) goto out; diff -urpN linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-mm/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-18 07:51:49.802793124 +0200 +++ linux-mm/sound/isa/ad1848/ad1848_lib.c 1970-01-01 01:00:00.000000000 +0100 @@ -1,505 +0,0 @@ -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Routines for control of AD1848/AD1847/CS4248 - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#define SNDRV_MAIN_OBJECT_FILE -#include <linux/delay.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/slab.h> -#include <linux/ioport.h> -#include <sound/core.h> -#include <sound/ad1848.h> -#include <sound/control.h> -#include <sound/tlv.h> -#include <sound/pcm_params.h> - -#include <asm/io.h> -#include <asm/dma.h> - -MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); -MODULE_DESCRIPTION("Routines for control of AD1848/AD1847/CS4248"); -MODULE_LICENSE("GPL"); - -#if 0 -#define SNDRV_DEBUG_MCE -#endif - -/* - * Some variables - */ - -static unsigned char snd_ad1848_original_image[16] = -{ - 0x00, /* 00 - lic */ - 0x00, /* 01 - ric */ - 0x9f, /* 02 - la1ic */ - 0x9f, /* 03 - ra1ic */ - 0x9f, /* 04 - la2ic */ - 0x9f, /* 05 - ra2ic */ - 0xbf, /* 06 - loc */ - 0xbf, /* 07 - roc */ - 0x20, /* 08 - dfr */ - AD1848_AUTOCALIB, /* 09 - ic */ - 0x00, /* 0a - pc */ - 0x00, /* 0b - ti */ - 0x00, /* 0c - mi */ - 0x00, /* 0d - lbc */ - 0x00, /* 0e - dru */ - 0x00, /* 0f - drl */ -}; - -/* - * Basic I/O functions - */ - -static void snd_ad1848_wait(struct snd_wss *chip) -{ - int timeout; - - for (timeout = 250; timeout > 0; timeout--) { - if ((inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) == 0) - break; - udelay(100); - } -} - -void snd_ad1848_out(struct snd_wss *chip, - unsigned char reg, - unsigned char value) -{ - snd_ad1848_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - " - "reg = 0x%x, value = 0x%x\n", reg, value); -#endif - outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); - outb(chip->image[reg] = value, chip->port + CS4231P(REG)); - mb(); - snd_printdd("codec out - reg 0x%x = 0x%x\n", - chip->mce_bit | reg, value); -} - -EXPORT_SYMBOL(snd_ad1848_out); - -static unsigned char snd_ad1848_in(struct snd_wss *chip, unsigned char reg) -{ - snd_ad1848_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - " - "reg = 0x%x\n", reg); -#endif - outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); - mb(); - return inb(chip->port + CS4231P(REG)); -} - -#if 0 - -static void snd_ad1848_debug(struct snd_wss *chip) -{ - printk(KERN_DEBUG "AD1848 REGS: INDEX = 0x%02x ", - inb(chip->port + CS4231P(REGSEL))); - printk(KERN_DEBUG " STATUS = 0x%02x\n", - inb(chip->port + CS4231P(STATUS))); - printk(KERN_DEBUG " 0x00: left input = 0x%02x ", - snd_ad1848_in(chip, 0x00)); - printk(KERN_DEBUG " 0x08: playback format = 0x%02x\n", - snd_ad1848_in(chip, 0x08)); - printk(KERN_DEBUG " 0x01: right input = 0x%02x ", - snd_ad1848_in(chip, 0x01)); - printk(KERN_DEBUG " 0x09: iface (CFIG 1) = 0x%02x\n", - snd_ad1848_in(chip, 0x09)); - printk(KERN_DEBUG " 0x02: AUXA left = 0x%02x ", - snd_ad1848_in(chip, 0x02)); - printk(KERN_DEBUG " 0x0a: pin control = 0x%02x\n", - snd_ad1848_in(chip, 0x0a)); - printk(KERN_DEBUG " 0x03: AUXA right = 0x%02x ", - snd_ad1848_in(chip, 0x03)); - printk(KERN_DEBUG " 0x0b: init & status = 0x%02x\n", - snd_ad1848_in(chip, 0x0b)); - printk(KERN_DEBUG " 0x04: AUXB left = 0x%02x ", - snd_ad1848_in(chip, 0x04)); - printk(KERN_DEBUG " 0x0c: revision & mode = 0x%02x\n", - snd_ad1848_in(chip, 0x0c)); - printk(KERN_DEBUG " 0x05: AUXB right = 0x%02x ", - snd_ad1848_in(chip, 0x05)); - printk(KERN_DEBUG " 0x0d: loopback = 0x%02x\n", - snd_ad1848_in(chip, 0x0d)); - printk(KERN_DEBUG " 0x06: left output = 0x%02x ", - snd_ad1848_in(chip, 0x06)); - printk(KERN_DEBUG " 0x0e: data upr count = 0x%02x\n", - snd_ad1848_in(chip, 0x0e)); - printk(KERN_DEBUG " 0x07: right output = 0x%02x ", - snd_ad1848_in(chip, 0x07)); - printk(KERN_DEBUG " 0x0f: data lwr count = 0x%02x\n", - snd_ad1848_in(chip, 0x0f)); -} - -#endif - -/* - * AD1848 detection / MCE routines - */ - -static void snd_ad1848_mce_up(struct snd_wss *chip) -{ - unsigned long flags; - int timeout; - - snd_ad1848_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); -#endif - spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit |= AD1848_MCE; - timeout = inb(chip->port + CS4231P(REGSEL)); - if (timeout == 0x80) - snd_printk(KERN_WARNING "mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); - if (!(timeout & AD1848_MCE)) - outb(chip->mce_bit | (timeout & 0x1f), - chip->port + CS4231P(REGSEL)); - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static void snd_ad1848_mce_down(struct snd_wss *chip) -{ - unsigned long flags, timeout; - int reg; - - spin_lock_irqsave(&chip->reg_lock, flags); - for (timeout = 5; timeout > 0; timeout--) - inb(chip->port + CS4231P(REGSEL)); - /* end of cleanup sequence */ - for (timeout = 12000; - timeout > 0 && (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT); - timeout--) - udelay(100); - - snd_printdd("(1) timeout = %ld\n", timeout); - -#ifdef CONFIG_SND_DEBUG - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING - "mce_down [0x%lx] - auto calibration time out (0)\n", - chip->port + CS4231P(REGSEL)); -#endif - - chip->mce_bit &= ~AD1848_MCE; - reg = inb(chip->port + CS4231P(REGSEL)); - outb(chip->mce_bit | (reg & 0x1f), chip->port + CS4231P(REGSEL)); - if (reg == 0x80) - snd_printk(KERN_WARNING "mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((reg & AD1848_MCE) == 0) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } - - /* - * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low. - * It may take up to 5 sample periods (at most 907 us @ 5.5125 kHz) for - * the process to _start_, so it is important to wait at least that long - * before checking. Otherwise we might think AC has finished when it - * has in fact not begun. It could take 128 (no AC) or 384 (AC) cycles - * for ACI to drop. This gives a wait of at most 70 ms with a more - * typical value of 3-9 ms. - */ - timeout = jiffies + msecs_to_jiffies(250); - do { - spin_unlock_irqrestore(&chip->reg_lock, flags); - msleep(1); - spin_lock_irqsave(&chip->reg_lock, flags); - reg = snd_ad1848_in(chip, AD1848_TEST_INIT) & - AD1848_CALIB_IN_PROGRESS; - } while (reg && time_before(jiffies, timeout)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (reg) - snd_printk(KERN_ERR - "mce_down - auto calibration time out (2)\n"); - - snd_printdd("(4) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", - inb(chip->port + CS4231P(REGSEL))); -} - -static irqreturn_t snd_ad1848_interrupt(int irq, void *dev_id) -{ - struct snd_wss *chip = dev_id; - - if ((chip->mode & WSS_MODE_PLAY) && chip->playback_substream) - snd_pcm_period_elapsed(chip->playback_substream); - if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) - snd_pcm_period_elapsed(chip->capture_substream); - outb(0, chip->port + CS4231P(STATUS)); /* clear global interrupt bit */ - return IRQ_HANDLED; -} - -/* - - */ - -static void snd_ad1848_thinkpad_twiddle(struct snd_wss *chip, int on) -{ - int tmp; - - if (!chip->thinkpad_flag) return; - - outb(0x1c, AD1848_THINKPAD_CTL_PORT1); - tmp = inb(AD1848_THINKPAD_CTL_PORT2); - - if (on) - /* turn it on */ - tmp |= AD1848_THINKPAD_CS4248_ENABLE_BIT; - else - /* turn it off */ - tmp &= ~AD1848_THINKPAD_CS4248_ENABLE_BIT; - - outb(tmp, AD1848_THINKPAD_CTL_PORT2); - -} - -#ifdef CONFIG_PM -static void snd_ad1848_suspend(struct snd_wss *chip) -{ - snd_pcm_suspend_all(chip->pcm); - if (chip->thinkpad_flag) - snd_ad1848_thinkpad_twiddle(chip, 0); -} - -static void snd_ad1848_resume(struct snd_wss *chip) -{ - int i; - - if (chip->thinkpad_flag) - snd_ad1848_thinkpad_twiddle(chip, 1); - - /* clear any pendings IRQ */ - inb(chip->port + CS4231P(STATUS)); - outb(0, chip->port + CS4231P(STATUS)); - mb(); - - snd_ad1848_mce_down(chip); - for (i = 0; i < 16; i++) - snd_ad1848_out(chip, i, chip->image[i]); - snd_ad1848_mce_up(chip); - snd_ad1848_mce_down(chip); -} -#endif /* CONFIG_PM */ - -static int snd_ad1848_probe(struct snd_wss *chip) -{ - unsigned long flags; - int i, id, rev, ad1847; - unsigned char *ptr; - -#if 0 - snd_ad1848_debug(chip); -#endif - id = ad1847 = 0; - for (i = 0; i < 1000; i++) { - mb(); - if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) - udelay(500); - else { - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_MISC_INFO, 0x00); - snd_ad1848_out(chip, AD1848_LEFT_INPUT, 0xaa); - snd_ad1848_out(chip, AD1848_RIGHT_INPUT, 0x45); - rev = snd_ad1848_in(chip, AD1848_RIGHT_INPUT); - if (rev == 0x65) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - ad1847 = 1; - break; - } - if (snd_ad1848_in(chip, AD1848_LEFT_INPUT) == 0xaa && rev == 0x45) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - id = 1; - break; - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - } - } - if (id != 1) - return -ENODEV; /* no valid device found */ - if (chip->hardware == WSS_HW_DETECT) { - if (ad1847) { - chip->hardware = WSS_HW_AD1847; - } else { - chip->hardware = WSS_HW_AD1848; - rev = snd_ad1848_in(chip, AD1848_MISC_INFO); - if (rev & 0x80) { - chip->hardware = WSS_HW_CS4248; - } else if ((rev & 0x0f) == 0x0a) { - snd_ad1848_out(chip, AD1848_MISC_INFO, 0x40); - for (i = 0; i < 16; ++i) { - if (snd_ad1848_in(chip, i) != snd_ad1848_in(chip, i + 16)) { - chip->hardware = WSS_HW_CMI8330; - break; - } - } - snd_ad1848_out(chip, AD1848_MISC_INFO, 0x00); - } - } - } - spin_lock_irqsave(&chip->reg_lock, flags); - inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ - outb(0, chip->port + CS4231P(STATUS)); - mb(); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->image[AD1848_MISC_INFO] = 0x00; - chip->image[AD1848_IFACE_CTRL] = - (chip->image[AD1848_IFACE_CTRL] & ~AD1848_SINGLE_DMA) | AD1848_SINGLE_DMA; - ptr = (unsigned char *) &chip->image; - snd_ad1848_mce_down(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - for (i = 0; i < 16; i++) /* ok.. fill all AD1848 registers */ - snd_ad1848_out(chip, i, *ptr++); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_up(chip); - /* init needed for WSS pcm */ - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | - AD1848_PLAYBACK_PIO | - AD1848_CAPTURE_ENABLE | - AD1848_CAPTURE_PIO | - AD1848_CALIB_MODE); - chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - return 0; /* all things are ok.. */ -} - -/* - - */ - -static int snd_ad1848_free(struct snd_wss *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->irq >= 0) - free_irq(chip->irq, (void *) chip); - if (chip->dma1 >= 0) { - snd_dma_disable(chip->dma1); - free_dma(chip->dma1); - } - kfree(chip); - return 0; -} - -static int snd_ad1848_dev_free(struct snd_device *device) -{ - struct snd_wss *chip = device->device_data; - return snd_ad1848_free(chip); -} - -int snd_ad1848_create(struct snd_card *card, - unsigned long port, - int irq, int dma, - unsigned short hardware, - struct snd_wss **rchip) -{ - static struct snd_device_ops ops = { - .dev_free = snd_ad1848_dev_free, - }; - struct snd_wss *chip; - int err; - - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - spin_lock_init(&chip->reg_lock); - chip->card = card; - chip->port = port; - chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->single_dma = 1; - chip->hardware = hardware; - memcpy(&chip->image, &snd_ad1848_original_image, sizeof(snd_ad1848_original_image)); - - if ((chip->res_port = request_region(port, 4, "AD1848")) == NULL) { - snd_printk(KERN_ERR "ad1848: can't grab port 0x%lx\n", port); - snd_ad1848_free(chip); - return -EBUSY; - } - if (request_irq(irq, snd_ad1848_interrupt, IRQF_DISABLED, "AD1848", (void *) chip)) { - snd_printk(KERN_ERR "ad1848: can't grab IRQ %d\n", irq); - snd_ad1848_free(chip); - return -EBUSY; - } - chip->irq = irq; - if (request_dma(dma, "AD1848")) { - snd_printk(KERN_ERR "ad1848: can't grab DMA %d\n", dma); - snd_ad1848_free(chip); - return -EBUSY; - } - chip->dma1 = dma; - chip->dma2 = dma; - - if (hardware == WSS_HW_THINKPAD) { - chip->thinkpad_flag = 1; - chip->hardware = WSS_HW_DETECT; /* reset */ - snd_ad1848_thinkpad_twiddle(chip, 1); - } - - if (snd_ad1848_probe(chip) < 0) { - snd_ad1848_free(chip); - return -ENODEV; - } - - /* Register device */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_ad1848_free(chip); - return err; - } - -#ifdef CONFIG_PM - chip->suspend = snd_ad1848_suspend; - chip->resume = snd_ad1848_resume; -#endif - - *rchip = chip; - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_create); - -/* - * INIT part - */ - -static int __init alsa_ad1848_init(void) -{ - return 0; -} - -static void __exit alsa_ad1848_exit(void) -{ -} - -module_init(alsa_ad1848_init) -module_exit(alsa_ad1848_exit) diff -urpN linux-alsa/sound/isa/sc6000.c linux-mm/sound/isa/sc6000.c --- linux-alsa/sound/isa/sc6000.c 2008-07-18 07:51:49.886753130 +0200 +++ linux-mm/sound/isa/sc6000.c 2008-07-18 13:34:54.738307314 +0200 @@ -29,7 +29,7 @@ #include <linux/io.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/opl3.h> #include <sound/mpu401.h> #include <sound/control.h> @@ -548,8 +548,8 @@ static int __devinit snd_sc6000_probe(st if (err < 0) goto err_unmap2; - err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, - WSS_HW_DETECT, &chip); + err = snd_wss_create(card, mss_port[dev] + 4, -1, xirq, xdma, -1, + WSS_HW_DETECT, 0, &chip); if (err < 0) goto err_unmap2; card->private_data = chip; diff -urpN linux-alsa/sound/isa/sgalaxy.c linux-mm/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-07-18 07:51:49.894756141 +0200 +++ linux-mm/sound/isa/sgalaxy.c 2008-07-18 13:34:54.742304370 +0200 @@ -31,7 +31,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/sb.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/control.h> #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA @@ -265,9 +265,10 @@ static int __devinit snd_sgalaxy_probe(s if ((err = snd_sgalaxy_detect(dev, xirq, xdma1)) < 0) goto _err; - if ((err = snd_ad1848_create(card, wssport[dev] + 4, - xirq, xdma1, - WSS_HW_DETECT, &chip)) < 0) + err = snd_wss_create(card, wssport[dev] + 4, -1, + xirq, xdma1, -1, + WSS_HW_DETECT, 0, &chip); + if (err < 0) goto _err; card->private_data = chip; @@ -329,8 +330,8 @@ static int snd_sgalaxy_resume(struct dev struct snd_wss *chip = card->private_data; chip->resume(chip); - snd_ad1848_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); - snd_ad1848_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); + snd_wss_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); + snd_wss_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff -urp linux-alsa/sound/isa/cmi8330.c linux-mm/sound/isa/cmi8330.c --- linux-alsa/sound/isa/cmi8330.c 2008-07-18 07:51:49.850762517 +0200 +++ linux-mm/sound/isa/cmi8330.c 2008-07-18 16:39:28.461170191 +0200 @@ -50,7 +50,7 @@ #include <linux/pnp.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/ad1848.h> +#include <sound/wss.h> #include <sound/sb.h> #include <sound/initval.h> @@ -178,9 +178,9 @@ static struct snd_kcontrol_new snd_cmi83 WSS_DOUBLE("Master Playback Volume", 0, CMI8330_MASTVOL, CMI8330_MASTVOL, 4, 0, 15, 0), WSS_SINGLE("Loud Playback Switch", 0, CMI8330_MUTEMUX, 6, 1, 1), -WSS_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, +WSS_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), WSS_DOUBLE("Line Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, 4, 3, 1, 0), @@ -480,12 +480,11 @@ static int __devinit snd_cmi8330_probe(s int i, err; acard = card->private_data; - if ((err = snd_ad1848_create(card, - wssport[dev] + 4, - wssirq[dev], - wssdma[dev], - WSS_HW_DETECT, - &acard->wss)) < 0) { + err = snd_wss_create(card, wssport[dev] + 4, -1, + wssirq[dev], + wssdma[dev], -1, + WSS_HW_DETECT, 0, &acard->wss); + if (err < 0) { snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } @@ -508,9 +507,10 @@ static int __devinit snd_cmi8330_probe(s return err; } - snd_ad1848_out(acard->wss, AD1848_MISC_INFO, 0x40); /* switch on MODE2 */ + snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */ for (i = CMI8330_RMUX3D; i <= CMI8330_CDINGAIN; i++) - snd_ad1848_out(acard->wss, i, snd_cmi8330_image[i - CMI8330_RMUX3D]); + snd_wss_out(acard->wss, i, + snd_cmi8330_image[i - CMI8330_RMUX3D]); if ((err = snd_cmi8330_mixer(card, acard)) < 0) { snd_printk(KERN_ERR PFX "failed to create mixers\n"); diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-18 07:51:49.990751242 +0200 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-07-18 18:34:34.992298787 +0200 @@ -363,7 +363,7 @@ static void snd_wss_busy_wait(struct snd for (timeout = 5; timeout > 0; timeout--) wss_inb(chip, CS4231P(REGSEL)); /* end of cleanup sequence */ - for (timeout = 2500; + for (timeout = 25000; timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(10); @@ -1050,7 +1050,11 @@ irqreturn_t snd_wss_interrupt(int irq, v struct snd_wss *chip = dev_id; unsigned char status; - status = snd_wss_in(chip, CS4231_IRQ_STATUS); + if (chip->hardware & WSS_HW_AD1848_MASK) + /* pretend it was the only possible irq for AD1848 */ + status = CS4231_PLAYBACK_IRQ; + else + status = snd_wss_in(chip, CS4231_IRQ_STATUS); if (status & CS4231_TIMER_IRQ) { if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); @@ -1082,7 +1086,11 @@ irqreturn_t snd_wss_interrupt(int irq, v } spin_lock(&chip->reg_lock); - snd_wss_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0); + status = ~CS4231_ALL_IRQS | ~status; + if (chip->hardware & WSS_HW_AD1848_MASK) + wss_outb(chip, CS4231P(STATUS), 0); + else + snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } @@ -1116,36 +1124,112 @@ static snd_pcm_uframes_t snd_wss_capture */ -static int snd_wss_probe(struct snd_wss *chip) +static int snd_ad1848_probe(struct snd_wss *chip) { unsigned long flags; - int i, id, rev; - unsigned char *ptr; - unsigned int hw; + int i, id, rev, ad1847; -#if 0 - snd_wss_debug(chip); -#endif id = 0; - for (i = 0; i < 50; i++) { + ad1847 = 0; + for (i = 0; i < 1000; i++) { mb(); - if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - udelay(2000); + if (inb(chip->port + CS4231P(REGSEL)) & CS4231_INIT) + msleep(1); else { spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - id = snd_wss_in(chip, CS4231_MISC_INFO) & 0x0f; + snd_wss_out(chip, CS4231_MISC_INFO, 0x00); + snd_wss_out(chip, CS4231_LEFT_INPUT, 0xaa); + snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x45); + rev = snd_wss_in(chip, CS4231_RIGHT_INPUT); + if (rev == 0x65) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + id = 1; + ad1847 = 1; + break; + } + if (snd_wss_in(chip, CS4231_LEFT_INPUT) == 0xaa && + rev == 0x45) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + id = 1; + break; + } spin_unlock_irqrestore(&chip->reg_lock, flags); - if (id == 0x0a) - break; /* this is valid value */ } } - snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id); - if (id != 0x0a) + if (id != 1) return -ENODEV; /* no valid device found */ + id = 0; + if (chip->hardware == WSS_HW_DETECT) + id = ad1847 ? WSS_HW_AD1847 : WSS_HW_AD1848; + + spin_lock_irqsave(&chip->reg_lock, flags); + inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ + outb(0, chip->port + CS4231P(STATUS)); + mb(); + if (id == WSS_HW_AD1848) { + /* check if there are more than 16 registers */ + rev = snd_wss_in(chip, CS4231_MISC_INFO); + snd_wss_out(chip, CS4231_MISC_INFO, 0x40); + for (i = 0; i < 16; ++i) { + if (snd_wss_in(chip, i) != snd_wss_in(chip, i + 16)) { + id = WSS_HW_CMI8330; + break; + } + } + snd_wss_out(chip, CS4231_MISC_INFO, 0x00); + if (id != WSS_HW_CMI8330 && (rev & 0x80)) + id = WSS_HW_CS4248; + if (id == WSS_HW_CMI8330 && (rev & 0x0f) != 0x0a) + id = 0; + } + if (id == WSS_HW_CMI8330) { + /* verify it is not CS4231 by changing the version register */ + /* on CMI8330 it is volume control register and can be set 0 */ + snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + snd_wss_dout(chip, CS4231_VERSION, 0x00); + rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; + if (rev) + id = 0; + snd_wss_out(chip, CS4231_MISC_INFO, 0); + } + if (id) + chip->hardware = id; + + spin_unlock_irqrestore(&chip->reg_lock, flags); + return 0; /* all things are ok.. */ +} + +static int snd_wss_probe(struct snd_wss *chip) +{ + unsigned long flags; + int i, id, rev, regnum; + unsigned char *ptr; + unsigned int hw; + + id = snd_ad1848_probe(chip); + if (id < 0) + return id; hw = chip->hardware; if ((hw & WSS_HW_TYPE_MASK) == WSS_HW_DETECT) { + for (i = 0; i < 50; i++) { + mb(); + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + msleep(2); + else { + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_MISC_INFO, + CS4231_MODE2); + id = snd_wss_in(chip, CS4231_MISC_INFO) & 0x0f; + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (id == 0x0a) + break; /* this is valid value */ + } + } + snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id); + if (id != 0x0a) + return -ENODEV; /* no valid device found */ + rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev); if (rev == 0x80) { @@ -1176,7 +1260,8 @@ static int snd_wss_probe(struct snd_wss mb(); spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->image[CS4231_MISC_INFO] = CS4231_MODE2; + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + chip->image[CS4231_MISC_INFO] = CS4231_MODE2; switch (chip->hardware) { case WSS_HW_INTERWAVE: chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3; @@ -1202,9 +1287,10 @@ static int snd_wss_probe(struct snd_wss chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } ptr = (unsigned char *) &chip->image; + regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32; snd_wss_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); - for (i = 0; i < 32; i++) /* ok.. fill all CS4231 registers */ + for (i = 0; i < regnum; i++) /* ok.. fill all registers */ snd_wss_out(chip, i, *ptr++); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_up(chip); @@ -1614,6 +1700,10 @@ static int snd_wss_new(struct snd_card * else memcpy(&chip->image, &snd_wss_original_image, sizeof(snd_wss_original_image)); + if (chip->hardware & WSS_HW_AD1848_MASK) { + chip->image[CS4231_PIN_CTRL] = 0; + chip->image[CS4231_TEST_INIT] = 0; + } *rchip = chip; return 0; @@ -1641,7 +1731,8 @@ int snd_wss_create(struct snd_card *card chip->dma1 = -1; chip->dma2 = -1; - if ((chip->res_port = request_region(port, 4, "CS4231")) == NULL) { + chip->res_port = request_region(port, 4, "WSS"); + if (chip->res_port == NULL) { snd_printk(KERN_ERR "wss: can't grab port 0x%lx\n", port); snd_wss_free(chip); return -EBUSY; @@ -1656,20 +1747,20 @@ int snd_wss_create(struct snd_card *card chip->cport = cport; if (!(hwshare & WSS_HWSHARE_IRQ)) if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED, - "CS4231", (void *) chip)) { + "WSS", (void *) chip)) { snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); snd_wss_free(chip); return -EBUSY; } chip->irq = irq; - if (!(hwshare & WSS_HWSHARE_DMA1) && request_dma(dma1, "CS4231 - 1")) { + if (!(hwshare & WSS_HWSHARE_DMA1) && request_dma(dma1, "WSS - 1")) { snd_printk(KERN_ERR "wss: can't grab DMA1 %d\n", dma1); snd_wss_free(chip); return -EBUSY; } chip->dma1 = dma1; if (!(hwshare & WSS_HWSHARE_DMA2) && dma1 != dma2 && - dma2 >= 0 && request_dma(dma2, "CS4231 - 2")) { + dma2 >= 0 && request_dma(dma2, "WSS - 2")) { snd_printk(KERN_ERR "wss: can't grab DMA2 %d\n", dma2); snd_wss_free(chip); return -EBUSY; @@ -1680,6 +1771,12 @@ int snd_wss_create(struct snd_card *card } else chip->dma2 = dma2; + if (hardware == WSS_HW_THINKPAD) { + chip->thinkpad_flag = 1; + chip->hardware = WSS_HW_DETECT; /* reset */ + snd_wss_thinkpad_twiddle(chip, 1); + } + /* global setup */ if (snd_wss_probe(chip) < 0) { snd_wss_free(chip); @@ -1749,12 +1846,6 @@ int snd_wss_pcm(struct snd_wss *chip, in snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); - /* temporary */ - if (chip->hardware & WSS_HW_AD1848_MASK) { - chip->rate_constraint = snd_wss_xrate; - chip->set_playback_format = snd_wss_playback_format; - chip->set_capture_format = snd_wss_capture_format; - } /* global setup */ pcm->private_data = chip; pcm->info_flags = 0; diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-mm/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-18 07:51:49.874759721 +0200 +++ linux-mm/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-18 16:54:21.131152444 +0200 @@ -33,11 +33,7 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#if defined(CS4231) || defined(OPTi93X) #include <sound/wss.h> -#else -#include <sound/ad1848.h> -#endif /* CS4231 */ #include <sound/mpu401.h> #include <sound/opl3.h> #ifndef OPTi93X @@ -53,16 +49,10 @@ MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("OPTi93X"); MODULE_SUPPORTED_DEVICE("{{OPTi,82C931/3}}"); #else /* OPTi93X */ -#ifdef CS4231 -MODULE_DESCRIPTION("OPTi92X - CS4231"); -MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (CS4231)}," - "{OPTi,82C925 (CS4231)}}"); -#else /* CS4231 */ -MODULE_DESCRIPTION("OPTi92X - AD1848"); -MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)}," - "{OPTi,82C925 (AD1848)}," +MODULE_DESCRIPTION("OPTi92X - WSS"); +MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (WSS)}," + "{OPTi,82C925 (WSS)}}," "{OAK,Mozart}}"); -#endif /* CS4231 */ #endif /* OPTi93X */ static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ @@ -144,9 +134,7 @@ struct snd_opti9xx { long wss_base; int irq; int dma1; -#if defined(CS4231) || defined(OPTi93X) int dma2; -#endif /* CS4231 || OPTi93X */ long fm_port; @@ -221,9 +209,7 @@ static int __devinit snd_opti9xx_init(st chip->wss_base = -1; chip->irq = -1; chip->dma1 = -1; -#if defined(CS4231) || defined (OPTi93X) chip->dma2 = -1; -#endif /* CS4231 || OPTi93X */ chip->fm_port = -1; chip->mpu_port = -1; chip->mpu_irq = -1; @@ -688,7 +674,7 @@ static void snd_card_opti9xx_free(struct if (chip) { #ifdef OPTi93X struct snd_wss *codec = chip->codec; - if (codec->irq > 0) { + if (codec && codec->irq > 0) { disable_irq(codec->irq); free_irq(codec->irq, codec); } @@ -736,7 +722,6 @@ static int __devinit snd_opti9xx_probe(s if (error) return error; -#if defined(CS4231) || defined(OPTi93X) error = snd_wss_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, #ifdef CS4231 @@ -750,12 +735,6 @@ static int __devinit snd_opti9xx_probe(s #ifdef OPTi93X chip->codec = codec; #endif -#else - error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, - chip->dma1, WSS_HW_DETECT, &codec); - if (error < 0) - return error; -#endif error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; ---------------------------------------------------------------------- Zobacz cala prawde o Lukaszu Podolskim! kliknij >>> http://link.interia.pl/f1e57

3 20

[PATCH] ALSA: wss_lib: opti92x-ad1848 WSS_HW_DETECT fix
by Rene Herman 04 Aug '08

04 Aug '08

snd-opti92x-ad1848 mistakingly passes WSS_HW_OPTI93X currently. This fixes it as tested with a OPTi 82C929A/AD1848 card. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/opti9xx/opti92x-ad1848.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index cb5f66b..19706b0 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -719,6 +719,8 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) chip->dma2 = dma2; +#else + chip->dma2 = -1; #endif if (chip->wss_base == SNDRV_AUTO_PORT) { @@ -734,10 +736,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) error = snd_wss_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, -#ifdef CS4231 - WSS_HW_DETECT, 0, -#else /* OPTi93x */ +#ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, +#else + WSS_HW_DETECT, 0, #endif &codec); if (error < 0) -- 1.5.5 --------------040800040305000006090808--

1 0

[PATCH] ALSA: snd-cs4236: add Techmakers MF-4236PW PnP card ID
by Rene Herman 04 Aug '08

04 Aug '08

Add the Techmakers MF-4236PW (Crystal CX4236B-XQ3) PnP card ID. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/cs423x/cs4236.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index dbe63db..860e90d 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -239,6 +239,8 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSC9836", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Gallant SC-70P */ { .id = "CSC9837", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, + /* Techmakers MF-4236PW */ + { .id = "CSCa736", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* TerraTec AudioSystem EWS64XL - CS4236B */ { .id = "CSCa836", .devs = { { "CSCa800" }, { "CSCa810" }, { "CSCa803" } } }, /* TerraTec AudioSystem EWS64XL - CS4236B */ -- 1.5.5

1 0

[alsa-devel] [PATCH] wss_lib: remove open_mutex
by Krzysztof Helt 03 Aug '08

03 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Open and close operations are mutexed in higher layer so remove open_mutex inside the driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-alsa/include/sound/wss.h linux-mm/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-07-31 20:39:54.000000000 +0200 +++ linux-mm/include/sound/wss.h 2008-08-02 07:42:41.256261329 +0200 @@ -110,7 +110,6 @@ struct snd_wss { spinlock_t reg_lock; struct mutex mce_mutex; - struct mutex open_mutex; int (*rate_constraint) (struct snd_pcm_runtime *runtime); void (*set_playback_format) (struct snd_wss *chip, diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-31 20:39:59.000000000 +0200 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-08-02 07:42:41.352275508 +0200 @@ -837,15 +837,11 @@ static int snd_wss_open(struct snd_wss * { unsigned long flags; - mutex_lock(&chip->open_mutex); if ((chip->mode & mode) || - ((chip->mode & WSS_MODE_OPEN) && chip->single_dma)) { - mutex_unlock(&chip->open_mutex); + ((chip->mode & WSS_MODE_OPEN) && chip->single_dma)) return -EAGAIN; - } if (chip->mode & WSS_MODE_OPEN) { chip->mode |= mode; - mutex_unlock(&chip->open_mutex); return 0; } /* ok. now enable and ack CODEC IRQ */ @@ -871,7 +867,6 @@ static int snd_wss_open(struct snd_wss * spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; - mutex_unlock(&chip->open_mutex); return 0; } @@ -879,10 +874,8 @@ static void snd_wss_close(struct snd_wss { unsigned long flags; - mutex_lock(&chip->open_mutex); chip->mode &= ~mode; if (chip->mode & WSS_MODE_OPEN) { - mutex_unlock(&chip->open_mutex); return; } snd_wss_calibrate_mute(chip, 1); @@ -922,7 +915,6 @@ static void snd_wss_close(struct snd_wss snd_wss_calibrate_mute(chip, 0); chip->mode = 0; - mutex_unlock(&chip->open_mutex); } /* @@ -1712,7 +1704,6 @@ static int snd_wss_new(struct snd_card * spin_lock_init(&chip->reg_lock); mutex_init(&chip->mce_mutex); - mutex_init(&chip->open_mutex); chip->card = card; chip->rate_constraint = snd_wss_xrate; chip->set_playback_format = snd_wss_playback_format; @@ -1868,7 +1859,6 @@ int snd_wss_pcm(struct snd_wss *chip, in spin_lock_init(&chip->reg_lock); mutex_init(&chip->mce_mutex); - mutex_init(&chip->open_mutex); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); ---------------------------------------------------------------------- Najciekawsze miejsca w Polsce i na swiecie! Zobacz >>> http://link.interia.pl/f1e60

2 2

[alsa-devel] ca0106 SPDIF Output Problems
by Ben Stanley 03 Aug '08

03 Aug '08

Hi, I am having some problems getting sound out of a ca0106 card using SPDIF. I have enabled SPDIF output using alsamixer. Some programs produce sound, and others stubbornly don't. I need some hints about what to try next, what information I could look at, to be able to diagnose some factor common to the failures, and hopefully the cause. I am familiar with part of the driver, I have already written a patch which used to enable 44.1kHz output on an earlier Ubuntu. Anyway, the details are all recorded below. I would appreciate it if a patient person could have a look through and throw me some ideas. Thanks, Ben Stanley. I did a git checkout yesterday and built alsa 1.0.17rc3 on Ubuntu 8.04 kernel 2.6.24-19-generic. I used the following configure options: ./configure --with-debug=full --with-cards=ca0106 --with-moddir=/lib/modules/`uname -r`/alsa-driver-orig (I am using the --with-moddir to avoid clobbering the distro alsa modules. I am selecting which modules to use by editing /etc/depmod.conf and changing the priority order, and then re-running depmod. I can verify the result by checking /var/lib/`uname -r`/modules.dep .) Everything builds and installs without any warnings or errors. So, after all that, I tested three applications: MythTV xine speaker-test My sound output is via SPDIF to a Yamaha RX-V557 digital amplifier. I only get sound out of MythTV. I get nothing out of xine (playing flacs) or out of speaker-test. However, I do get signal out of xine playing DVDs. Details: ================================================================ speaker-test --device-hw:0,0 --channels=2 --rate=48000 This should work, but instead it produces no SPDIF output. The amplifier shows no digital signals. root@mythtv:/proc/asound/card0/pcm0p/sub0# cat hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 4096 buffer_size: 16384 ================================================================ Testing with xine playing a 44100Hz stereo S16_LE flac: root@mythtv:/proc/asound/card0/pcm0p/sub0# cat hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 1024 buffer_size: 8192 The amplifier says 'Unknown Digital', NO SOUND PRODUCED. ================================================================Testing with xine playing 'Riverdance' DVD in DTS surround with AC3 Passthrough: Sound output WORKS Amplifier says 48kHz DTS root@mythtv:/proc/asound/card0/pcm0p/sub0# cat hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 2048 buffer_size: 16384 ================================================================ Running MythTV: Configured to open ALSA:hw:0,0 directly. root@mythtv:/proc/asound/card0/pcm0p/sub0# cat hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 8192 buffer_size: 16384 The amplifier identifies this as a 48kHz PCM signal. Sound output WORKS. ================================================================ As a general observation, this problem was present in Ubuntu 7.10, but was 'intermittent' (I used to be able to get speaker-test and xine playing flacs to work most of the time.) The failures now seem to be permanent after upgrading to Ubuntu 8.04, whether latest alsa-git is installed or not. Now, I need some ideas for what to look for here. I can't see differentiating factor between things that work and things that do not. Perhaps I need to look at some registers. I don't know what relationship the files at /proc/asound/CA0106/ca0106_reg* have to the registers declared in alsa-kmirror/pcm/ca0106/ca0106.h but I think I understand the registers from the header file. Ideas please? Once I get this sorted out I can test my patch allowing 44100Hz sampling rates to be output to SPDIF. At the moment this bug is preventing me from testing... Thanks, Ben Stanley.

1 1

[alsa-devel] dmix in a forked process
by Bankim Bhavsar 03 Aug '08

03 Aug '08

Sorry for spaming this list but I've no clue why the message is getting truncated. Anyways I had posted the same question on alsa-user mailing list --> http://tinyurl.com/6b959z -Bankim.

1 0

[alsa-devel] dmix in a forked process
by Bankim Bhavsar 03 Aug '08

03 Aug '08

Reading the source of alsa-lib I see that dmix runs in a forked process *without* exec. file: src/pcm/pcm_direct.c function: snd_pcm_direct_server_create ret = fork(); if (ret < 0) { close(dmix->server_fd); return ret; } else if (ret == 0) { ret = fork(); if (ret == 0) server_job(dmix); _exit(EXIT_SUCCESS); } else { waitpid(ret, NULL, 0); } _From what I understand, this is a concern since if our app is the first to start dmix then a copy of the process (app) will remain till the system is shutdown ? Also I see that from alsa-lib 1.0.12 a check is performed before starting dmix. file: src/pcm/pcm_dmix.c function: snd_pcm_dmix_open if (dmix->shmptr->use_server) { dmix->server_free = dmix_server_free; ret = snd_pcm_direct_server_create(dmix); if (ret < 0) { SNDERR("unable to create server"); goto _err; } } When I checked how dmix->shmptr->use_server is set file: src/pcm/pcm_dmix.c function: snd_pcm_direct_initialize_slave int ver = 0; ioctl(spcm->poll_fd, SNDRV_PCM_IOCTL_PVERSION, &ver); if (ver < SNDRV_PROTOCOL_VERSION(2, 0, 8)) dmix->shmptr->use_server = 1; So if sndrv version is older than 2.0.8, use_server is set! So even using alsa-lib greater than 1.0.12 doesn't guarantee that the forked process (without exec) code path will be avoided. I'm currently porting the sound code of our app from OSS to ALSA but this problem is preventing me from doing so. Is there any way to get around this problem? Help here will be greatly appreciated. Thanks, Bankim. PS: I've tried using the ioctl mentioned above after copying the necessary macros from kernel headers but ioctl returns error "Inappropriate ioctl for this device" on my Ubuntu 7.04 system.

1 0

[alsa-devel] amixer and mono
by Jon Smirl 03 Aug '08

03 Aug '08

Amixer says my device is mono: root@phyCORE-MPC5200B-tiny:~ amixer Simple mixer control 'PCM',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 1023 Mono: Playback 942 [92%] root@phyCORE-MPC5200B-tiny:~ That's coming from the Volume control.... if (snd_mixer_selem_has_playback_volume(elem) || snd_mixer_selem_has_playback_switch(elem)) { printf("%sPlayback channels:", space); if (snd_mixer_selem_is_playback_mono(elem)) { printf(" Mono"); } else { The device's master volume control controls all channels through one value. The device really is stereo. You use biquads if you want to process the channels differently. What do I do about this? -- Jon Smirl jonsmirl(a)gmail.com

2 1

[alsa-devel] ultra small alsa library
by Marco Schuster 02 Aug '08

02 Aug '08

hi all, i try to run a creative extigy via the usb port on a nslu2. the nslu2 is a arm-based (xscale) mini-server with 32mb of ram and so very resource limited if you let services run on it. when i now attach the extigy and play some random stuff - doesn't matter which player - the sound is very intermittent and not recognizable. i think this is due to massive swapping somewhere. what are the minimum prerequisites for using s-alsa (esp. kernel!)? with 2.6.24 and debian i got segfaults. thank you, marco

2 1

[alsa-devel] git clone alsa-driver behind a firewall?
by John L. Utz III 02 Aug '08

02 Aug '08

Hi guys; I am trying to re-sync so that i can try to get my vt1618 stuff into alsa that got rejected last year. But since my last attempted patch, alsa's moved from hg to git and i cant seem to clone the repository. jutz@jutz-gnto64 ~/Compile/AlsaDev $ git clone git://git.alsa-project.org/alsa-driver.git alsa-driver Initialized empty Git repository in /home/jutz/Compile/AlsaDev/alsa-driver/.git/ git.alsa-project.org[0: 212.20.107.51]: errno=Connection timed out fatal: unable to connect a socket (Connection timed out) fetch-pack from 'git://git.alsa-project.org/alsa-driver.git' failed. am i correct in assuming that git runs on a port that a diligent admin would not have open by default? tnx! johnu

2 6

[alsa-devel] [PATCH] ALSA: ASoC: fix SNDCTL_DSP_SYNC support in Freescale 8610 sound drivers
by Timur Tabi 01 Aug '08

01 Aug '08

If an OSS application calls SNDCTL_DSP_SYNC, then ALSA will call the driver's _hw_params and _prepare functions again. On the Freescale MPC8610 DMA ASoC driver, this caused the DMA controller to be unneccessarily re-programmed, and apparently it doesn't like that. The DMA will then not operate when instructed. This patch relocates much of the DMA programming to fsl_dma_open(), which is called only once. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for 2.6.27. A separate patch for ASoC V2 will be posted later. sound/soc/fsl/fsl_dma.c | 235 +++++++++++++++++++++++++---------------------- 1 files changed, 124 insertions(+), 111 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 7ceea2b..d2d3da9 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -327,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, * fsl_dma_open: open a new substream. * * Each substream has its own DMA buffer. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that ping-pong from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. */ static int fsl_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private; + struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; unsigned int channel; int ret = 0; + unsigned int i; /* * Reject any DMA buffer whose size is not a multiple of the period @@ -395,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); runtime->private_data = dma_private; + /* Program the fixed DMA controller parameters */ + + dma_channel = dma_private->dma_channel; + + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* + * Program the mode register for interrupts, external master control, + * and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transferring. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + return 0; } /** - * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors. - * - * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link - * descriptors that ping-pong from one period to the next. For example, if - * there are six periods and two link descriptors, this is how they look - * before playback starts: - * - * The last link descriptor - * ____________ points back to the first - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * | | - * V V - * _________________________________________ - * | | | | | | | The DMA buffer is - * | | | | | | | divided into 6 parts - * |______|______|______|______|______|______| - * - * and here's how they look after the first period is finished playing: - * - * ____________ - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * |______________ - * | | - * V V - * _________________________________________ - * | | | | | | | - * | | | | | | | - * |______|______|______|______|______|______| + * fsl_dma_hw_params: continue initializing the DMA links * - * The first link descriptor now points to the third period. The DMA - * controller is currently playing the second period. When it finishes, it - * will jump back to the first descriptor and play the third period. - * - * There are four reasons we do this: - * - * 1. The only way to get the DMA controller to automatically restart the - * transfer when it gets to the end of the buffer is to use chaining - * mode. Basic direct mode doesn't offer that feature. - * 2. We need to receive an interrupt at the end of every period. The DMA - * controller can generate an interrupt at the end of every link transfer - * (aka segment). Making each period into a DMA segment will give us the - * interrupts we need. - * 3. By creating only two link descriptors, regardless of the number of - * periods, we do not need to reallocate the link descriptors if the - * number of periods changes. - * 4. All of the audio data is still stored in a single, contiguous DMA - * buffer, which is what ALSA expects. We're just dividing it into - * contiguous parts, and creating a link descriptor for each one. + * This function obtains hardware parameters about the opened stream and + * programs the DMA controller accordingly. * * Note that due to a quirk of the SSI's STX register, the target address * for the DMA operations depends on the sample size. So we don't program @@ -468,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; dma_addr_t temp_addr; /* Pointer to next period */ - u64 temp_link; /* Pointer to next link descriptor */ - u32 mr; /* Temporary variable for MR register */ unsigned int i; @@ -490,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, dma_private->dma_buf_next = dma_private->dma_buf_phys; /* - * Initialize each link descriptor. - * * The actual address in STX0 (destination for playback, source for * capture) is based on the sample size, but we don't know the sample * size in this function, so we'll have to adjust that later. See @@ -507,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * buffer itself. */ temp_addr = substream->dma_buffer.addr; - temp_link = dma_private->ld_buf_phys + - sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; link->count = cpu_to_be32(period_size); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) link->source_addr = cpu_to_be32(temp_addr); @@ -524,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, link->dest_addr = cpu_to_be32(temp_addr); temp_addr += period_size; - temp_link += sizeof(struct fsl_dma_link_descriptor); } - /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); - - /* Tell the DMA controller where the first link descriptor is */ - out_be32(&dma_channel->clndar, - CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); - out_be32(&dma_channel->eclndar, - CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); - - /* The manual says the BCR must be clear before enabling EMP */ - out_be32(&dma_channel->bcr, 0); - - /* - * Program the mode register for interrupts, external master control, - * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* - * We want External Master Start and External Master Pause enabled, - * because the SSI is controlling the DMA controller. We want the DMA - * controller to be set up in advance, and then we signal only the SSI - * to start transfering. - * - * We want End-Of-Segment Interrupts enabled, because this will generate - * an interrupt at the end of each segment (each link descriptor - * represents one segment). Each DMA segment is the same thing as an - * ALSA period, so this is how we get an interrupt at the end of every - * period. - * - * We want Error Interrupt enabled, so that we can get an error if - * the DMA controller is mis-programmed somehow. - */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | - CCSR_DMA_MR_EMS_EN; - - /* For playback, we want the destination address to be held. For - capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; - - out_be32(&dma_channel->mr, mr); return 0; } -- 1.5.5

1 0

[alsa-devel] [PATCH 01/14] ASoC: Update Poodle to current ASoC API
by Mark Brown 01 Aug '08

01 Aug '08

From: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Richard Purdie <rpurdie(a)rpsys.net> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This is a compile fix and it would therefore be good if it could be considered for merge into 2.6.27. sound/soc/pxa/poodle.c | 8 ++------ 1 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 65a4e9a..d968cf7 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream) } /* we need to unmute the HP at shutdown as the mute burns power on poodle */ -static int poodle_shutdown(struct snd_pcm_substream *substream) +static void poodle_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; - /* set = unmute headphone */ locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - return 0; } static int poodle_hw_params(struct snd_pcm_substream *substream, @@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = { SOC_ENUM_SINGLE_EXT(2, spk_function), }; -static const snd_kcontrol_new_t wm8731_poodle_controls[] = { +static const struct snd_kcontrol_new wm8731_poodle_controls[] = { SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, poodle_set_jack), SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, -- 1.5.6.3

3 36

[alsa-devel] WM9712 Not Giving output
by Nitin Mahajan 01 Aug '08

01 Aug '08

Hello! I am using a WM9712 codec chip with the AC97 controller on AMD AU1250. Linux kernel version 2.6.24.2. I have used the WM9712 codec driver from sound/soc/codecs. This I got after applying some patch available at wolfson website. The driver comes up, Identifies the codec chip etc. While I do a playback using aplay I cannot hear anything. I probed the pin 39 of the codec chip , but nothing is coming there, how ever on the SDATAOUT of the code chip I can see the signals while I do a playback. On AC97 controller's side, I can see the DMA happening on the host processor, interrupts also coming. On the mixer front, I did unmute and set the vol to max for PCM and Headphone. I did not find any Master control with this chip. Isn't there any master control with this chip? Can some one please throw more light on the Mixer part, specific to this chip. What Mixer controls do I need to set with what values for playback and capture respectively with this chip? Thanks in advance regards -Nitin Mahajan Send instant messages to your online friends http://uk.messenger.yahoo.com

3 5

[alsa-devel] pcm_lib.c: XRUN alsa-back-porting
by Timur Tabi 01 Aug '08

01 Aug '08

A customer is reporting that he's seeing this message every now and then, but when I try to examine the kernel source code, I don't see the source of it. In fact, when I do a google search of "alsa-back-porting", I get zero hits. So maybe the customer is using some older kernel, but when I scan the git change history for sound/core/pcm_lib.c, I still don't see it. So can someone tell me what "alsa-back-porting" means, and where it comes from? -- Timur Tabi Linux kernel developer at Freescale

2 1

Re: [alsa-devel] [PATCH] Support for Terrasoniq/MUSONIK TS22 PCI card
by Takashi Iwai 01 Aug '08

01 Aug '08

At Thu, 31 Jul 2008 14:40:45 -0400, Misha Zhilin wrote: > > Added support for Terrasoniq/MUSONIK TS22 PCI card. > > Signed-off-by: Misha Zhilin <misha(a)epiphan.com> Thanks, applied now. Takashi

1 0

[alsa-devel] [PATCH: alsa-utils] iecset update for new consumer status channel bits
by Pawel MOLL 01 Aug '08

01 Aug '08

Added iecset support for some of consumer status channel bits, new in the third edition od IEC60958-3 spec. Signed-off-by: Pawel Moll <pawel.moll(a)st.com> diff --git a/iecset/iecbits.c b/iecset/iecbits.c index 8c62749..84d439b 100644 --- a/iecset/iecbits.c +++ b/iecset/iecbits.c @@ -26,17 +26,39 @@ struct category_str { }; static struct category_str con_category[] = { + { IEC958_AES1_CON_GENERAL, "general" }, + + { IEC958_AES1_CON_IEC908_CD, "CD" }, + { IEC958_AES1_CON_NON_IEC908_CD, "non-IEC908 CD" }, + { IEC958_AES1_CON_MINI_DISC, "Mini-Disc" }, + { IEC958_AES1_CON_DVD, "DVD" }, + + { IEC958_AES1_CON_PCM_CODER, "PCM coder" }, + { IEC958_AES1_CON_MIXER, "digital signal mixer" }, + { IEC958_AES1_CON_RATE_CONVERTER, "rate converter" }, + { IEC958_AES1_CON_SAMPLER, "sampler" }, + { IEC958_AES1_CON_DSP, "digital sound processor" }, + { IEC958_AES1_CON_DAT, "DAT" }, { IEC958_AES1_CON_VCR, "VCR" }, - { IEC958_AES1_CON_MICROPHONE, "microphone" }, + { IEC958_AES1_CON_DCC, "DCC" }, + { IEC958_AES1_CON_MAGNETIC_DISC, "magnetic disc" }, + + { IEC958_AES1_CON_DAB_JAPAN, "digital audio broadcast (Japan)" }, + { IEC958_AES1_CON_DAB_EUROPE, "digital audio broadcast (Europe)" }, + { IEC958_AES1_CON_DAB_USA, "digital audio broadcast (USA)" }, + { IEC958_AES1_CON_SOFTWARE, "software delivery" }, + { IEC958_AES1_CON_SYNTHESIZER, "synthesizer" }, - { IEC958_AES1_CON_RATE_CONVERTER, "rate converter" }, - { IEC958_AES1_CON_MIXER, "mixer" }, - { IEC958_AES1_CON_SAMPLER, "sampler" }, - { IEC958_AES1_CON_PCM_CODER, "PCM coder" }, - { IEC958_AES1_CON_IEC908_CD, "CD" }, - { IEC958_AES1_CON_NON_IEC908_CD, "non-IEC908 CD" }, - { IEC958_AES1_CON_GENERAL, "general" }, + { IEC958_AES1_CON_MICROPHONE, "microphone" }, + + { IEC958_AES1_CON_ADC, "ADC without copyright information" }, + + { IEC958_AES1_CON_ADC_COPYRIGHT, "ADC with copyright information" }, + + { IEC958_AES1_CON_SOLIDMEM_DIGITAL_RECORDER_PLAYER, "flash memory recorder/player" }, + + { IEC958_AES1_CON_EXPERIMENTAL, "experimental" }, }; @@ -57,14 +79,38 @@ void dump_iec958(snd_aes_iec958_t *iec) } printf("Rate: "); switch (iec->status[3] & IEC958_AES3_CON_FS) { + case IEC958_AES3_CON_FS_22050: + printf("22050 Hz\n"); + break; + case IEC958_AES3_CON_FS_24000: + printf("24000 Hz\n"); + break; + case IEC958_AES3_CON_FS_32000: + printf("32000 Hz\n"); + break; case IEC958_AES3_CON_FS_44100: printf("44100 Hz\n"); break; case IEC958_AES3_CON_FS_48000: printf("48000 Hz\n"); break; - case IEC958_AES3_CON_FS_32000: - printf("32000 Hz\n"); + case IEC958_AES3_CON_FS_88200: + printf("88200 Hz\n"); + break; + case IEC958_AES3_CON_FS_96000: + printf("96000 Hz\n"); + break; + case IEC958_AES3_CON_FS_176400: + printf("176400 Hz\n"); + break; + case IEC958_AES3_CON_FS_192000: + printf("192000 Hz\n"); + break; + case IEC958_AES3_CON_FS_768000: + printf("768000 Hz\n"); + break; + case IEC958_AES3_CON_FS_NOTID: + printf("not indicated\n"); break; default: printf("unknown\n"); diff --git a/iecset/iecset.c b/iecset/iecset.c index a2fe4d1..44c43ab 100644 --- a/iecset/iecset.c +++ b/iecset/iecset.c @@ -58,7 +58,7 @@ static struct cmdtbl cmds[] = { { "aud", IDX_NOAUDIO, CMD_BOOL_INV, "audio (common)\n\ton = audio mode, off = non-audio mode" }, { "rat", IDX_RATE, CMD_INT, - "rate (common)\n\tsample rate in Hz" }, + "rate (common)\n\tsample rate in Hz (0 = not indicated)" }, { "emp", IDX_EMP, CMD_INT, "emphasis (common)\n\t0 = none, 1 = 50/15us, 2 = CCITT" }, { "loc", IDX_UNLOCK, CMD_BOOL_INV, @@ -194,14 +194,38 @@ static int update_iec958_status(snd_aes_iec958_t *iec958, int *parms) } else { iec958->status[3] &= ~IEC958_AES3_CON_FS; switch (parms[IDX_RATE]) { + case 22050: + iec958->status[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + iec958->status[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + iec958->status[3] |= IEC958_AES3_CON_FS_32000; + break; case 44100: iec958->status[3] |= IEC958_AES3_CON_FS_44100; break; case 48000: iec958->status[3] |= IEC958_AES3_CON_FS_48000; break; - case 32000: - iec958->status[3] |= IEC958_AES3_CON_FS_32000; + case 88200: + iec958->status[3] |= IEC958_AES3_CON_FS_88200;; + break; + case 96000: + iec958->status[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + iec958->status[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + iec958->status[3] |= IEC958_AES3_CON_FS_192000; + break; + case 768000: + iec958->status[3] |= IEC958_AES3_CON_FS_768000; + break; + default: + iec958->status[3] |= IEC958_AES3_CON_FS_NOTID; break; } } -- 1.5.5.1

1 0

[alsa-devel] [PATCH: alsa-lib] IEC958 definition for consumer status channel update
by Pawel MOLL 01 Aug '08

01 Aug '08

Updated IEC958 consumer status channel definitions according to the third edition of IEC60958-3 spec. Signed-off-by: Pawel Moll <pawel.moll(a)st.com> diff --git a/include/asoundef.h b/include/asoundef.h index e780c65..c6c4eec 100644 --- a/include/asoundef.h +++ b/include/asoundef.h @@ -69,35 +69,56 @@ extern "C" { #define IEC958_AES1_PRO_USERBITS_UDEF (12<<4) /**< user defined application */ #define IEC958_AES1_CON_CATEGORY 0x7f /**< consumer category */ #define IEC958_AES1_CON_GENERAL 0x00 /**< general category */ -#define IEC958_AES1_CON_EXPERIMENTAL 0x40 /**< experimental category */ -#define IEC958_AES1_CON_SOLIDMEM_MASK 0x0f /**< ??? */ -#define IEC958_AES1_CON_SOLIDMEM_ID 0x08 /**< ??? */ -#define IEC958_AES1_CON_BROADCAST1_MASK 0x07 /**< ??? */ -#define IEC958_AES1_CON_BROADCAST1_ID 0x04 /**< ??? */ -#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07 /**< digital<->digital converter mask */ -#define IEC958_AES1_CON_DIGDIGCONV_ID 0x02 /**< digital<->digital converter id */ -#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f /**< ADC Copyright mask */ -#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x06 /**< ADC Copyright ID */ -#define IEC958_AES1_CON_ADC_MASK 0x1f /**< ADC Mask */ -#define IEC958_AES1_CON_ADC_ID 0x16 /**< ADC ID */ -#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f /**< Broadcast mask */ -#define IEC958_AES1_CON_BROADCAST2_ID 0x0e /**< Broadcast ID */ #define IEC958_AES1_CON_LASEROPT_MASK 0x07 /**< Laser-optical mask */ #define IEC958_AES1_CON_LASEROPT_ID 0x01 /**< Laser-optical ID */ -#define IEC958_AES1_CON_MUSICAL_MASK 0x07 /**< Musical device mask */ -#define IEC958_AES1_CON_MUSICAL_ID 0x05 /**< Musical device ID */ -#define IEC958_AES1_CON_MAGNETIC_MASK 0x07 /**< Magnetic device mask */ -#define IEC958_AES1_CON_MAGNETIC_ID 0x03 /**< Magnetic device ID */ #define IEC958_AES1_CON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x00) /**< IEC958 CD compatible device */ #define IEC958_AES1_CON_NON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x08) /**< non-IEC958 CD compatible device */ +#define IEC958_AES1_CON_MINI_DISC (IEC958_AES1_CON_LASEROPT_ID|0x48) /**< Mini-Disc device */ +#define IEC958_AES1_CON_DVD (IEC958_AES1_CON_LASEROPT_ID|0x18) /**< DVD device */ +#define IEC958_AES1_CON_LASTEROPT_OTHER (IEC958_AES1_CON_LASEROPT_ID|0x78) /**< Other laser-optical product */ +#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07 /**< digital<->digital converter mask */ +#define IEC958_AES1_CON_DIGDIGCONV_ID 0x02 /**< digital<->digital converter id */ #define IEC958_AES1_CON_PCM_CODER (IEC958_AES1_CON_DIGDIGCONV_ID|0x00) /**< PCM coder */ -#define IEC958_AES1_CON_SAMPLER (IEC958_AES1_CON_DIGDIGCONV_ID|0x20) /**< PCM sampler */ -#define IEC958_AES1_CON_MIXER (IEC958_AES1_CON_DIGDIGCONV_ID|0x10) /**< Mixer */ +#define IEC958_AES1_CON_MIXER (IEC958_AES1_CON_DIGDIGCONV_ID|0x10) /**< Digital signal mixer */ #define IEC958_AES1_CON_RATE_CONVERTER (IEC958_AES1_CON_DIGDIGCONV_ID|0x18) /**< Rate converter */ -#define IEC958_AES1_CON_SYNTHESIZER (IEC958_AES1_CON_MUSICAL_ID|0x00) /**< Synthesizer */ -#define IEC958_AES1_CON_MICROPHONE (IEC958_AES1_CON_MUSICAL_ID|0x08) /**< Microphone */ +#define IEC958_AES1_CON_SAMPLER (IEC958_AES1_CON_DIGDIGCONV_ID|0x20) /**< PCM sampler */ +#define IEC958_AES1_CON_DSP (IEC958_AES1_CON_DIGDIGCONV_ID|0x28) /**< Digital sound processor */ +#define IEC958_AES1_CON_DIGDIGCONV_OTHER (IEC958_AES1_CON_DIGDIGCONV_ID|0x78) /**< Other digital<->digital product */ +#define IEC958_AES1_CON_MAGNETIC_MASK 0x07 /**< Magnetic device mask */ +#define IEC958_AES1_CON_MAGNETIC_ID 0x03 /**< Magnetic device ID */ #define IEC958_AES1_CON_DAT (IEC958_AES1_CON_MAGNETIC_ID|0x00) /**< Digital Audio Tape */ #define IEC958_AES1_CON_VCR (IEC958_AES1_CON_MAGNETIC_ID|0x08) /**< Video recorder */ +#define IEC958_AES1_CON_DCC (IEC958_AES1_CON_MAGNETIC_ID|0x40) /**< Digital compact cassette */ +#define IEC958_AES1_CON_MAGNETIC_DISC (IEC958_AES1_CON_MAGNETIC_ID|0x18) /**< Magnetic disc digital audio device */ +#define IEC958_AES1_CON_MAGNETIC_OTHER (IEC958_AES1_CON_MAGNETIC_ID|0x78) /**< Other magnetic device */ +#define IEC958_AES1_CON_BROADCAST1_MASK 0x07 /**< Broadcast mask */ +#define IEC958_AES1_CON_BROADCAST1_ID 0x04 /**< Broadcast ID */ +#define IEC958_AES1_CON_DAB_JAPAN (IEC958_AES1_CON_BROADCAST1_ID|0x00) /**< Digital audio broadcast (Japan) */ +#define IEC958_AES1_CON_DAB_EUROPE (IEC958_AES1_CON_BROADCAST1_ID|0x08) /**< Digital audio broadcast (Europe) */ +#define IEC958_AES1_CON_DAB_USA (IEC958_AES1_CON_BROADCAST1_ID|0x60) /**< Digital audio broadcast (USA) */ +#define IEC958_AES1_CON_SOFTWARE (IEC958_AES1_CON_BROADCAST1_ID|0x40) /**< Electronic software delivery */ +#define IEC958_AES1_CON_IEC62105 (IEC958_AES1_CON_BROADCAST1_ID|0x20) /**< Used by another standard (IEC 62105) */ +#define IEC958_AES1_CON_BROADCAST1_OTHER (IEC958_AES1_CON_BROADCAST1_ID|0x78) /**< Other broadcast product */ +#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f /**< Broadcast alternative mask */ +#define IEC958_AES1_CON_BROADCAST2_ID 0x0e /**< Broadcast alternative ID */ +#define IEC958_AES1_CON_MUSICAL_MASK 0x07 /**< Musical device mask */ +#define IEC958_AES1_CON_MUSICAL_ID 0x05 /**< Musical device ID */ +#define IEC958_AES1_CON_SYNTHESIZER (IEC958_AES1_CON_MUSICAL_ID|0x00) /**< Synthesizer */ +#define IEC958_AES1_CON_MICROPHONE (IEC958_AES1_CON_MUSICAL_ID|0x08) /**< Microphone */ +#define IEC958_AES1_CON_MUSICAL_OTHER (IEC958_AES1_CON_MUSICAL_ID|0x78) /**< Other musical device */ +#define IEC958_AES1_CON_ADC_MASK 0x1f /**< ADC Mask */ +#define IEC958_AES1_CON_ADC_ID 0x06 /**< ADC ID */ +#define IEC958_AES1_CON_ADC (IEC958_AES1_CON_ADC_ID|0x00) /**< ADC without copyright information */ +#define IEC958_AES1_CON_ADC_OTHER (IEC958_AES1_CON_ADC_ID|0x60) /**< Other ADC product (with no copyright information) */ +#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f /**< ADC Copyright mask */ +#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x16 /**< ADC Copyright ID */ +#define IEC958_AES1_CON_ADC_COPYRIGHT (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x00) /**< ADC with copyright information */ +#define IEC958_AES1_CON_ADC_COPYRIGHT_OTHER (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x60) /**< Other ADC with copyright information product */ +#define IEC958_AES1_CON_SOLIDMEM_MASK 0x0f /**< Solid memory based products mask */ +#define IEC958_AES1_CON_SOLIDMEM_ID 0x08 /**< Solid memory based products ID */ +#define IEC958_AES1_CON_SOLIDMEM_DIGITAL_RECORDER_PLAYER (IEC958_AES1_CON_SOLIDMEM_ID|0x00) /**< Digital audio recorder and player using solid state memory */ +#define IEC958_AES1_CON_SOLIDMEM_OTHER (IEC958_AES1_CON_SOLIDMEM_ID|0x70) /**< Other solid state memory based product */ +#define IEC958_AES1_CON_EXPERIMENTAL 0x40 /**< experimental category */ #define IEC958_AES1_CON_ORIGINAL (1<<7) /**< this bits depends on the category code */ #define IEC958_AES2_PRO_SBITS (7<<0) /**< mask - sample bits */ #define IEC958_AES2_PRO_SBITS_20 (2<<0) /**< 20-bit - coordination */ @@ -115,8 +136,16 @@ extern "C" { #define IEC958_AES2_CON_CHANNEL_UNSPEC (0<<4) /**< channel number unspecified */ #define IEC958_AES3_CON_FS (15<<0) /**< mask - sample frequency */ #define IEC958_AES3_CON_FS_44100 (0<<0) /**< 44.1kHz */ +#define IEC958_AES3_CON_FS_NOTID (1<<0) /**< sample frequency non indicated */ #define IEC958_AES3_CON_FS_48000 (2<<0) /**< 48kHz */ #define IEC958_AES3_CON_FS_32000 (3<<0) /**< 32kHz */ +#define IEC958_AES3_CON_FS_22050 (4<<0) /**< 22.05kHz */ +#define IEC958_AES3_CON_FS_24000 (6<<0) /**< 24kHz */ +#define IEC958_AES3_CON_FS_88200 (8<<0) /**< 88.2kHz */ +#define IEC958_AES3_CON_FS_768000 (9<<0) /**< 768kHz */ +#define IEC958_AES3_CON_FS_96000 (10<<0) /**< 96kHz */ +#define IEC958_AES3_CON_FS_176400 (12<<0) /**< 176.4kHz */ +#define IEC958_AES3_CON_FS_192000 (14<<0) /**< 192kHz */ #define IEC958_AES3_CON_CLOCK (3<<4) /**< mask - clock accuracy */ #define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /**< 1000 ppm */ #define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /**< 50 ppm */ @@ -129,6 +158,26 @@ extern "C" { #define IEC958_AES4_CON_WORDLEN_23_19 (4<<1) /**< 23-bit or 19-bit */ #define IEC958_AES4_CON_WORDLEN_24_20 (5<<1) /**< 24-bit or 20-bit */ #define IEC958_AES4_CON_WORDLEN_21_17 (6<<1) /**< 21-bit or 17-bit */ +#define IEC958_AES4_CON_ORIGFS (15<<4) /**< mask - original sample frequency */ +#define IEC958_AES4_CON_ORIGFS_NOTID (0<<4) /**< original sample frequency not indicated */ +#define IEC958_AES4_CON_ORIGFS_192000 (1<<4) /**< 192kHz */ +#define IEC958_AES4_CON_ORIGFS_12000 (2<<4) /**< 12kHz */ +#define IEC958_AES4_CON_ORIGFS_176400 (3<<4) /**< 176.4kHz */ +#define IEC958_AES4_CON_ORIGFS_96000 (5<<4) /**< 96kHz */ +#define IEC958_AES4_CON_ORIGFS_8000 (6<<4) /**< 8kHz */ +#define IEC958_AES4_CON_ORIGFS_88200 (7<<4) /**< 88.2kHz */ +#define IEC958_AES4_CON_ORIGFS_16000 (8<<4) /**< 16kHz */ +#define IEC958_AES4_CON_ORIGFS_24000 (9<<4) /**< 24kHz */ +#define IEC958_AES4_CON_ORIGFS_11025 (10<<4) /**< 11.025kHz */ +#define IEC958_AES4_CON_ORIGFS_22050 (11<<4) /**< 22.05kHz */ +#define IEC958_AES4_CON_ORIGFS_32000 (12<<4) /**< 32kHz */ +#define IEC958_AES4_CON_ORIGFS_48000 (13<<4) /**< 48kHz */ +#define IEC958_AES4_CON_ORIGFS_44100 (15<<4) /**< 44.1kHz */ +#define IEC958_AES5_CON_CGMSA (3<<0) /**< mask - CGMS-A */ +#define IEC958_AES5_CON_CGMSA_COPYFREELY (0<<0) /**< copying is permitted without restriction */ +#define IEC958_AES5_CON_CGMSA_COPYONCE (1<<0) /**< one generation of copies may be made */ +#define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /**< condition not be used */ +#define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /**< no copying is permitted */ /** \} */ -- 1.5.5.1

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[alsa-devel] [PATCH: alsa-kernel] ALSA: IEC958 definition for consumer status channel update
by Pawel MOLL 01 Aug '08

01 Aug '08

Updated IEC958 consumer status channel definitions according to the third edition of IEC60958-3 spec. Signed-off-by: Pawel Moll <pawel.moll(a)st.com> diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index a6e0fac..20ebf32 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -60,35 +60,56 @@ #define IEC958_AES1_PRO_USERBITS_UDEF (12<<4) /* user defined application */ #define IEC958_AES1_CON_CATEGORY 0x7f #define IEC958_AES1_CON_GENERAL 0x00 -#define IEC958_AES1_CON_EXPERIMENTAL 0x40 -#define IEC958_AES1_CON_SOLIDMEM_MASK 0x0f -#define IEC958_AES1_CON_SOLIDMEM_ID 0x08 -#define IEC958_AES1_CON_BROADCAST1_MASK 0x07 -#define IEC958_AES1_CON_BROADCAST1_ID 0x04 -#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07 -#define IEC958_AES1_CON_DIGDIGCONV_ID 0x02 -#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f -#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x06 -#define IEC958_AES1_CON_ADC_MASK 0x1f -#define IEC958_AES1_CON_ADC_ID 0x16 -#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f -#define IEC958_AES1_CON_BROADCAST2_ID 0x0e #define IEC958_AES1_CON_LASEROPT_MASK 0x07 #define IEC958_AES1_CON_LASEROPT_ID 0x01 -#define IEC958_AES1_CON_MUSICAL_MASK 0x07 -#define IEC958_AES1_CON_MUSICAL_ID 0x05 -#define IEC958_AES1_CON_MAGNETIC_MASK 0x07 -#define IEC958_AES1_CON_MAGNETIC_ID 0x03 #define IEC958_AES1_CON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x00) #define IEC958_AES1_CON_NON_IEC908_CD (IEC958_AES1_CON_LASEROPT_ID|0x08) +#define IEC958_AES1_CON_MINI_DISC (IEC958_AES1_CON_LASEROPT_ID|0x48) +#define IEC958_AES1_CON_DVD (IEC958_AES1_CON_LASEROPT_ID|0x18) +#define IEC958_AES1_CON_LASTEROPT_OTHER (IEC958_AES1_CON_LASEROPT_ID|0x78) +#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07 +#define IEC958_AES1_CON_DIGDIGCONV_ID 0x02 #define IEC958_AES1_CON_PCM_CODER (IEC958_AES1_CON_DIGDIGCONV_ID|0x00) -#define IEC958_AES1_CON_SAMPLER (IEC958_AES1_CON_DIGDIGCONV_ID|0x20) #define IEC958_AES1_CON_MIXER (IEC958_AES1_CON_DIGDIGCONV_ID|0x10) #define IEC958_AES1_CON_RATE_CONVERTER (IEC958_AES1_CON_DIGDIGCONV_ID|0x18) -#define IEC958_AES1_CON_SYNTHESIZER (IEC958_AES1_CON_MUSICAL_ID|0x00) -#define IEC958_AES1_CON_MICROPHONE (IEC958_AES1_CON_MUSICAL_ID|0x08) +#define IEC958_AES1_CON_SAMPLER (IEC958_AES1_CON_DIGDIGCONV_ID|0x20) +#define IEC958_AES1_CON_DSP (IEC958_AES1_CON_DIGDIGCONV_ID|0x28) +#define IEC958_AES1_CON_DIGDIGCONV_OTHER (IEC958_AES1_CON_DIGDIGCONV_ID|0x78) +#define IEC958_AES1_CON_MAGNETIC_MASK 0x07 +#define IEC958_AES1_CON_MAGNETIC_ID 0x03 #define IEC958_AES1_CON_DAT (IEC958_AES1_CON_MAGNETIC_ID|0x00) #define IEC958_AES1_CON_VCR (IEC958_AES1_CON_MAGNETIC_ID|0x08) +#define IEC958_AES1_CON_DCC (IEC958_AES1_CON_MAGNETIC_ID|0x40) +#define IEC958_AES1_CON_MAGNETIC_DISC (IEC958_AES1_CON_MAGNETIC_ID|0x18) +#define IEC958_AES1_CON_MAGNETIC_OTHER (IEC958_AES1_CON_MAGNETIC_ID|0x78) +#define IEC958_AES1_CON_BROADCAST1_MASK 0x07 +#define IEC958_AES1_CON_BROADCAST1_ID 0x04 +#define IEC958_AES1_CON_DAB_JAPAN (IEC958_AES1_CON_BROADCAST1_ID|0x00) +#define IEC958_AES1_CON_DAB_EUROPE (IEC958_AES1_CON_BROADCAST1_ID|0x08) +#define IEC958_AES1_CON_DAB_USA (IEC958_AES1_CON_BROADCAST1_ID|0x60) +#define IEC958_AES1_CON_SOFTWARE (IEC958_AES1_CON_BROADCAST1_ID|0x40) +#define IEC958_AES1_CON_IEC62105 (IEC958_AES1_CON_BROADCAST1_ID|0x20) +#define IEC958_AES1_CON_BROADCAST1_OTHER (IEC958_AES1_CON_BROADCAST1_ID|0x78) +#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f +#define IEC958_AES1_CON_BROADCAST2_ID 0x0e +#define IEC958_AES1_CON_MUSICAL_MASK 0x07 +#define IEC958_AES1_CON_MUSICAL_ID 0x05 +#define IEC958_AES1_CON_SYNTHESIZER (IEC958_AES1_CON_MUSICAL_ID|0x00) +#define IEC958_AES1_CON_MICROPHONE (IEC958_AES1_CON_MUSICAL_ID|0x08) +#define IEC958_AES1_CON_MUSICAL_OTHER (IEC958_AES1_CON_MUSICAL_ID|0x78) +#define IEC958_AES1_CON_ADC_MASK 0x1f +#define IEC958_AES1_CON_ADC_ID 0x06 +#define IEC958_AES1_CON_ADC (IEC958_AES1_CON_ADC_ID|0x00) +#define IEC958_AES1_CON_ADC_OTHER (IEC958_AES1_CON_ADC_ID|0x60) +#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f +#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x16 +#define IEC958_AES1_CON_ADC_COPYRIGHT (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x00) +#define IEC958_AES1_CON_ADC_COPYRIGHT_OTHER (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x60) +#define IEC958_AES1_CON_SOLIDMEM_MASK 0x0f +#define IEC958_AES1_CON_SOLIDMEM_ID 0x08 +#define IEC958_AES1_CON_SOLIDMEM_DIGITAL_RECORDER_PLAYER (IEC958_AES1_CON_SOLIDMEM_ID|0x00) +#define IEC958_AES1_CON_SOLIDMEM_OTHER (IEC958_AES1_CON_SOLIDMEM_ID|0x70) +#define IEC958_AES1_CON_EXPERIMENTAL 0x40 #define IEC958_AES1_CON_ORIGINAL (1<<7) /* this bits depends on the category code */ #define IEC958_AES2_PRO_SBITS (7<<0) /* mask - sample bits */ #define IEC958_AES2_PRO_SBITS_20 (2<<0) /* 20-bit - coordination */ @@ -106,8 +127,16 @@ #define IEC958_AES2_CON_CHANNEL_UNSPEC (0<<4) /* unspecified */ #define IEC958_AES3_CON_FS (15<<0) /* mask - sample frequency */ #define IEC958_AES3_CON_FS_44100 (0<<0) /* 44.1kHz */ +#define IEC958_AES3_CON_FS_NOTID (1<<0) /* non indicated */ #define IEC958_AES3_CON_FS_48000 (2<<0) /* 48kHz */ #define IEC958_AES3_CON_FS_32000 (3<<0) /* 32kHz */ +#define IEC958_AES3_CON_FS_22050 (4<<0) /* 22.05kHz */ +#define IEC958_AES3_CON_FS_24000 (6<<0) /* 24kHz */ +#define IEC958_AES3_CON_FS_88200 (8<<0) /* 88.2kHz */ +#define IEC958_AES3_CON_FS_768000 (9<<0) /* 768kHz */ +#define IEC958_AES3_CON_FS_96000 (10<<0) /* 96kHz */ +#define IEC958_AES3_CON_FS_176400 (12<<0) /* 176.4kHz */ +#define IEC958_AES3_CON_FS_192000 (14<<0) /* 192kHz */ #define IEC958_AES3_CON_CLOCK (3<<4) /* mask - clock accuracy */ #define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /* 1000 ppm */ #define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /* 50 ppm */ @@ -120,6 +149,26 @@ #define IEC958_AES4_CON_WORDLEN_23_19 (4<<1) /* 23-bit or 19-bit */ #define IEC958_AES4_CON_WORDLEN_24_20 (5<<1) /* 24-bit or 20-bit */ #define IEC958_AES4_CON_WORDLEN_21_17 (6<<1) /* 21-bit or 17-bit */ +#define IEC958_AES4_CON_ORIGFS (15<<4) /* mask - original sample frequency */ +#define IEC958_AES4_CON_ORIGFS_NOTID (0<<4) /* not indicated */ +#define IEC958_AES4_CON_ORIGFS_192000 (1<<4) /* 192kHz */ +#define IEC958_AES4_CON_ORIGFS_12000 (2<<4) /* 12kHz */ +#define IEC958_AES4_CON_ORIGFS_176400 (3<<4) /* 176.4kHz */ +#define IEC958_AES4_CON_ORIGFS_96000 (5<<4) /* 96kHz */ +#define IEC958_AES4_CON_ORIGFS_8000 (6<<4) /* 8kHz */ +#define IEC958_AES4_CON_ORIGFS_88200 (7<<4) /* 88.2kHz */ +#define IEC958_AES4_CON_ORIGFS_16000 (8<<4) /* 16kHz */ +#define IEC958_AES4_CON_ORIGFS_24000 (9<<4) /* 24kHz */ +#define IEC958_AES4_CON_ORIGFS_11025 (10<<4) /* 11.025kHz */ +#define IEC958_AES4_CON_ORIGFS_22050 (11<<4) /* 22.05kHz */ +#define IEC958_AES4_CON_ORIGFS_32000 (12<<4) /* 32kHz */ +#define IEC958_AES4_CON_ORIGFS_48000 (13<<4) /* 48kHz */ +#define IEC958_AES4_CON_ORIGFS_44100 (15<<4) /* 44.1kHz */ +#define IEC958_AES5_CON_CGMSA (3<<0) /* mask - CGMS-A */ +#define IEC958_AES5_CON_CGMSA_COPYFREELY (0<<0) /* copying is permitted without restriction */ +#define IEC958_AES5_CON_CGMSA_COPYONCE (1<<0) /* one generation of copies may be made */ +#define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /* condition not be used */ +#define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /* no copying is permitted */ /***************************************************************************** * * -- 1.5.5.1

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Re: [alsa-devel] [Fwd: [Alsa-user] Intel Poulsbo(SCH) chipset does not recognize Adi 1986 codec under linux]]
by Kan-I Jyo 01 Aug '08

01 Aug '08

Dear list, My original mail on this topic was posted to the alsa-user list, while I was informed that this alsa-devel list would be the right place to post. The problem I had encountered is that the Adi 1986 codec chip on my device with Poulsbo(SCH) chipset could not be recognized by the snd_hda_intel driver module. Tobin has kindly given me an advice on trying an ALSA driver that is later than 1.0.16.final. For a cross-test pupose, I have tried both the following 1.0.16 and the latest 1.0.17-rc2. [1.0.16] ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.16.tar.bz2 [1.0.17-rc2] ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.17rc2.tar.bz2 By following the guide described in the INSTALL text file in both the above two packages, I have done make and make install without a problem. Unfortunately, both these two version of ALSA driver does not recognized the Adi 1986 codec and in the dmesg output claiming that "no codecs found!". <dmesg> -- snip -- ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 16 (level, low) -> IRQ 17 PCI: Setting latency timer of device 0000:00:1b.0 to 64 ALSA /root/alsa-driver-1.0.16/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1887: hda-intel: no codecs found! ACPI: PCI interrupt for device 0000:00:1b.0 disabled ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 16 (level, low) -> IRQ 17 PCI: Setting latency timer of device 0000:00:1b.0 to 64 ALSA /root/alsa-driver-1.0.17rc2/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:2142: hda-intel: no codecs found! ACPI: PCI interrupt for device 0000:00:1b.0 disabled -- snip -- I have also tried applying some module parameters like probe_mask and position_fix but with no luck. Actually, we have two machines with Poulsbo(SCH) chipset inside. The ALSA driver works fine in one(with a chipset's revision of "04") while it does not work on the other (with a chipset's revision of "06"). Here I attach the "lspci -nnvvvxxxx" result from both two machines. Any comment would be greatly appericiated. If anyone needs more information, please let me know. ---------- Forwarded message ---------- 2008/6/24 stan <ghjeold_i_mwee(a)cox.net>: > This is a response from the alsa-devel list. I forwarded your message there > as that was the more appropriate list. > > > Not sure what version of alsa driver you are using, but the driver for that > chipset was added prior to 1.0.16 final. Anything after that should work. > > Tobin > > On Mon, 2008-06-23 at 06:20 -0700, stan wrote: > > Sent to wrong list. > > email message attachment ([Alsa-user] Intel Poulsbo(SCH) chipset does not > recognize Adi 1986 codec under linux.eml) > > -------- Forwarded Message -------- > From: Kan-I Jyo <cecilhsujp(a)gmail.com> > To: alsa-user(a)lists.sourceforge.net > Subject: [Alsa-user] Intel Poulsbo(SCH) chipset does not recognize Adi 1986 > codec under linux > Date: Mon, 23 Jun 2008 20:53:42 +0900 > > Dear list, > > I have recently got a device with Intel's Poulsbo chipset and Adi 1986 > codec attatched. > > The Adi 1986 codec along with the Poulsbo chipset could been > recognized by installing the SoundMAX driver under Windows XP. > > While switching to Linux, the snd_intel_hda is loaded but the Adi 1986 > codec could not been seen and there is no "codec#0" entry under > /proc/asound/card0. > > >From what I can see from dmesg, the hda_intel module had complained > about "no codecs found!". > > <dmesg.txt> > -- snip -- > [ 93.463277] PCI: Setting latency timer of device 0000:00:1b.0 to 64 > [ 93.463286] Do fixup for Poulsbo <6>D0 or newer stepping > [ 93.463439] HDA snoop disabled, try to enable ... OK > [ 93.491624] hda-intel: no codecs found! > -- snip -- > > I have attached both output of dmesg command the "lspci -nnvvvxxx" > result along with this mail. Any comment would be greatly appreciated. > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > > _______________________________________________ Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Tobin Davis > > 22. The only copy of Norton Utilities was on THAT disk??? > > --Top 100 things you don't want the sysadmin to say > > -- Sincerely, Jyo

3 11

[alsa-devel] snd_pcm_sw_params_set_xfer_align replacement?
by Aaron J. Grier 01 Aug '08

01 Aug '08

snd_pcm_sw_params_set_xfer_align() has been deprecated but the docs don't mention what it was replaced with. I don't ever want or need partial reads and writes -- my app would actually be happiest being completely synchronous to the hardware with fixed-size buffers. any suggestions how to get ALSA to do this? -- Aaron J. Grier | "Not your ordinary poofy goof." | agrier(a)poofygoof.com

3 6

Re: [alsa-devel] [PATCH 01/11] wss_lib: move cs4231_lib into wss_lib
by Rene Herman 31 Jul '08

31 Jul '08

On 31-07-08 21:00, Krzysztof Helt wrote: > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > Move the file sound/isa/cs423x/cs4231_lib.c > into sound/isa/cs423x/wss_lib.c > This is the first step toward merging all libraries > for Windows Sound System compatible chips > into a single library. > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > Acked-by: Rene Herman <rene.herman(a)gmail.com> > --- > This is second revision of the patch set. It is redone versus linux-2.6.27-rc1-mm. > > This patch has not been changed since previous posting. Thanks for noting that. If you'd know how hot is was here, you'd forgive me for postponing looking until tomorrow. Unlike the others upto now, you addressed this first message to "Alsa-devel(a)interia.pl" and "<alsa-devel(a)alsa-project.org>@interia.pl" and it has indeed not made it to the regular alsa-devel list. Rene.

1 0

[alsa-devel] [PATCH 02/11] wss_lib: rename cs4231.h into wss.h
by Krzysztof Helt 31 Jul '08

31 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Rename file include/sound/cs4231.h into include/sound/wss.h Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> Acked-by: Rene Herman <rene.herman(a)gmail.com> --- This patch has not been changed since previous posting. diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h deleted file mode 100644 index f0785f9..0000000 --- a/include/sound/cs4231.h +++ /dev/null @@ -1,175 +0,0 @@ -#ifndef __SOUND_CS4231_H -#define __SOUND_CS4231_H - -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Definitions for CS4231 & InterWave chips & compatible chips - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include "control.h" -#include "pcm.h" -#include "timer.h" - -#include "cs4231-regs.h" - -/* defines for codec.mode */ - -#define CS4231_MODE_NONE 0x0000 -#define CS4231_MODE_PLAY 0x0001 -#define CS4231_MODE_RECORD 0x0002 -#define CS4231_MODE_TIMER 0x0004 -#define CS4231_MODE_OPEN (CS4231_MODE_PLAY|CS4231_MODE_RECORD|CS4231_MODE_TIMER) - -/* defines for codec.hardware */ - -#define CS4231_HW_DETECT 0x0000 /* let CS4231 driver detect chip */ -#define CS4231_HW_DETECT3 0x0001 /* allow mode 3 */ -#define CS4231_HW_TYPE_MASK 0xff00 /* type mask */ -#define CS4231_HW_CS4231_MASK 0x0100 /* CS4231 serie */ -#define CS4231_HW_CS4231 0x0100 /* CS4231 chip */ -#define CS4231_HW_CS4231A 0x0101 /* CS4231A chip */ -#define CS4231_HW_AD1845 0x0102 /* AD1845 chip */ -#define CS4231_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */ -#define CS4231_HW_CS4232 0x0200 /* CS4232 */ -#define CS4231_HW_CS4232A 0x0201 /* CS4232A */ -#define CS4231_HW_CS4236 0x0202 /* CS4236 */ -#define CS4231_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */ -#define CS4231_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */ -#define CS4231_HW_CS4236B 0x0401 /* CS4236B */ -#define CS4231_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ -#define CS4231_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ -#define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ -/* compatible, but clones */ -#define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ -#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ -#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ - -/* defines for codec.hwshare */ -#define CS4231_HWSHARE_IRQ (1<<0) -#define CS4231_HWSHARE_DMA1 (1<<1) -#define CS4231_HWSHARE_DMA2 (1<<2) - -struct snd_cs4231 { - unsigned long port; /* base i/o port */ - struct resource *res_port; - unsigned long cport; /* control base i/o port (CS4236) */ - struct resource *res_cport; - int irq; /* IRQ line */ - int dma1; /* playback DMA */ - int dma2; /* record DMA */ - unsigned short version; /* version of CODEC chip */ - unsigned short mode; /* see to CS4231_MODE_XXXX */ - unsigned short hardware; /* see to CS4231_HW_XXXX */ - unsigned short hwshare; /* shared resources */ - unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ - ebus_flag:1; /* SPARC: EBUS present */ - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - struct snd_timer *timer; - - unsigned char image[32]; /* registers image */ - unsigned char eimage[32]; /* extended registers image */ - unsigned char cimage[16]; /* control registers image */ - int mce_bit; - int calibrate_mute; - int sw_3d_bit; - unsigned int p_dma_size; - unsigned int c_dma_size; - - spinlock_t reg_lock; - struct mutex mce_mutex; - struct mutex open_mutex; - - int (*rate_constraint) (struct snd_pcm_runtime *runtime); - void (*set_playback_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char pdfr); - void (*set_capture_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char cdfr); - void (*trigger) (struct snd_cs4231 *chip, unsigned int what, int start); -#ifdef CONFIG_PM - void (*suspend) (struct snd_cs4231 *chip); - void (*resume) (struct snd_cs4231 *chip); -#endif - void *dma_private_data; - int (*claim_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); - int (*release_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); -}; - -/* exported functions */ - -void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); -unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg); -void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); -unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); -void snd_cs4231_mce_up(struct snd_cs4231 *chip); -void snd_cs4231_mce_down(struct snd_cs4231 *chip); - -void snd_cs4231_overrange(struct snd_cs4231 *chip); - -irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); - -const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); - -int snd_cs4231_create(struct snd_card *card, - unsigned long port, - unsigned long cport, - int irq, int dma1, int dma2, - unsigned short hardware, - unsigned short hwshare, - struct snd_cs4231 ** rchip); -int snd_cs4231_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); -int snd_cs4231_timer(struct snd_cs4231 * chip, int device, struct snd_timer **rtimer); -int snd_cs4231_mixer(struct snd_cs4231 * chip); - -int snd_cs4236_create(struct snd_card *card, - unsigned long port, - unsigned long cport, - int irq, int dma1, int dma2, - unsigned short hardware, - unsigned short hwshare, - struct snd_cs4231 ** rchip); -int snd_cs4236_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); -int snd_cs4236_mixer(struct snd_cs4231 * chip); - -/* - * mixer library - */ - -#define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_single, \ - .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - -#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_double, \ - .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - -#endif /* __SOUND_CS4231_H */ diff --git a/include/sound/snd_wavefront.h b/include/sound/snd_wavefront.h index 9688d4b..fa149ca 100644 --- a/include/sound/snd_wavefront.h +++ b/include/sound/snd_wavefront.h @@ -1,7 +1,6 @@ #ifndef __SOUND_SND_WAVEFRONT_H__ #define __SOUND_SND_WAVEFRONT_H__ -#include "cs4231.h" #include "mpu401.h" #include "hwdep.h" #include "rawmidi.h" diff --git a/include/sound/wss.h b/include/sound/wss.h new file mode 100644 index 0000000..f0785f9 --- /dev/null +++ b/include/sound/wss.h @@ -0,0 +1,175 @@ +#ifndef __SOUND_CS4231_H +#define __SOUND_CS4231_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Definitions for CS4231 & InterWave chips & compatible chips + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include "control.h" +#include "pcm.h" +#include "timer.h" + +#include "cs4231-regs.h" + +/* defines for codec.mode */ + +#define CS4231_MODE_NONE 0x0000 +#define CS4231_MODE_PLAY 0x0001 +#define CS4231_MODE_RECORD 0x0002 +#define CS4231_MODE_TIMER 0x0004 +#define CS4231_MODE_OPEN (CS4231_MODE_PLAY|CS4231_MODE_RECORD|CS4231_MODE_TIMER) + +/* defines for codec.hardware */ + +#define CS4231_HW_DETECT 0x0000 /* let CS4231 driver detect chip */ +#define CS4231_HW_DETECT3 0x0001 /* allow mode 3 */ +#define CS4231_HW_TYPE_MASK 0xff00 /* type mask */ +#define CS4231_HW_CS4231_MASK 0x0100 /* CS4231 serie */ +#define CS4231_HW_CS4231 0x0100 /* CS4231 chip */ +#define CS4231_HW_CS4231A 0x0101 /* CS4231A chip */ +#define CS4231_HW_AD1845 0x0102 /* AD1845 chip */ +#define CS4231_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */ +#define CS4231_HW_CS4232 0x0200 /* CS4232 */ +#define CS4231_HW_CS4232A 0x0201 /* CS4232A */ +#define CS4231_HW_CS4236 0x0202 /* CS4236 */ +#define CS4231_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */ +#define CS4231_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */ +#define CS4231_HW_CS4236B 0x0401 /* CS4236B */ +#define CS4231_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ +#define CS4231_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ +#define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ +/* compatible, but clones */ +#define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ +#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ + +/* defines for codec.hwshare */ +#define CS4231_HWSHARE_IRQ (1<<0) +#define CS4231_HWSHARE_DMA1 (1<<1) +#define CS4231_HWSHARE_DMA2 (1<<2) + +struct snd_cs4231 { + unsigned long port; /* base i/o port */ + struct resource *res_port; + unsigned long cport; /* control base i/o port (CS4236) */ + struct resource *res_cport; + int irq; /* IRQ line */ + int dma1; /* playback DMA */ + int dma2; /* record DMA */ + unsigned short version; /* version of CODEC chip */ + unsigned short mode; /* see to CS4231_MODE_XXXX */ + unsigned short hardware; /* see to CS4231_HW_XXXX */ + unsigned short hwshare; /* shared resources */ + unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ + ebus_flag:1; /* SPARC: EBUS present */ + + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + struct snd_timer *timer; + + unsigned char image[32]; /* registers image */ + unsigned char eimage[32]; /* extended registers image */ + unsigned char cimage[16]; /* control registers image */ + int mce_bit; + int calibrate_mute; + int sw_3d_bit; + unsigned int p_dma_size; + unsigned int c_dma_size; + + spinlock_t reg_lock; + struct mutex mce_mutex; + struct mutex open_mutex; + + int (*rate_constraint) (struct snd_pcm_runtime *runtime); + void (*set_playback_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char pdfr); + void (*set_capture_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char cdfr); + void (*trigger) (struct snd_cs4231 *chip, unsigned int what, int start); +#ifdef CONFIG_PM + void (*suspend) (struct snd_cs4231 *chip); + void (*resume) (struct snd_cs4231 *chip); +#endif + void *dma_private_data; + int (*claim_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); + int (*release_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); +}; + +/* exported functions */ + +void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); +unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg); +void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); +unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); +void snd_cs4231_mce_up(struct snd_cs4231 *chip); +void snd_cs4231_mce_down(struct snd_cs4231 *chip); + +void snd_cs4231_overrange(struct snd_cs4231 *chip); + +irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); + +const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); + +int snd_cs4231_create(struct snd_card *card, + unsigned long port, + unsigned long cport, + int irq, int dma1, int dma2, + unsigned short hardware, + unsigned short hwshare, + struct snd_cs4231 ** rchip); +int snd_cs4231_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); +int snd_cs4231_timer(struct snd_cs4231 * chip, int device, struct snd_timer **rtimer); +int snd_cs4231_mixer(struct snd_cs4231 * chip); + +int snd_cs4236_create(struct snd_card *card, + unsigned long port, + unsigned long cport, + int irq, int dma1, int dma2, + unsigned short hardware, + unsigned short hwshare, + struct snd_cs4231 ** rchip); +int snd_cs4236_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); +int snd_cs4236_mixer(struct snd_cs4231 * chip); + +/* + * mixer library + */ + +#define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_cs4231_info_single, \ + .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + +int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); + +#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_cs4231_info_double, \ + .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } + +int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); + +#endif /* __SOUND_CS4231_H */ diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index 154e728..a24db09 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -38,7 +38,7 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/initval.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index e9462b9..7da28e7 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -27,7 +27,7 @@ #include <linux/wait.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index dbe63db..7b9a3cc 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -26,7 +26,7 @@ #include <linux/pnp.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index de71910..eb227d8 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -85,7 +85,7 @@ #include <linux/time.h> #include <linux/wait.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/asoundef.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index f87c623..cd82891 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -28,7 +28,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/gus.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index ca0d7ac..eac8cc7 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -32,7 +32,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/gus.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #ifdef SNDRV_STB #include <sound/tea6330t.h> #endif diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 854a9f7..e81cbe8 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -28,7 +28,7 @@ #include <linux/pnp.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 2a1e2f5..59f7c55 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl4.h> #include <sound/control.h> diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0797ca4..93a03d9 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -34,7 +34,7 @@ #include <asm/dma.h> #include <sound/core.h> #if defined(CS4231) || defined(OPTi93X) -#include <sound/cs4231.h> +#include <sound/wss.h> #else #include <sound/ad1848.h> #endif /* CS4231 */ diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 06ad786..1dc4224 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -31,7 +31,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/hwdep.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 3a6c6fe..2fb058b 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -29,6 +29,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/opl3.h> +#include <sound/wss.h> #include <sound/snd_wavefront.h> MODULE_AUTHOR("Paul Barton-Davis <pbd(a)op.net>"); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index c5795f3..549e6ab 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -31,7 +31,7 @@ #include <linux/slab.h> #include <linux/ioport.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/pcm_params.h> #include <asm/io.h> -- 1.5.5 ---------------------------------------------------------------------- Partyjka w Chinczyka? Graj >>> http://link.interia.pl/f1e67

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[alsa-devel] I2S simple driver
by Alessio Sangalli 31 Jul '08

31 Jul '08

Hi, we are writing a small driver to control a I2S interface on our ARM-based system... Right now we are going to implement a "easier" interface that allows to set some parameters, and then simply write to the device that will automatically control the DAC and then to the speakers. Now, this is of course not ALSA compliant. How can we estimate the effort to make this simple driver ALSA compatible so that every ALSA application can use this device for playback? We do not have any volume control and the range of sound "modes" is extremely limited. bye as

2 2

[alsa-devel] Proper way to set SCMS copy prohibited bits on IEC958 S/PDIF
by Andrew Paprocki 31 Jul '08

31 Jul '08

I briefly checked the code for the Delta1010 (ice1712) to see if there was a way from user space to force the SCMS copy prohibited bits on in the S/PDIF stream. I see the driver initializes the bits to 0. Is there already a way to do this from user space, or would I have to modify the driver to force them on (or use a module parameter)? Thanks, -Andrew

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Re: [alsa-devel] [PATCH] pxa2xx-lib: support building for several pxa's
by Dmitry 31 Jul '08

31 Jul '08

Hi, 2008/7/30 Russ Dill <russ.dill(a)gmail.com>: > Here's a version of the third patch in the series without all the > #ifdef's. Just a quick question for Dmitry, why all the if (! > cpu_is_pxa3xx())'s around the pxa_gpio_mode's? Also, there was a build > error when just building the sound/arm pxa2xx-ac97. Good! AFAIR pxa_gpio_mode isn't valid for the the pxa3xx, so cpu_is_pxa3xx(). > > Support building pxa2xx-lib for several pxa chip versions by making code > run-time selected, not only compile-time > > Signed-off-by: Russ Dill <russ.dill(a)gmail.com> > --- > include/asm-arm/arch-pxa/pxa-regs.h | 4 +- > sound/arm/Kconfig | 2 +- > sound/arm/pxa2xx-ac97-lib.c | 174 ++++++++++++++++------------------ > sound/soc/pxa/pxa2xx-i2s.c | 16 ++- > 4 files changed, 95 insertions(+), 101 deletions(-) > -- With best wishes Dmitry

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Re: [alsa-devel] [PATCH] pxa2xx-lib: support building for several pxa's
by Eric Miao 31 Jul '08

31 Jul '08

> -----Original Message----- > From: Russ Dill [mailto:russ.dill@gmail.com] > Sent: Thursday, July 31, 2008 7:29 AM > To: Dmitry > Cc: alsa-devel(a)alsa-project.org; broonie(a)opensource.wolfsonmicro.com; Eric > Miao > Subject: Re: [PATCH] pxa2xx-lib: support building for several pxa's > > On Wed, Jul 30, 2008 at 3:41 PM, Dmitry <dbaryshkov(a)gmail.com> wrote: > > Hi, > > > > 2008/7/30 Russ Dill <russ.dill(a)gmail.com>: > >> Here's a version of the third patch in the series without all the > >> #ifdef's. Just a quick question for Dmitry, why all the if (! > >> cpu_is_pxa3xx())'s around the pxa_gpio_mode's? Also, there was a build > >> error when just building the sound/arm pxa2xx-ac97. > > > > Good! > > > > AFAIR pxa_gpio_mode isn't valid for the the pxa3xx, so cpu_is_pxa3xx(). > > I can't find any reason within the source that it wouldn't work. pxafb > uses it, which is used by littleton and zylonite. I'm not sure when > pxa_gpio_mode should be used vs the mfp pin config. > pxa_gpio_mode() is deprecated, please don't use that, even with pxa2xx platforms. The reason some legacy pxa_gpio_mode() is still there, e.g. pxafb, is that not all platforms have transitioned into use the new MFP API, til then, I'll schedule a cleanup to remove all those pxa_gpio_mode, so whenever possible, don't introduce any new pxa_gpio_mode(). P.S. the reason pxa_gpio_mode() doesn't cause any obvious consequence is that PXA3xx no longer has those GAFR registers, but that's really going to affect other registers, esp. the GPDR, which isn't correct. > >> > >> Support building pxa2xx-lib for several pxa chip versions by making code > >> run-time selected, not only compile-time > >> > >> Signed-off-by: Russ Dill <russ.dill(a)gmail.com> > >> --- > >> include/asm-arm/arch-pxa/pxa-regs.h | 4 +- > >> sound/arm/Kconfig | 2 +- > >> sound/arm/pxa2xx-ac97-lib.c | 174 > ++++++++++++++++------------------ > >> sound/soc/pxa/pxa2xx-i2s.c | 16 ++- > >> 4 files changed, 95 insertions(+), 101 deletions(-) > >> > > -- > > With best wishes > > Dmitry > >

1 0

[alsa-devel] device_create_drvdata missing on older kernels
by Eliot Blennerhassett 31 Jul '08

31 Jul '08

I could not compile current git. (Error on acore/init.c) Adding this to adriver.h fixed it for me. I'm not sure if this is the correct general fix? #ifndef CONFIG_SND_HAS_DEVICE_CREATE_DRVDATA #define device_create_drvdata(c,d,m,p,s,n) device_create((c),(d),(m),(s),(n)) #endif (My kernel 2.6.24-19-generic) -- -- Eliot Blennerhassett www.audioscience.com

3 3

Re: [alsa-devel] alsa-lib has anyone ever looked at taking it cross platform?
by Peter Dolding 30 Jul '08

30 Jul '08

> Peter Dolding wrote: >> This would kinda make logical sence. >> >> Alsa-lib plugin system in theory should be able to take any kind sound >> output under it. Simplify development of all applications on top of >> it. If alsa was cross platform why would you use something else as a >> middle body wrapper when you port. >> >> Windows and Mac support could be done as plugins. >> >> Or is there something that is in the alsa-lib that is a problem? >> >> Its all about reducing distance between application and hardware and >> reduce amount of coding. > > I'd rather see pulseaudio go fully cross platform. It's got some older > windows ports but if pulse were to get a push in the right direction for > windows and OSX then this lets alsa get on with the job of interfacing > with the audio on linux (that's what the L stands for after all), and > let higher level applications work with a more appropriate system with > more appropriate interfaces for practical use. Ok sorry to say the idea of let sound server do it is the same as saying we will put up for every with xlib being single threaded and have the toolkits above it deal with it. Instead of taking on the problem. xcb is taking on the problem in the X11 stack of libs. Except its many times worse. Pulseaudio and all other Sound Servers since its a extra service adds a extra point of complete sound failure to the Linux sound stack or what ever stack they happen to be operating on top of. It more of a hack around the limitations of the alsa framework. ALSA was really designed to avoid the need of such extras. The existence of Pulseaudio is really sign of failure. Reason Sound Servers existed at first due to the failures of OSS to provide a multi application sound support. ALSA was really designed to do away for the need of sound servers completely. Now alsa is failing to provide per application sound control and a sound server is back for another pass. Now why are not alsa interfaces practical to use. It is ment to be "Advanced Linux Sound Architecture" The complete Architecture. Not something you have to tack bits onto because its not functional or hard to use. Have we got to the point like xlib where the complete alsa-lib has to be reworked for usability. Now if you want to keep on thinking along that line has ALSA got to the point it should be dumped like OSS has been dumped because of its limitations? alsa-lib to freebsd or somewhere with oss most likely could be done really quickly since we already have a oss output option. Its also like saying the Linux Standard Base is only used by Linux. Its not its also used by a few non Linux's. Driver section of ALSA most likely will always be Linux only. But the wrapper layer that hides the hardware from the software really does not have to stay Linux only. Alsa is going up to be part of the Linux Standard Base its about time we all start looking at how it can be made most useful. Provide all the features it should and work for stability. Has anyone one attempted it or not. I guess not. Peter Dolding

4 3

[alsa-devel] [PATCH] Permit simultaneous compilation of both PXA AC97 drivers
by Dmitry Baryshkov 30 Jul '08

30 Jul '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2df03ee..45ac468 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -360,7 +360,7 @@ int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_soc_platform pxa2xx_soc_platform = { .name = "pxa2xx-audio", .pcm_ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_pcm_new, + .pcm_new = pxa2xx_soc_pcm_new, .pcm_free = pxa2xx_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); -- 1.5.6.2

5 8

[alsa-devel] [PATCH 1/1] ALSA: Fix limit of 8 PCM devices in SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE
by Pawel MOLL 30 Jul '08

30 Jul '08

ALSA: Fix limit of 8 PCM devices in SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine to have more than 8 PCM devices per card, except one place - the SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate devices > 7. This patch fixes the issue, changing the devices list organisation. Instead of adding new device to the tail, the list is now kept always ordered (by card number, then device number). Thus, during enumeration, it is easy to discover the fact that there is no more given card's devices. The same limit was present in OSS emulation code. It has been fixed as well. Additionally the device field of struct snd_pcm is now int, instead of unsigned int, as there is no obvious reason for keeping it unsigned. This caused a lot of problems with comparing this value with other (almost always signed) variables. There is just one more place where device number is unsigned - in struct snd_pcm_info, which should be also sorted out in future. Signed-off-by: Pawel MOLL <pawel.moll(a)st.com> diff --git a/include/sound/minors.h b/include/sound/minors.h index 46bcd20..a81798a 100644 --- a/include/sound/minors.h +++ b/include/sound/minors.h @@ -21,6 +21,8 @@ * */ +#define SNDRV_OS_MINORS 256 + #define SNDRV_MINOR_DEVICES 32 #define SNDRV_MINOR_CARD(minor) ((minor) >> 5) #define SNDRV_MINOR_DEVICE(minor) ((minor) & 0x001f) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 51d58cc..bfc096a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -84,8 +84,6 @@ struct snd_pcm_ops { * */ -#define SNDRV_PCM_DEVICES 8 - #define SNDRV_PCM_IOCTL1_FALSE ((void *)0) #define SNDRV_PCM_IOCTL1_TRUE ((void *)1) @@ -416,7 +414,7 @@ struct snd_pcm_str { struct snd_pcm { struct snd_card *card; struct list_head list; - unsigned int device; /* device number */ + int device; /* device number */ unsigned int info_flags; unsigned short dev_class; unsigned short dev_subclass; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 4c601b1..4ccd761 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2947,7 +2947,7 @@ static void register_oss_dsp(struct snd_pcm *pcm, int index) static int snd_pcm_oss_register_minor(struct snd_pcm *pcm) { pcm->oss.reg = 0; - if (dsp_map[pcm->card->number] == (int)pcm->device) { + if (dsp_map[pcm->card->number] == pcm->device) { char name[128]; int duplex; register_oss_dsp(pcm, 0); @@ -2963,7 +2963,7 @@ static int snd_pcm_oss_register_minor(struct snd_pcm *pcm) pcm->oss.reg++; pcm->oss.reg_mask |= 1; } - if (adsp_map[pcm->card->number] == (int)pcm->device) { + if (adsp_map[pcm->card->number] == pcm->device) { register_oss_dsp(pcm, 1); pcm->oss.reg++; pcm->oss.reg_mask |= 2; @@ -2988,7 +2988,7 @@ static int snd_pcm_oss_disconnect_minor(struct snd_pcm *pcm) snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_PCM, pcm->card, 1); } - if (dsp_map[pcm->card->number] == (int)pcm->device) { + if (dsp_map[pcm->card->number] == pcm->device) { #ifdef SNDRV_OSS_INFO_DEV_AUDIO snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_AUDIO, pcm->card->number); #endif @@ -3019,12 +3019,12 @@ static int __init alsa_pcm_oss_init(void) /* check device map table */ for (i = 0; i < SNDRV_CARDS; i++) { - if (dsp_map[i] < 0 || dsp_map[i] >= SNDRV_PCM_DEVICES) { + if (dsp_map[i] < 0 || dsp_map[i] >= SNDRV_OS_MINORS) { snd_printk(KERN_ERR "invalid dsp_map[%d] = %d\n", i, dsp_map[i]); dsp_map[i] = 0; } - if (adsp_map[i] < 0 || adsp_map[i] >= SNDRV_PCM_DEVICES) { + if (adsp_map[i] < 0 || adsp_map[i] >= SNDRV_OS_MINORS) { snd_printk(KERN_ERR "invalid adsp_map[%d] = %d\n", i, adsp_map[i]); adsp_map[i] = 1; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9dd9bc7..df4c8a3 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -42,7 +42,7 @@ static int snd_pcm_dev_free(struct snd_device *device); static int snd_pcm_dev_register(struct snd_device *device); static int snd_pcm_dev_disconnect(struct snd_device *device); -static struct snd_pcm *snd_pcm_search(struct snd_card *card, int device) +static inline struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) { struct snd_pcm *pcm; @@ -53,6 +53,37 @@ static struct snd_pcm *snd_pcm_search(struct snd_card *card, int device) return NULL; } +static inline int snd_pcm_next(struct snd_card *card, int device) +{ + struct snd_pcm *pcm; + + list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->card == card && pcm->device > device) + return pcm->device; + else if (pcm->card->number > card->number) + return -1; + } + return -1; +} + +static inline int snd_pcm_add(struct snd_pcm *newpcm) +{ + struct snd_pcm *pcm; + + list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->card == newpcm->card && pcm->device == newpcm->device) + return -EBUSY; + if (pcm->card->number > newpcm->card->number || + (pcm->card == newpcm->card && + pcm->device > newpcm->device)) { + list_add(&newpcm->list, pcm->list.prev); + return 0; + } + } + list_add_tail(&newpcm->list, &snd_pcm_devices); + return 0; +} + static int snd_pcm_control_ioctl(struct snd_card *card, struct snd_ctl_file *control, unsigned int cmd, unsigned long arg) @@ -65,14 +96,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; - while (device < SNDRV_PCM_DEVICES) { - if (snd_pcm_search(card, device)) - break; - device++; - } - if (device == SNDRV_PCM_DEVICES) - device = -1; + device = snd_pcm_next(card, device); mutex_unlock(&register_mutex); if (put_user(device, (int __user *)arg)) return -EFAULT; @@ -98,7 +122,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(&register_mutex); - pcm = snd_pcm_search(card, device); + pcm = snd_pcm_get(card, device); if (pcm == NULL) { err = -ENXIO; goto _error; @@ -931,11 +955,11 @@ static int snd_pcm_dev_register(struct snd_device *device) snd_assert(pcm != NULL && device != NULL, return -ENXIO); mutex_lock(&register_mutex); - if (snd_pcm_search(pcm->card, pcm->device)) { + err = snd_pcm_add(pcm); + if (err) { mutex_unlock(&register_mutex); - return -EBUSY; + return err; } - list_add_tail(&pcm->list, &snd_pcm_devices); for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; if (pcm->streams[cidx].substream == NULL) diff --git a/sound/core/sound.c b/sound/core/sound.c index 1003ae3..838dd9e 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -34,8 +34,6 @@ #include <linux/kmod.h> #include <linux/mutex.h> -#define SNDRV_OS_MINORS 256 - static int major = CONFIG_SND_MAJOR; int snd_major; EXPORT_SYMBOL(snd_major); -- 1.5.5.1

3 6

[alsa-devel] multiple identical controls
by Jon Smirl 30 Jul '08

30 Jul '08

My hardware has four channels, each with seven identical biquads. I'm making a custom control to handle them. Don't I want to give then an index from 0 to 6? Why is snd_soc_cnew() setting index to zero? struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, char *long_name) { struct snd_kcontrol_new template; memcpy(&template, _template, sizeof(template)); if (long_name) template.name = long_name; template.index = 0; return snd_ctl_new1(&template, data); } EXPORT_SYMBOL_GPL(snd_soc_cnew); -- Jon Smirl jonsmirl(a)gmail.com

2 1

[alsa-devel] [PATCH v2] asoc: restrict sample rate and size in Freescale MPC8610 sound drivers
by Timur Tabi 29 Jul '08

29 Jul '08

The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Please apply this patch for 2.6.26.1 and for 2.6.27-rc1. Without this patch, recording on the MPC8610 is just plain broken, so it is a must-fix for 2.6.26. sound/soc/fsl/fsl_dma.c | 7 ++++- sound/soc/fsl/fsl_ssi.c | 74 ++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 70 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc59..7ceea2b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33..157a789 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,8 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: pointer to the stream that was opened first + * @second_stream: pointer to second stream * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +81,8 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) */ } + if (!ssi_private->first_stream) + ssi_private->first_stream = substream; + else { + /* This is the second stream open, so we need to impose sample + * rate and maybe sample size constraints. Note that this can + * cause a race condition if the second stream is opened before + * the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); + + ssi_private->second_stream = substream; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ssi_private->playback++; @@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ssi_private->capture--; + if (ssi_private->first_stream == substream) + ssi_private->first_stream = ssi_private->second_stream; + + ssi_private->second_stream = NULL; + /* * If this is the last active substream, disable the SSI and release * the IRQ. -- 1.5.5

2 3

[alsa-devel] [PATCH] ASOC: convert use of uint to unsigned int
by Jon Smirl 29 Jul '08

29 Jul '08

ASOC: convert use of uint to unsigned int Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5ca231f..a1e0357 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -527,7 +527,7 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max; - uint reg, rreg, shift, rshift, invert; + unsigned int reg, rreg, shift, rshift, invert; }; /* enumerated kcontrol */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 11a881c..1563cee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1457,8 +1457,8 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - uint shift = mc->min; - uint rshift = mc->rshift; + unsigned int shift = mc->min; + unsigned int rshift = mc->rshift; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1487,12 +1487,12 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - uint reg = mc->reg; - uint shift = mc->shift; - uint rshift = mc->rshift; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int max = mc->max; - uint mask = (1 << fls(max)) - 1; - uint invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1526,12 +1526,12 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - uint reg = mc->reg; - uint shift = mc->shift; - uint rshift = mc->rshift; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int max = mc->max; - uint mask = (1 << fls(max)) - 1; - uint invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); @@ -1594,12 +1594,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - uint reg = mc->reg; - uint reg2 = mc->rreg; - uint shift = mc->shift; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; int max = mc->max; - uint mask = (1<<fls(max))-1; - uint invert = mc->invert; + unsigned int mask = (1<<fls(max))-1; + unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1631,12 +1631,12 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - uint reg = mc->reg; - uint reg2 = mc->rreg; - uint shift = mc->shift; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; int max = mc->max; - uint mask = (1 << fls(max)) - 1; - uint invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; int err; unsigned short val, val2, val_mask; @@ -1701,7 +1701,7 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - uint reg = mc->reg; + unsigned int reg = mc->reg; int min = mc->min; int val = snd_soc_read(codec, reg); @@ -1728,7 +1728,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - uint reg = mc->reg; + unsigned int reg = mc->reg; int min = mc->min; unsigned short val; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f08be8a..7a88f76 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -106,11 +106,11 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrols[i].private_value; - uint reg = mc->reg; - uint shift = mc->shift; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; int max = mc->max; - uint mask = (1 << fls(max)) - 1; - uint invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; val = snd_soc_read(w->codec, reg); val = (val >> shift) & mask; @@ -252,11 +252,11 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) if (widget->num_kcontrols && k) { struct soc_mixer_control *mc = (struct soc_mixer_control *)k->private_value; - uint reg = mc->reg; - uint shift = mc->shift; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; int max = mc->max; - uint mask = (1 << fls(max)) - 1; - uint invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; if (power) { int i; @@ -1141,12 +1141,12 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - uint reg = mc->reg; - uint shift = mc->shift; - uint rshift = mc->rshift; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int max = mc->max; - uint invert = mc->invert; - uint mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; /* return the saved value if we are powered down */ if (widget->id == snd_soc_dapm_pga && !widget->power) { @@ -1186,12 +1186,12 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - uint reg = mc->reg; - uint shift = mc->shift; - uint rshift = mc->rshift; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int max = mc->max; - uint mask = (1 << fls(max)) - 1; - uint invert = mc->invert; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val2, val_mask; int ret;

3 2

[alsa-devel] [PATCH - asihpi 1/5] split public and private defs
by linux＠audioscience.com 29 Jul '08

29 Jul '08

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/hpi.h b/pci/asihpi/hpi.h index 4f52d09..1892271 100644 --- a/pci/asihpi/hpi.h +++ b/pci/asihpi/hpi.h @@ -21,9 +21,7 @@ AudioScience digital audio adapters You must define one operating systems that the HPI is to be compiled under - HPI_OS_WIN16 16bit Windows HPI_OS_WIN32_USER 32bit Windows - HPI_OS_DSP_563XX DSP563XX environment HPI_OS_DSP_C6000 DSP TI C6000 environment HPI_OS_WDM Windows WDM kernel driver HPI_OS_LINUX Linux kernel driver @@ -32,7 +30,6 @@ ******************************************************************************/ #ifndef _HPI_H_ #define _HPI_H_ - /* HPI Version If HPI_VER_MINOR is odd then its a development release not intended for the public. If HPI_VER_MINOR is even then is a release version @@ -46,79 +43,10 @@ i.e 3.05.02 is a development version #define HPI_VER_RELEASE(v) ((int)(v & 0xFF)) /* Use single digits for versions less that 10 to avoid octal. */ -#define HPI_VER HPI_VERSION_CONSTRUCTOR(3L, 10, 3) - -#ifdef _DOX_ONLY_ -/*****************************************************************************/ -/** \defgroup compile_time_defines HPI compile time defines - -This section descibes the usage of HPI defines to set the target compile -environment. -The definitions are used in the build environment to control how HPI is built. -They are NOT predefined in hpi.h! -\{ -*/ -/** \defgroup os_type_defines HPI_OS_XXX build environment defines - -Define exactly one of these depending on which OS you are compiling for. -Should we also include DSP_53XXX, C6000, WDM etc? - -\{ -*/ -/**Define when creating a 16 bit Windows user application. */ -#define HPI_OS_WIN16 -/** Define when creating a 32 bit Windows user application. */ -#define HPI_OS_WIN32_USER -/** Define when creating a Linux application.*/ -#define HPI_OS_LINUX -/* Should we also include DSP_53XXX, C6000, WDM etc? */ -/**\}*/ -/** \def HPI64BIT -Define this when building a 64 bit application. -When not defined a 32 bit environment will be assumed. -Currently only supported under Linux (autodetected). -*/ -#define HPI64BIT - -/** Define this to remove public definition of deprecated functions and defines. -Use this to reveal where the deprecated functionality is used -*/ -#define HPI_EXCLUDE_DEPRECATED -/** \defgroup hpi_dll_defines HPI DLL function attributes -DLL environment defines. -\{ -*/ -/** Define when building an application that links to ASIHPI32.LIB - (imports HPI functions from ASIHPI32.DLL). */ -#define HPIDLL_IMPORTS -/** Define when creating an application that uses the HPI DLL. */ -#define HPIDLL_EXPORTS -/** Define when building an application that compiles in - HPIFUNC.C and does not use ASIHPI32.DLL */ -#define HPIDLL_STATIC -/**\}*/ +#define HPI_VER HPI_VERSION_CONSTRUCTOR(3L, 11, 9) -/** \defgroup hpi_dspcode_defines HPI DSP code loading method -Define exactly one of these to select how the DSP code is supplied to -the adapter. - -End users writing applications that use the HPI interface do not have to -use any of the below defines. -They are only necessary for building drivers and 16 bit Windows applications. -\{ -*/ -/** DSP code is supplied as a series of arrays that are linked into - the driver/application. Only valid when \ref HPI_OS_WIN16 is defined. */ -#define HPI_DSPCODE_ARRAY -/** DSP code is supplied as a file that is opened and read from by the driver.*/ -#define HPI_DSPCODE_FILE -/** DSP code is read using the hotplug firmware loader module. - Only valid when compiling the HPI kernel driver under Linux. */ -#define HPI_DSPCODE_FIRMWARE -/**\}*/ -/**\}*/ -/*****************************************************************************/ -#endif /* #ifdef _DOX_ONLY_ */ +/* Library version as documented in hpi-api-versions.txt */ +#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(3, 3, 0) #ifdef __cplusplus /* *INDENT-OFF* */ @@ -126,26 +54,7 @@ extern "C" { /* *INDENT-ON* */ #endif -/** maximum number of memory regions mapped to an adapter */ -#define HPI_MAX_ADAPTER_MEM_SPACES (2) - -#include "hpios.h" - -/* A few convenience macros */ -#ifndef DEPRECATED -#define DEPRECATED -#endif - -#ifndef ARRAY_SIZE -#define ARRAY_SIZE(A) (sizeof(A)/sizeof((A)[0])) -#endif - -#ifndef __stringify -#define __stringify_1(x) #x -#define __stringify(x) __stringify_1(x) -#endif - -#define HPI_UNUSED(param) (void)param +#include <linux/types.h> /******************************************************************************/ /******************************************************************************/ @@ -153,10 +62,6 @@ extern "C" { /******************************************************************************/ /******************************************************************************/ -/* //////////////////////////////////////////////////////////////////////// */ -/* BASIC TYPES */ -/* u8, u16, u32 MUST BE DEFINED IN HPIOS_xxx.H */ - /* ////////////////////////////////////////////////////////////////////// */ /** \addtogroup hpi_defines HPI constant definitions \{ @@ -557,20 +462,21 @@ The range is +1.0 to -1.0, which corresponds to digital fullscale. */ #define HPI_SOURCENODE_BASE 100 /** Out Stream (Play) node. */ -#define HPI_SOURCENODE_OSTREAM 101 -#define HPI_SOURCENODE_LINEIN 102 /**< Line in node. */ +#define HPI_SOURCENODE_OSTREAM 101 +/** Line in node - could be analog. AES/EBU or network. */ +#define HPI_SOURCENODE_LINEIN 102 #define HPI_SOURCENODE_AESEBU_IN 103 /**< AES/EBU input node. */ #define HPI_SOURCENODE_TUNER 104 /**< Tuner node. */ -#define HPI_SOURCENODE_RF 105 /**< RF input node. */ -#define HPI_SOURCENODE_CLOCK_SOURCE 106 /**< Clock source node. */ -#define HPI_SOURCENODE_RAW_BITSTREAM 107 /**< Raw bitstream node. */ -#define HPI_SOURCENODE_MICROPHONE 108 /**< Microphone node. */ +#define HPI_SOURCENODE_RF 105 /**< RF input node. */ +#define HPI_SOURCENODE_CLOCK_SOURCE 106 /**< Clock source node. */ +#define HPI_SOURCENODE_RAW_BITSTREAM 107 /**< Raw bitstream node. */ +#define HPI_SOURCENODE_MICROPHONE 108 /**< Microphone node. */ /** Cobranet input node. Audio samples come from the Cobranet network and into the device. */ -#define HPI_SOURCENODE_COBRANET 109 -/*! Update this if you add a new sourcenode type, AND hpidebug.h */ -#define HPI_SOURCENODE_LAST_INDEX 109 -/* AX4 max sourcenode type = 15 */ +#define HPI_SOURCENODE_COBRANET 109 +#define HPI_SOURCENODE_ANALOG 110 /**< Analog input node. */ +/* Update this if you add a new sourcenode type, AND hpidebug.h */ +#define HPI_SOURCENODE_LAST_INDEX 110 /* AX6 max sourcenode type = 15 */ /**\}*/ @@ -578,19 +484,19 @@ The range is +1.0 to -1.0, which corresponds to digital fullscale. /** \defgroup dest_nodes Destination node types \{ */ -#define HPI_DESTNODE_BASE 200 +#define HPI_DESTNODE_BASE 200 /** In Stream (Record) node. */ #define HPI_DESTNODE_ISTREAM 201 #define HPI_DESTNODE_LINEOUT 202 /**< Line Out node. */ -#define HPI_DESTNODE_AESEBU_OUT 203 /**< AES/EBU output node. */ -#define HPI_DESTNODE_RF 204 /**< RF output node. */ +#define HPI_DESTNODE_AESEBU_OUT 203 /**< AES/EBU output node. */ +#define HPI_DESTNODE_RF 204 /**< RF output node. */ #define HPI_DESTNODE_SPEAKER 205 /**< Speaker output node. */ /** Cobranet output node. Audio samples from the device are sent out on the Cobranet network.*/ #define HPI_DESTNODE_COBRANET 206 +#define HPI_DESTNODE_ANALOG 207 /**< Analog output node. */ /*! Update this if you add a new destnode type. , AND hpidebug.h */ -#define HPI_DESTNODE_LAST_INDEX 206 -/* AX4 max destnode type = 7 */ +#define HPI_DESTNODE_LAST_INDEX 207 /* AX6 max destnode type = 15 */ /**\}*/ @@ -610,9 +516,9 @@ Audio samples from the device are sent out on the Cobranet network.*/ #define HPI_CONTROL_TUNER 9 /**< Tuner control. */ /*#define HPI_CONTROL_ONOFFSWITCH 10 */ #define HPI_CONTROL_VOX 11 /**< Vox control. */ -#define HPI_CONTROL_AES18_TRANSMITTER 12 /**< AES-18 transmitter control. */ -#define HPI_CONTROL_AES18_RECEIVER 13 /**< AES-18 receiver control. */ -#define HPI_CONTROL_AES18_BLOCKGENERATOR 14 /**< AES-18 block generator control. */ +/* #define HPI_CONTROL_AES18_TRANSMITTER 12 */ +/* #define HPI_CONTROL_AES18_RECEIVER 13 */ +/* #define HPI_CONTROL_AES18_BLOCKGENERATOR 14 */ #define HPI_CONTROL_CHANNEL_MODE 15 /**< Channel mode control. */ /* WARNING types 16 or greater impact bit packing in @@ -844,18 +750,6 @@ when the number of channels is set to 1 will return an error. #define HPI_MPEG_MODE_DUALCHANNEL (3) /** \} */ -/* Locked memory buffer alloc/free phases */ -/** use one message to allocate or free physical memory */ -#define HPI_BUFFER_CMD_EXTERNAL 0 -/** alloc physical memory */ -#define HPI_BUFFER_CMD_INTERNAL_ALLOC 1 -/** send physical memory address to adapter */ -#define HPI_BUFFER_CMD_INTERNAL_GRANTADAPTER 2 -/** notify adapter to stop using physical buffer */ -#define HPI_BUFFER_CMD_INTERNAL_REVOKEADAPTER 3 -/** free physical buffer */ -#define HPI_BUFFER_CMD_INTERNAL_FREE 4 - /******************************************* MIXER ATTRIBUTES ****/ /* \defgroup mixer_flags Mixer flags for HPI_MIXER_GET_CONTROL_MULTIPLE_VALUES @@ -882,56 +776,11 @@ enum HPI_MIXER_STORE_COMMAND { HPI_MIXER_STORE_SAVE_SINGLE = 6 }; -/******************************************* CONTROL ATTRIBUTES ****/ -/* (in order of control type ID as above */ - -/* This allows for 255 control types, 256 unique attributes each */ -#define HPI_CTL_ATTR(ctl, ai) (HPI_CONTROL_##ctl * 0x100 + ai) - -/* Get the sub-index of the attribute for a control type */ -#define HPI_CTL_ATTR_INDEX(i) (i&0xff) - -/* Original 0-based non-unique attributes, might become unique later */ -#define HPI_CTL_ATTR0(ctl, ai) (ai) - -/* Generic control attributes. If a control uses any of these attributes - its other attributes must also be defined using HPI_CTL_ATTR() -*/ - -/** Enable a control. -0=disable, 1=enable -\note generic to all mixer plugins? -*/ -#define HPI_GENERIC_ENABLE HPI_CTL_ATTR(GENERIC, 1) - -/** Enable event generation for a control. -0=disable, 1=enable -\note generic to all controls that can generate events -*/ -#define HPI_GENERIC_EVENT_ENABLE HPI_CTL_ATTR(GENERIC, 2) - +/************************************* CONTROL ATTRIBUTE VALUES ****/ /* Used by mixer plugin enable functions */ #define HPI_SWITCH_OFF 0 /**< Turn the mixer plugin on. */ #define HPI_SWITCH_ON 1 /**< Turn the mixer plugin off. */ -/* Volume Control attributes */ -#define HPI_VOLUME_GAIN HPI_CTL_ATTR0(VOLUME, 1) -/** log fade - dB attenuation changes linearly over time */ -#define HPI_VOLUME_AUTOFADE_LOG HPI_CTL_ATTR0(VOLUME, 2) -#define HPI_VOLUME_AUTOFADE HPI_VOLUME_AUTOFADE_LOG -/** linear fade - amplitude changes linearly */ -#define HPI_VOLUME_AUTOFADE_LINEAR HPI_CTL_ATTR0(VOLUME, 3) -/** Not implemented yet -may be special profiles */ -#define HPI_VOLUME_AUTOFADE_1 HPI_CTL_ATTR0(VOLUME, 4) -#define HPI_VOLUME_AUTOFADE_2 HPI_CTL_ATTR0(VOLUME, 5) -/** For HPI_ControlQuery() to get the number of channels of a volume control*/ -#define HPI_VOLUME_NUM_CHANNELS HPI_CTL_ATTR0(VOLUME, 6) -#define HPI_VOLUME_RANGE HPI_CTL_ATTR0(VOLUME, 10) - -/** Level Control attributes */ -#define HPI_LEVEL_GAIN HPI_CTL_ATTR0(LEVEL, 1) -#define HPI_LEVEL_RANGE HPI_CTL_ATTR0(LEVEL, 10) - /* Volume control special gain values */ /** volumes units are 100ths of a dB */ #define HPI_UNITS_PER_dB 100 @@ -940,42 +789,11 @@ enum HPI_MIXER_STORE_COMMAND { /** value returned for no signal */ #define HPI_METER_MINIMUM (-150 * HPI_UNITS_PER_dB) -/* Meter Control attributes */ -/** return RMS signal level */ -#define HPI_METER_RMS HPI_CTL_ATTR0(METER, 1) -/** return peak signal level */ -#define HPI_METER_PEAK HPI_CTL_ATTR0(METER, 2) -/** ballistics for ALL rms meters on adapter */ -#define HPI_METER_RMS_BALLISTICS HPI_CTL_ATTR0(METER, 3) -/** ballistics for ALL peak meters on adapter */ -#define HPI_METER_PEAK_BALLISTICS HPI_CTL_ATTR0(METER, 4) -/** For HPI_ControlQuery() to get the number of channels of a meter control*/ -#define HPI_METER_NUM_CHANNELS HPI_CTL_ATTR0(METER, 5) - -/* Multiplexer control attributes */ -#define HPI_MULTIPLEXER_SOURCE HPI_CTL_ATTR0(MULTIPLEXER, 1) -#define HPI_MULTIPLEXER_QUERYSOURCE HPI_CTL_ATTR0(MULTIPLEXER, 2) - -/** AES/EBU transmitter control attributes */ -/** AESEBU or SPDIF */ -#define HPI_AESEBUTX_FORMAT HPI_CTL_ATTR0(AESEBUTX, 1) -#define HPI_AESEBUTX_SAMPLERATE HPI_CTL_ATTR0(AESEBUTX, 3) -#define HPI_AESEBUTX_CHANNELSTATUS HPI_CTL_ATTR0(AESEBUTX, 4) -#define HPI_AESEBUTX_USERDATA HPI_CTL_ATTR0(AESEBUTX, 5) - -/** AES/EBU receiver control attributes */ -#define HPI_AESEBURX_FORMAT HPI_CTL_ATTR0(AESEBURX, 1) -#define HPI_AESEBURX_ERRORSTATUS HPI_CTL_ATTR0(AESEBURX, 2) -#define HPI_AESEBURX_SAMPLERATE HPI_CTL_ATTR0(AESEBURX, 3) -#define HPI_AESEBURX_CHANNELSTATUS HPI_CTL_ATTR0(AESEBURX, 4) -#define HPI_AESEBURX_USERDATA HPI_CTL_ATTR0(AESEBURX, 5) - -/* NOTE: old defs only make sense when rx & tx not unique */ -#define HPI_AESEBU_FORMAT 1 -#define HPI_AESEBU_ERRORSTATUS 2 -#define HPI_AESEBU_SAMPLERATE 3 -#define HPI_AESEBU_CHANNELSTATUS 4 -#define HPI_AESEBU_USERDATA 5 +/* autofade profiles */ +/** log fade - dB attenuation changes linearly over time */ +#define HPI_VOLUME_AUTOFADE_LOG 2 +/** linear fade - amplitude changes linearly */ +#define HPI_VOLUME_AUTOFADE_LINEAR 3 /** AES/EBU physical format - AES/EBU balanced "professional" */ #define HPI_AESEBU_FORMAT_AESEBU 1 @@ -1000,64 +818,24 @@ enum HPI_MIXER_STORE_COMMAND { /** \defgroup tuner_defs Tuners \{ */ -/** \defgroup tuner_attrs Tuner control attributes -\{ -*/ -#define HPI_TUNER_BAND HPI_CTL_ATTR0(TUNER, 1) -#define HPI_TUNER_FREQ HPI_CTL_ATTR0(TUNER, 2) -#define HPI_TUNER_LEVEL HPI_CTL_ATTR0(TUNER, 3) -#define HPI_TUNER_AUDIOMUTE HPI_CTL_ATTR0(TUNER, 4) -/* use TUNER_STATUS instead */ -#define HPI_TUNER_VIDEO_STATUS HPI_CTL_ATTR0(TUNER, 5) -#define HPI_TUNER_GAIN HPI_CTL_ATTR0(TUNER, 6) -#define HPI_TUNER_STATUS HPI_CTL_ATTR0(TUNER, 7) -#define HPI_TUNER_MODE HPI_CTL_ATTR0(TUNER, 8) -/** RDS data. */ -#define HPI_TUNER_RDS HPI_CTL_ATTR0(TUNER, 9) -/** Audio pre-emphasis. */ -#define HPI_TUNER_DEEMPHASIS HPI_CTL_ATTR(TUNER, 10) -/** HD-Radio tuner program control. */ -#define HPI_TUNER_PROGRAM HPI_CTL_ATTR(TUNER, 11) -/** HD-Radio tuner digital signal quality. */ -#define HPI_TUNER_HDRADIO_SIGNAL_QUALITY HPI_CTL_ATTR(TUNER, 12) -/** HD-Radio SDK firmware version. */ -#define HPI_TUNER_HDRADIO_SDK_VERSION HPI_CTL_ATTR(TUNER, 13) -/** HD-Radio DSP firmware version. */ -#define HPI_TUNER_HDRADIO_DSP_VERSION HPI_CTL_ATTR(TUNER, 14) -/** \} */ /** \defgroup pads_attrs Tuner PADs control attributes \{ */ /** The text string containing the station/channel combination. */ -#define HPI_PAD_CHANNEL_NAME HPI_CTL_ATTR(PAD, 1) #define HPI_PAD_CHANNEL_NAME_LEN 16 /** The text string containing the artist. */ -#define HPI_PAD_ARTIST HPI_CTL_ATTR(PAD, 2) #define HPI_PAD_ARTIST_LEN 64 /** The text string containing the title. */ -#define HPI_PAD_TITLE HPI_CTL_ATTR(PAD, 3) #define HPI_PAD_TITLE_LEN 64 /** The text string containing the comment. */ -#define HPI_PAD_COMMENT HPI_CTL_ATTR(PAD, 4) #define HPI_PAD_COMMENT_LEN 256 -/** The integer containing the PTY code. */ -#define HPI_PAD_PROGRAM_TYPE HPI_CTL_ATTR(PAD, 5) -/** The integer containing the program identification. */ -#define HPI_PAD_PROGRAM_ID HPI_CTL_ATTR(PAD, 6) -/** The integer containing whether traffic information is supported. -Contains either 1 or 0. */ -#define HPI_PAD_TA_SUPPORT HPI_CTL_ATTR(PAD, 7) -/** The integer containing whether traffic announcement is in progress. -Contains either 1 or 0. */ -#define HPI_PAD_TA_ACTIVE HPI_CTL_ATTR(PAD, 8) /** Data types for PTY string translation. */ enum eHPI_RDS_type { - HPI_RDS_DATATYPE_RDS = 0, /**< RDS bitstream. */ - HPI_RDS_DATATYPE_RDBS = 1, /**< RDBS bitstream. */ - HPI_RDS_DATATYPE_FUTURE = 2 /**< Future bitstream. */ + HPI_RDS_DATATYPE_RDS = 0, /**< RDS bitstream.*/ + HPI_RDS_DATATYPE_RBDS = 1 /**< RBDS bitstream.*/ }; /** \} */ @@ -1107,12 +885,6 @@ Multiple fields are returned from a single call. /** \} */ /** \} */ -/* VOX control attributes */ -#define HPI_VOX_THRESHOLD HPI_CTL_ATTR0(VOX, 1) - -/*?? channel mode used hpi_multiplexer_source attribute == 1 */ -#define HPI_CHANNEL_MODE_MODE HPI_CTL_ATTR0(CHANNEL_MODE, 1) - /** \defgroup channel_modes Channel Modes Used for HPI_ChannelModeSet/Get() \{ @@ -1134,29 +906,6 @@ Used for HPI_ChannelModeSet/Get() #define HPI_CHANNEL_MODE_LAST 6 /** \} */ -/* Bitstream control set attributes */ -#define HPI_BITSTREAM_DATA_POLARITY HPI_CTL_ATTR0(BITSTREAM, 1) -#define HPI_BITSTREAM_CLOCK_EDGE HPI_CTL_ATTR0(BITSTREAM, 2) -#define HPI_BITSTREAM_CLOCK_SOURCE HPI_CTL_ATTR0(BITSTREAM, 3) - -#define HPI_POLARITY_POSITIVE 0 -#define HPI_POLARITY_NEGATIVE 1 - -/* -Currently fixed at EXTERNAL -#define HPI_SOURCE_EXTERNAL 0 -#define HPI_SOURCE_INTERNAL 1 -*/ - -/* Bitstream control get attributes */ -#define HPI_BITSTREAM_ACTIVITY 1 - -/* SampleClock control attributes */ -#define HPI_SAMPLECLOCK_SOURCE HPI_CTL_ATTR0(SAMPLECLOCK, 1) -#define HPI_SAMPLECLOCK_SAMPLERATE HPI_CTL_ATTR0(SAMPLECLOCK, 2) -#define HPI_SAMPLECLOCK_SOURCE_INDEX HPI_CTL_ATTR0(SAMPLECLOCK, 3) -#define HPI_SAMPLECLOCK_LOCAL_SAMPLERATE HPI_CTL_ATTR0(SAMPLECLOCK, 4) - /** \defgroup sampleclock_source SampleClock source values \{ */ @@ -1175,7 +924,9 @@ Currently fixed at EXTERNAL #define HPI_SAMPLECLOCK_SOURCE_SMPTE 5 /** One of the aesebu inputs */ #define HPI_SAMPLECLOCK_SOURCE_AESEBU_INPUT 6 -/** The first aesebu input with a valid signal */ +/** \deprecated The first aesebu input with a valid signal +Superseded by separate Auto enable flag +*/ #define HPI_SAMPLECLOCK_SOURCE_AESEBU_AUTO 7 /** From a network interface e.g. Cobranet or Livewire at either 48 or 96kHz */ #define HPI_SAMPLECLOCK_SOURCE_NETWORK 8 @@ -1185,18 +936,6 @@ Currently fixed at EXTERNAL #define HPI_SAMPLECLOCK_SOURCE_LAST 10 /** \} */ -/* Microphone control attributes */ -#define HPI_MICROPHONE_PHANTOM_POWER HPI_CTL_ATTR0(MICROPHONE, 1) - -/** Equalizer control attributes -*/ -/** Used to get number of filters in an EQ. (Can't set) */ -#define HPI_EQUALIZER_NUM_FILTERS HPI_CTL_ATTR0(EQUALIZER, 1) -/** Set/get the filter by type, freq, Q, gain */ -#define HPI_EQUALIZER_FILTER HPI_CTL_ATTR0(EQUALIZER, 2) -/** Get the biquad coefficients */ -#define HPI_EQUALIZER_COEFFICIENTS HPI_CTL_ATTR0(EQUALIZER, 3) - /** Equalizer filter types. Used by HPI_ParametricEQ_SetBand() */ enum HPI_FILTER_TYPE { HPI_FILTER_TYPE_BYPASS = 0, /**< Filter is turned off */ @@ -1211,129 +950,6 @@ enum HPI_FILTER_TYPE { HPI_FILTER_TYPE_BANDSTOP = 7 /**< Standard band stop/notch */ }; -#define HPI_COMPANDER_PARAMS HPI_CTL_ATTR(COMPANDER, 1) - -/* Cobranet control attributes. - MUST be distinct from all other control attributes. - This is so that host side processing can easily identify a Cobranet control - and apply additional host side operations (like copying data) as required. -*/ -#define HPI_COBRANET_SET HPI_CTL_ATTR(COBRANET, 1) -#define HPI_COBRANET_GET HPI_CTL_ATTR(COBRANET, 2) -#define HPI_COBRANET_SET_DATA HPI_CTL_ATTR(COBRANET, 3) -#define HPI_COBRANET_GET_DATA HPI_CTL_ATTR(COBRANET, 4) -#define HPI_COBRANET_GET_STATUS HPI_CTL_ATTR(COBRANET, 5) -#define HPI_COBRANET_SEND_PACKET HPI_CTL_ATTR(COBRANET, 6) -#define HPI_COBRANET_GET_PACKET HPI_CTL_ATTR(COBRANET, 7) - -/*------------------------------------------------------------ - Cobranet Chip Bridge - copied from HMI.H -------------------------------------------------------------*/ -#define HPI_COBRANET_HMI_cobraBridge 0x20000 -#define HPI_COBRANET_HMI_cobraBridgeTxPktBuf \ - (HPI_COBRANET_HMI_cobraBridge + 0x1000) -#define HPI_COBRANET_HMI_cobraBridgeRxPktBuf \ - (HPI_COBRANET_HMI_cobraBridge + 0x2000) -#define HPI_COBRANET_HMI_cobraIfTable1 0x110000 -#define HPI_COBRANET_HMI_cobraIfPhyAddress \ - (HPI_COBRANET_HMI_cobraIfTable1 + 0xd) -#define HPI_COBRANET_HMI_cobraProtocolIP 0x72000 -#define HPI_COBRANET_HMI_cobraIpMonCurrentIP \ - (HPI_COBRANET_HMI_cobraProtocolIP + 0x0) -#define HPI_COBRANET_HMI_cobraIpMonStaticIP \ - (HPI_COBRANET_HMI_cobraProtocolIP + 0x2) -#define HPI_COBRANET_HMI_cobraSys 0x100000 -#define HPI_COBRANET_HMI_cobraSysDesc \ - (HPI_COBRANET_HMI_cobraSys + 0x0) -#define HPI_COBRANET_HMI_cobraSysObjectID \ - (HPI_COBRANET_HMI_cobraSys + 0x100) -#define HPI_COBRANET_HMI_cobraSysContact \ - (HPI_COBRANET_HMI_cobraSys + 0x200) -#define HPI_COBRANET_HMI_cobraSysName \ - (HPI_COBRANET_HMI_cobraSys + 0x300) -#define HPI_COBRANET_HMI_cobraSysLocation \ - (HPI_COBRANET_HMI_cobraSys + 0x400) - -/*------------------------------------------------------------ - Cobranet Chip Status bits -------------------------------------------------------------*/ -#define HPI_COBRANET_HMI_STATUS_RXPACKET 2 -#define HPI_COBRANET_HMI_STATUS_TXPACKET 3 - -/*------------------------------------------------------------ - Ethernet header size -------------------------------------------------------------*/ -#define HPI_ETHERNET_HEADER_SIZE (16) - -/* These defines are used to fill in protocol information for an Ethernet packet - sent using HMI on CS18102 */ -/** ID supplied by Cirrius for ASI packets. */ -#define HPI_ETHERNET_PACKET_ID 0x85 -/** Simple packet - no special routing required */ -#define HPI_ETHERNET_PACKET_V1 0x01 -/** This packet must make its way to the host across the HPI interface */ -#define HPI_ETHERNET_PACKET_HOSTED_VIA_HMI 0x20 -/** This packet must make its way to the host across the HPI interface */ -#define HPI_ETHERNET_PACKET_HOSTED_VIA_HMI_V1 0x21 -/** This packet must make its way to the host across the HPI interface */ -#define HPI_ETHERNET_PACKET_HOSTED_VIA_HPI 0x40 -/** This packet must make its way to the host across the HPI interface */ -#define HPI_ETHERNET_PACKET_HOSTED_VIA_HPI_V1 0x41 - -#define HPI_ETHERNET_UDP_PORT (44600) /*!< UDP messaging port */ - -/** Base network time out is set to 100 milli-seconds. */ -#define HPI_ETHERNET_TIMEOUT_MS (100) - -/** \defgroup tonedet_attr Tonedetector attributes -\{ -Used by HPI_ToneDetector_Set() and HPI_ToneDetector_Get() -*/ - -/** Set the threshold level of a tonedetector, -Threshold is a -ve number in units of dB/100, -*/ -#define HPI_TONEDETECTOR_THRESHOLD HPI_CTL_ATTR(TONEDETECTOR, 1) - -/** Get the current state of tonedetection -The result is a bitmap of detected tones. pairs of bits represent the left -and right channels, with left channel in LSB. -The lowest frequency detector state is in the LSB -*/ -#define HPI_TONEDETECTOR_STATE HPI_CTL_ATTR(TONEDETECTOR, 2) - -/** Get the frequency of a tonedetector band. -*/ -#define HPI_TONEDETECTOR_FREQUENCY HPI_CTL_ATTR(TONEDETECTOR, 3) - -/**\}*/ - -/** \defgroup silencedet_attr SilenceDetector attributes -\{ -*/ - -/** Get the current state of tonedetection -The result is a bitmap with 1s for silent channels. Left channel is in LSB -*/ -#define HPI_SILENCEDETECTOR_STATE \ - HPI_CTL_ATTR(SILENCEDETECTOR, 2) - -/** Set the threshold level of a SilenceDetector, -Threshold is a -ve number in units of dB/100, -*/ -#define HPI_SILENCEDETECTOR_THRESHOLD \ - HPI_CTL_ATTR(SILENCEDETECTOR, 1) - -/** get/set the silence time before the detector triggers -*/ -#define HPI_SILENCEDETECTOR_DELAY \ - HPI_CTL_ATTR(SILENCEDETECTOR, 3) - -/**\}*/ - -/******************************************* CONTROLX ATTRIBUTES ****/ -/* NOTE: All controlx attributes must be unique, unlike control attributes */ -/******************************************* ASYNC ATTRIBUTES ****/ /** \defgroup async_event Async Event sources \{ */ @@ -1505,6 +1121,8 @@ to ASI if further capabilities updates are required */ #define HPI_ERROR_I2C_MISSING_ACK HPI_ERROR_CONTROL_I2C_MISSING_ACK #define HPI_ERROR_I2C_BAD_ADR 460 +/* AES18 specific errors were 500..507 */ + /** custom error to use for debugging */ #define HPI_ERROR_CUSTOM 600 @@ -1565,17 +1183,6 @@ struct hpi_anc_frame { u8 bData[HPI_MAX_ANC_BYTES_PER_FRAME]; }; -struct hpi_punchinout { - u32 dwPunchInSample; - u32 dwPunchOutSample; -}; - -struct hpi_streamid { - u16 wObjectType; - /**< Type of object, HPI_OBJ_OSTREAM or HPI_OBJ_ISTREAM. */ - u16 wStreamIndex; /**< OStream or IStream index. */ -}; - /** An object for containing a single async event. */ struct hpi_async_event { @@ -1847,7 +1454,7 @@ u16 HPI_GpioReadBit( u16 HPI_GpioReadAllBits( struct hpi_hsubsys *phSubSys, u32 hGpio, - u16 *pwBitData + u16 *pwAllBitData ); u16 HPI_GpioWriteBit( @@ -1857,6 +1464,12 @@ u16 HPI_GpioWriteBit( u16 wBitData ); +u16 HPI_GpioWriteStatus( + struct hpi_hsubsys *phSubSys, + u32 hGpio, + u16 *pwAllBitData +); + /**********************/ /* Async Event Object */ /**********************/ @@ -2195,41 +1808,6 @@ u16 HPI_MixerStore( /*************************/ /* mixer CONTROLS */ /*************************/ - -/* Generic query of available control settings */ -u16 HPI_ControlQuery( - const struct hpi_hsubsys *phSubSys, - const u32 hControl, - const u16 wAttrib, - const u32 dwIndex, - const u32 dwParam, - u32 *pdwSetting -); - -#ifndef SWIG -/* Generic setting of control attribute value */ -u16 HPI_ControlParamSet( - const struct hpi_hsubsys *phSubSys, - const u32 hControl, - const u16 wAttrib, - const u32 dwParam1, - const u32 dwParam2 -); - -/* generic getting of control attribute value. - Null pointers allowed for return values -*/ -u16 HPI_ControlParamGet( - const struct hpi_hsubsys *phSubSys, - const u32 hControl, - const u16 wAttrib, - u32 dwParam1, - u32 dwParam2, - u32 *pdwParam1, - u32 *pdwParam2 -); -#endif - /*************************/ /* volume control */ /*************************/ @@ -2254,6 +1832,12 @@ u16 HPI_VolumeQueryRange( short *nStepGain_01dB ); +u16 HPI_Volume_QueryChannels( + const struct hpi_hsubsys *phSubSys, + const u32 hVolume, + u32 *pChannels +); + u16 HPI_VolumeAutoFade( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2270,8 +1854,16 @@ u16 HPI_VolumeAutoFadeProfile( ); /*************************/ -/* level control */ +/* level control */ /*************************/ +u16 HPI_LevelQueryRange( + struct hpi_hsubsys *phSubSys, + u32 hControl, + short *nMinGain_01dB, + short *nMaxGain_01dB, + short *nStepGain_01dB +); + u16 HPI_LevelSetGain( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2287,6 +1879,12 @@ u16 HPI_LevelGetGain( /*************************/ /* meter control */ /*************************/ +u16 HPI_Meter_QueryChannels( + const struct hpi_hsubsys *phSubSys, + const u32 hMeter, + u32 *pChannels +); + u16 HPI_MeterGetPeak( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2330,6 +1928,13 @@ u16 HPI_MeterGetRmsBallistics( /*************************/ /* channel mode control */ /*************************/ +u16 HPI_ChannelMode_QueryMode( + const struct hpi_hsubsys *phSubSys, + const u32 hMode, + const u32 dwIndex, + u16 *pwMode +); + u16 HPI_ChannelModeSet( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2345,6 +1950,13 @@ u16 HPI_ChannelModeGet( /*************************/ /* Tuner control */ /*************************/ +u16 HPI_Tuner_QueryBand( + const struct hpi_hsubsys *phSubSys, + const u32 hTuner, + const u32 dwIndex, + u16 *pwBand +); + u16 HPI_Tuner_SetBand( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2357,6 +1969,14 @@ u16 HPI_Tuner_GetBand( u16 *pwBand ); +u16 HPI_Tuner_QueryFrequency( + const struct hpi_hsubsys *phSubSys, + const u32 hTuner, + const u32 dwIndex, + const u16 band, + u32 *pdwFreq +); + u16 HPI_Tuner_SetFrequency( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2381,6 +2001,13 @@ u16 HPI_Tuner_GetRawRFLevel( short *pwLevel ); +u16 HPI_Tuner_QueryGain( + const struct hpi_hsubsys *phSubSys, + const u32 hTuner, + const u32 dwIndex, + u16 *pwGain +); + u16 HPI_Tuner_SetGain( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2419,6 +2046,15 @@ u16 HPI_Tuner_GetRDS( u32 hControl, char *pRdsData ); + +u16 HPI_Tuner_QueryDeemphasis( + const struct hpi_hsubsys *phSubSys, + const u32 hTuner, + const u32 dwIndex, + const u16 band, + u32 *pdwDeemphasis +); + u16 HPI_Tuner_SetDeemphasis( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2429,22 +2065,32 @@ u16 HPI_Tuner_GetDeemphasis( u32 hControl, u32 *pdwDeemphasis ); + +u16 HPI_Tuner_QueryProgram( + const struct hpi_hsubsys *phSubSys, + const u32 hTuner, + u32 *pbitmapProgram +); + u16 HPI_Tuner_SetProgram( struct hpi_hsubsys *phSubSys, u32 hControl, u32 dwProgram ); + u16 HPI_Tuner_GetProgram( struct hpi_hsubsys *phSubSys, u32 hControl, u32 *pdwProgram ); + u16 HPI_Tuner_GetHdRadioDspVersion( struct hpi_hsubsys *phSubSys, u32 hControl, char *pszDspVersion, const u32 dwStringSize ); + u16 HPI_Tuner_GetHdRadioSdkVersion( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2514,6 +2160,13 @@ u16 HPI_PAD_GetProgramTypeString( /****************************/ /* AES/EBU Receiver control */ /****************************/ +u16 HPI_AESEBU_Receiver_QueryFormat( + const struct hpi_hsubsys *phSubSys, + const u32 hAesRx, + const u32 dwIndex, + u16 *pwFormat +); + u16 HPI_AESEBU_Receiver_SetFormat( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2540,16 +2193,14 @@ u16 HPI_AESEBU_Receiver_GetUserData( ); u16 HPI_AESEBU_Receiver_GetChannelStatus( - struct hpi_hsubsys - *phSubSys, + struct hpi_hsubsys *phSubSys, u32 hControl, u16 wIndex, u16 *pwData ); u16 HPI_AESEBU_Receiver_GetErrorStatus( - struct hpi_hsubsys - *phSubSys, + struct hpi_hsubsys *phSubSys, u32 hControl, u16 *pwErrorData ); @@ -2558,36 +2209,39 @@ u16 HPI_AESEBU_Receiver_GetErrorStatus( /* AES/EBU Transmitter control */ /*******************************/ u16 HPI_AESEBU_Transmitter_SetSampleRate( - struct hpi_hsubsys - *phSubSys, + struct hpi_hsubsys *phSubSys, u32 hControl, u32 dwSampleRate ); u16 HPI_AESEBU_Transmitter_SetUserData( - struct hpi_hsubsys - *phSubSys, + struct hpi_hsubsys *phSubSys, u32 hControl, u16 wIndex, u16 wData ); u16 HPI_AESEBU_Transmitter_SetChannelStatus( - struct hpi_hsubsys - *phSubSys, + struct hpi_hsubsys *phSubSys, u32 hControl, u16 wIndex, u16 wData ); u16 HPI_AESEBU_Transmitter_GetChannelStatus( - struct hpi_hsubsys - *phSubSys, + struct hpi_hsubsys *phSubSys, u32 hControl, u16 wIndex, u16 *pwData ); +u16 HPI_AESEBU_Transmitter_QueryFormat( + const struct hpi_hsubsys *phSubSys, + const u32 hAesTx, + const u32 dwIndex, + u16 *pwFormat +); + u16 HPI_AESEBU_Transmitter_SetFormat( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2665,6 +2319,14 @@ u16 HPI_Bitstream_GetActivity( /***********************/ /* SampleClock control */ /***********************/ + +u16 HPI_SampleClock_QuerySource( + const struct hpi_hsubsys *phSubSys, + const u32 hClock, + const u32 dwIndex, + u16 *pwSource +); + u16 HPI_SampleClock_SetSource( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2677,6 +2339,14 @@ u16 HPI_SampleClock_GetSource( u16 *pwSource ); +u16 HPI_SampleClock_QuerySourceIndex( + const struct hpi_hsubsys *phSubSys, + const u32 hClock, + const u32 dwIndex, + const u32 dwSource, + u16 *pwSourceIndex +); + u16 HPI_SampleClock_SetSourceIndex( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2695,6 +2365,13 @@ u16 HPI_SampleClock_GetSampleRate( u32 *pdwSampleRate ); +u16 HPI_SampleClock_QueryLocalRate( + const struct hpi_hsubsys *phSubSys, + const u32 hClock, + const u32 dwIndex, + u32 *pdwSource +); + u16 HPI_SampleClock_SetLocalRate( struct hpi_hsubsys *phSubSys, u32 hControl, @@ -2707,6 +2384,18 @@ u16 HPI_SampleClock_GetLocalRate( u32 *pdwSampleRate ); +u16 HPI_SampleClock_SetAuto( + struct hpi_hsubsys *phSubSys, + u32 hControl, + u32 dwAuto +); + +u16 HPI_SampleClock_GetAuto( + struct hpi_hsubsys *phSubSys, + u32 hControl, + u32 *pdwAuto +); + /***********************/ /* Microphone control */ /***********************/ @@ -3051,978 +2740,6 @@ u16 HPI_FormatCreate( /* Until it's verified, this function is for Windows OSs only */ -/*****************************************************************************/ -/*****************************************************************************/ -/******** HPI LOW LEVEL MESSAGES *******/ -/*****************************************************************************/ -/*****************************************************************************/ -/** Pnp ids */ -/** "ASI" - actual is "ASX" - need to change */ -#define HPI_ID_ISAPNP_AUDIOSCIENCE 0x0669 -/** PCI vendor ID that AudioScience uses */ -#define HPI_PCI_VENDOR_ID_AUDIOSCIENCE 0x175C -/** PCI vendor ID that the DSP56301 has */ -#define HPI_PCI_VENDOR_ID_MOTOROLA 0x1057 -/** PCI vendor ID that TI uses */ -#define HPI_PCI_VENDOR_ID_TI 0x104C - -#define HPI_USB_VENDOR_ID_AUDIOSCIENCE 0x1257 -#define HPI_USB_W2K_TAG 0x57495341 /* "ASIW" */ -#define HPI_USB_LINUX_TAG 0x4C495341 /* "ASIL" */ - -/******************************************* message types */ -#define HPI_TYPE_MESSAGE 1 -#define HPI_TYPE_RESPONSE 2 -#define HPI_TYPE_DATA 3 -#define HPI_TYPE_SSX2BYPASS_MESSAGE 4 - -/******************************************* object types */ -#define HPI_OBJ_SUBSYSTEM 1 -#define HPI_OBJ_ADAPTER 2 -#define HPI_OBJ_OSTREAM 3 -#define HPI_OBJ_ISTREAM 4 -#define HPI_OBJ_MIXER 5 -#define HPI_OBJ_NODE 6 -#define HPI_OBJ_CONTROL 7 -#define HPI_OBJ_NVMEMORY 8 -#define HPI_OBJ_GPIO 9 -#define HPI_OBJ_WATCHDOG 10 -#define HPI_OBJ_CLOCK 11 -#define HPI_OBJ_PROFILE 12 -#define HPI_OBJ_CONTROLEX 13 -#define HPI_OBJ_ASYNCEVENT 14 - -#define HPI_OBJ_MAXINDEX 14 - -/******************************************* methods/functions */ - -#define HPI_OBJ_FUNCTION_SPACING 0x100 -#define HPI_MAKE_INDEX(obj, index) (obj * HPI_OBJ_FUNCTION_SPACING + index) -#define HPI_EXTRACT_INDEX(fn) (fn & 0xff) - -/* SUB-SYSTEM */ -#define HPI_SUBSYS_OPEN HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 1) -#define HPI_SUBSYS_GET_VERSION HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 2) -#define HPI_SUBSYS_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 3) -#define HPI_SUBSYS_FIND_ADAPTERS HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 4) -#define HPI_SUBSYS_CREATE_ADAPTER HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 5) -#define HPI_SUBSYS_CLOSE HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 6) -#define HPI_SUBSYS_DELETE_ADAPTER HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 7) -#define HPI_SUBSYS_DRIVER_LOAD HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 8) -#define HPI_SUBSYS_DRIVER_UNLOAD HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 9) - /*SGT*/ -#define HPI_SUBSYS_READ_PORT_8 HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 10) -#define HPI_SUBSYS_WRITE_PORT_8 HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 11) -#define HPI_SUBSYS_GET_NUM_ADAPTERS HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 12) -#define HPI_SUBSYS_GET_ADAPTER HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 13) -#define HPI_SUBSYS_SET_NETWORK_INTERFACE HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 14) -/* ADAPTER */ -#define HPI_ADAPTER_OPEN HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 1) -#define HPI_ADAPTER_CLOSE HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 2) -#define HPI_ADAPTER_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 3) -#define HPI_ADAPTER_GET_ASSERT HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 4) -#define HPI_ADAPTER_TEST_ASSERT HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 5) -#define HPI_ADAPTER_SET_MODE HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 6) -#define HPI_ADAPTER_GET_MODE HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 7) -#define HPI_ADAPTER_ENABLE_CAPABILITY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 8) -#define HPI_ADAPTER_SELFTEST HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 9) -#define HPI_ADAPTER_FIND_OBJECT HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 10) -#define HPI_ADAPTER_QUERY_FLASH HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 11) -#define HPI_ADAPTER_START_FLASH HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 12) -#define HPI_ADAPTER_PROGRAM_FLASH HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 13) -#define HPI_ADAPTER_SET_PROPERTY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 14) -#define HPI_ADAPTER_GET_PROPERTY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 15) -#define HPI_ADAPTER_ENUM_PROPERTY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 16) -#define HPI_ADAPTER_MODULE_INFO HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 17) -#define HPI_ADAPTER_DEBUG_READ HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 18) -#define HPI_ADAPTER_FUNCTION_COUNT 18 -/* OUTPUT STREAM */ -#define HPI_OSTREAM_OPEN HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 1) -#define HPI_OSTREAM_CLOSE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 2) -#define HPI_OSTREAM_WRITE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 3) -#define HPI_OSTREAM_START HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 4) -#define HPI_OSTREAM_STOP HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 5) -#define HPI_OSTREAM_RESET HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 6) -#define HPI_OSTREAM_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 7) -#define HPI_OSTREAM_QUERY_FORMAT HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 8) -#define HPI_OSTREAM_DATA HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 9) -#define HPI_OSTREAM_SET_VELOCITY HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 10) -#define HPI_OSTREAM_SET_PUNCHINOUT HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 11) -#define HPI_OSTREAM_SINEGEN HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 12) -#define HPI_OSTREAM_ANC_RESET HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 13) -#define HPI_OSTREAM_ANC_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 14) -#define HPI_OSTREAM_ANC_READ HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 15) -#define HPI_OSTREAM_SET_TIMESCALE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 16) -#define HPI_OSTREAM_SET_FORMAT HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 17) -#define HPI_OSTREAM_HOSTBUFFER_ALLOC HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 18) -#define HPI_OSTREAM_HOSTBUFFER_FREE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 19) -#define HPI_OSTREAM_GROUP_ADD HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 20) -#define HPI_OSTREAM_GROUP_GETMAP HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 21) -#define HPI_OSTREAM_GROUP_RESET HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 22) -#define HPI_OSTREAM_FUNCTION_COUNT (22) -/* INPUT STREAM */ -#define HPI_ISTREAM_OPEN HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 1) -#define HPI_ISTREAM_CLOSE HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 2) -#define HPI_ISTREAM_SET_FORMAT HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 3) -#define HPI_ISTREAM_READ HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 4) -#define HPI_ISTREAM_START HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 5) -#define HPI_ISTREAM_STOP HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 6) -#define HPI_ISTREAM_RESET HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 7) -#define HPI_ISTREAM_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 8) -#define HPI_ISTREAM_QUERY_FORMAT HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 9) -#define HPI_ISTREAM_ANC_RESET HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 10) -#define HPI_ISTREAM_ANC_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 11) -#define HPI_ISTREAM_ANC_WRITE HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 12) -#define HPI_ISTREAM_HOSTBUFFER_ALLOC HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 13) -#define HPI_ISTREAM_HOSTBUFFER_FREE HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 14) -#define HPI_ISTREAM_GROUP_ADD HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 15) -#define HPI_ISTREAM_GROUP_GETMAP HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 16) -#define HPI_ISTREAM_GROUP_RESET HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 17) -#define HPI_ISTREAM_FUNCTION_COUNT (17) -/* MIXER */ -/* NOTE: - GET_NODE_INFO, SET_CONNECTION, GET_CONNECTIONS are not currently used */ -#define HPI_MIXER_OPEN HPI_MAKE_INDEX(HPI_OBJ_MIXER, 1) -#define HPI_MIXER_CLOSE HPI_MAKE_INDEX(HPI_OBJ_MIXER, 2) -#define HPI_MIXER_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_MIXER, 3) -#define HPI_MIXER_GET_NODE_INFO HPI_MAKE_INDEX(HPI_OBJ_MIXER, 4) -#define HPI_MIXER_GET_CONTROL HPI_MAKE_INDEX(HPI_OBJ_MIXER, 5) -#define HPI_MIXER_SET_CONNECTION HPI_MAKE_INDEX(HPI_OBJ_MIXER, 6) -#define HPI_MIXER_GET_CONNECTIONS HPI_MAKE_INDEX(HPI_OBJ_MIXER, 7) -#define HPI_MIXER_GET_CONTROL_BY_INDEX HPI_MAKE_INDEX(HPI_OBJ_MIXER, 8) -#define HPI_MIXER_GET_CONTROL_ARRAY_BY_INDEX HPI_MAKE_INDEX(HPI_OBJ_MIXER, 9) -#define HPI_MIXER_GET_CONTROL_MULTIPLE_VALUES HPI_MAKE_INDEX(HPI_OBJ_MIXER, 10) -#define HPI_MIXER_STORE HPI_MAKE_INDEX(HPI_OBJ_MIXER, 11) -#define HPI_MIXER_FUNCTION_COUNT 11 -/* MIXER CONTROLS */ -#define HPI_CONTROL_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_CONTROL, 1) -#define HPI_CONTROL_GET_STATE HPI_MAKE_INDEX(HPI_OBJ_CONTROL, 2) -#define HPI_CONTROL_SET_STATE HPI_MAKE_INDEX(HPI_OBJ_CONTROL, 3) -#define HPI_CONTROL_FUNCTION_COUNT 3 -/* NONVOL MEMORY */ -#define HPI_NVMEMORY_OPEN HPI_MAKE_INDEX(HPI_OBJ_NVMEMORY, 1) -#define HPI_NVMEMORY_READ_BYTE HPI_MAKE_INDEX(HPI_OBJ_NVMEMORY, 2) -#define HPI_NVMEMORY_WRITE_BYTE HPI_MAKE_INDEX(HPI_OBJ_NVMEMORY, 3) -#define HPI_NVMEMORY_FUNCTION_COUNT 3 -/* GPIO */ -#define HPI_GPIO_OPEN HPI_MAKE_INDEX(HPI_OBJ_GPIO, 1) -#define HPI_GPIO_READ_BIT HPI_MAKE_INDEX(HPI_OBJ_GPIO, 2) -#define HPI_GPIO_WRITE_BIT HPI_MAKE_INDEX(HPI_OBJ_GPIO, 3) -#define HPI_GPIO_READ_ALL HPI_MAKE_INDEX(HPI_OBJ_GPIO, 4) -#define HPI_GPIO_FUNCTION_COUNT 4 -/* ASYNC EVENT */ -#define HPI_ASYNCEVENT_OPEN HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 1) -#define HPI_ASYNCEVENT_CLOSE HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 2) -#define HPI_ASYNCEVENT_WAIT HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 3) -#define HPI_ASYNCEVENT_GETCOUNT HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 4) -#define HPI_ASYNCEVENT_GET HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 5) -#define HPI_ASYNCEVENT_SENDEVENTS HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 6) -#define HPI_ASYNCEVENT_FUNCTION_COUNT 6 -/* WATCH-DOG */ -#define HPI_WATCHDOG_OPEN HPI_MAKE_INDEX(HPI_OBJ_WATCHDOG, 1) -#define HPI_WATCHDOG_SET_TIME HPI_MAKE_INDEX(HPI_OBJ_WATCHDOG, 2) -#define HPI_WATCHDOG_PING HPI_MAKE_INDEX(HPI_OBJ_WATCHDOG, 3) -/* CLOCK */ -#define HPI_CLOCK_OPEN HPI_MAKE_INDEX(HPI_OBJ_CLOCK, 1) -#define HPI_CLOCK_SET_TIME HPI_MAKE_INDEX(HPI_OBJ_CLOCK, 2) -#define HPI_CLOCK_GET_TIME HPI_MAKE_INDEX(HPI_OBJ_CLOCK, 3) -/* PROFILE */ -#define HPI_PROFILE_OPEN_ALL HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 1) -#define HPI_PROFILE_START_ALL HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 2) -#define HPI_PROFILE_STOP_ALL HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 3) -#define HPI_PROFILE_GET HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 4) -#define HPI_PROFILE_GET_IDLECOUNT HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 5) -#define HPI_PROFILE_GET_NAME HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 6) -#define HPI_PROFILE_GET_UTILIZATION HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 7) -#define HPI_PROFILE_FUNCTION_COUNT 7 -/* ////////////////////////////////////////////////////////////////////// */ -/* STRUCTURES */ -#ifndef DISABLE_PRAGMA_PACK1 -#pragma pack(push, 1) -#endif -/** PCI bus resource */ - struct hpi_pci { - u32 __iomem *apMemBase[HPI_MAX_ADAPTER_MEM_SPACES]; - struct pci_dev *pOsData; - -#ifndef HPI64BIT /* keep structure size constant */ - u32 dwPadding[HPI_MAX_ADAPTER_MEM_SPACES + 1]; -#endif - u16 wVendorId; - u16 wDeviceId; - u16 wSubSysVendorId; - u16 wSubSysDeviceId; - u16 wBusNumber; - u16 wDeviceNumber; - u32 wInterrupt; -}; - -struct hpi_resource { - union { - struct hpi_pci *Pci; - char *net_if; - } r; -#ifndef HPI64BIT /* keep structure size constant */ - u32 dwPadTo64; -#endif - u16 wBusType; /* HPI_BUS_PNPISA, _PCI, _USB etc */ - u16 wPadding; - -}; - -/** Format info used inside struct hpi_message - Not the same as public API struct hpi_format */ -struct hpi_msg_format { - u32 dwSampleRate; - /**< 11025, 32000, 44100 ... */ - u32 dwBitRate; /**< for MPEG */ - u32 dwAttributes; - /**< Stereo/JointStereo/Mono */ - u16 wChannels; /**< 1,2..., (or ancillary mode or idle bit */ - u16 wFormat; /**< HPI_FORMAT_PCM16, _MPEG etc. see \ref formats. */ -}; - -/** Buffer+format structure. - Must be kept 7 * 32 bits to match public struct hpi_datastruct */ -struct hpi_msg_data { - struct hpi_msg_format Format; - u8 *pbData; -#ifndef HPI64BIT - u32 dwPadding; -#endif - u32 dwDataSize; -}; - -/** struct hpi_datastructure used up to 3.04 driver */ -struct hpi_data_legacy32 { - struct hpi_format Format; - u32 pbData; - u32 dwDataSize; -}; - -#ifdef HPI64BIT -/* Compatibility version of struct hpi_data*/ -struct hpi_data_compat32 { - struct hpi_msg_format Format; - u32 pbData; - u32 dwPadding; - u32 dwDataSize; -}; -#endif - -struct hpi_buffer { - /** placehoder for backward compatability (see dwBufferSize) */ - struct hpi_msg_format reserved; - u32 dwCommand; /**< HPI_BUFFER_CMD_xxx*/ - u32 dwPciAddress; /**< PCI physical address of buffer for DSP DMA */ - u32 dwBufferSize; /**< must line up with dwDataSize of HPI_DATA*/ -}; - -struct hpi_subsys_msg { - struct hpi_resource Resource; -}; - -struct hpi_subsys_res { - u32 dwVersion; - u32 dwData; /* used to return extended version */ - u16 wNumAdapters; /* number of adapters */ - u16 wAdapterIndex; - u16 awAdapterList[HPI_MAX_ADAPTERS]; -}; - -struct hpi_adapter_msg { - u32 dwAdapterMode; /* adapter mode */ - u16 wAssertId; /* assert number for "test assert" call - wObjectIndex for find object call - wQueryOrSet for HPI_AdapterSetModeEx() */ - u16 wObjectType; /* for adapter find object call */ -}; - -union hpi_adapterx_msg { - struct hpi_adapter_msg adapter; - struct { - u32 dwOffset; - } query_flash; - struct { - u32 dwOffset; - u32 dwLength; - u32 dwKey; - } start_flash; - struct { - u32 dwChecksum; - u16 wSequence; - u16 wLength; - } program_flash; - struct { - u16 wProperty; - u16 wParameter1; - u16 wParameter2; - } property_set; - struct { - u16 wIndex; - u16 wWhat; - u16 wPropertyIndex; - } property_enum; - struct { - u16 index; - } module_info; - struct { - u32 dwDspAddress; - u32 dwCountBytes; - } debug_read; -}; - -struct hpi_adapter_res { - u32 dwSerialNumber; - u16 wAdapterType; - u16 wAdapterIndex; /* Is this needed? also used for wDspIndex */ - u16 wNumIStreams; - u16 wNumOStreams; - u16 wNumMixers; - u16 wVersion; - u8 szAdapterAssert[STR_SIZE(HPI_STRING_LEN)]; -}; - -union hpi_adapterx_res { - struct hpi_adapter_res adapter; - struct { - u32 dwChecksum; - u32 dwLength; - u32 dwVersion; - } query_flash; - struct { - u16 wSequence; - } program_flash; - struct { - u16 wParameter1; - u16 wParameter2; - } property_get; -}; - -struct hpi_stream_msg { - union { - struct hpi_msg_data Data; - struct hpi_data_legacy32 Data32; - u16 wVelocity; - struct hpi_punchinout Pio; - u32 dwTimeScale; - struct hpi_buffer Buffer; - struct hpi_streamid Stream; - } u; - u16 wStreamIndex; - u16 wIStreamIndex; -}; - -struct hpi_stream_res { - union { - struct { - /* size of hardware buffer */ - u32 dwBufferSize; - /* OutStream - data to play, - InStream - data recorded */ - u32 dwDataAvailable; - /* OutStream - samples played, - InStream - samples recorded */ - u32 dwSamplesTransferred; - /* Adapter - OutStream - data to play, - InStream - data recorded */ - u32 dwAuxiliaryDataAvailable; - u16 wState; /* HPI_STATE_PLAYING, _STATE_STOPPED */ - u16 wPadding; - } stream_info; - struct { - u32 dwBufferSize; - u32 dwDataAvailable; - u32 dwSamplesTransfered; - u16 wState; - u16 wOStreamIndex; - u16 wIStreamIndex; - u16 wPadding; - u32 dwAuxiliaryDataAvailable; - } legacy_stream_info; - struct { - /* bitmap of grouped OutStreams */ - u32 dwOutStreamGroupMap; - /* bitmap of grouped InStreams */ - u32 dwInStreamGroupMap; - } group_info; - } u; -}; - -struct hpi_mixer_msg { - u16 wControlIndex; - u16 wControlType; /* = HPI_CONTROL_METER _VOLUME etc */ - u16 wPadding1; /* Maintain alignment of subsequent fields */ - u16 wNodeType1; /* = HPI_SOURCENODE_LINEIN etc */ - u16 wNodeIndex1; /* = 0..N */ - u16 wNodeType2; - u16 wNodeIndex2; - u16 wPadding2; /* round to 4 bytes */ -}; - -struct hpi_mixer_res { - u16 wSrcNodeType; /* = HPI_SOURCENODE_LINEIN etc */ - u16 wSrcNodeIndex; /* = 0..N */ - u16 wDstNodeType; - u16 wDstNodeIndex; - /* Also controlType for MixerGetControlByIndex */ - u16 wControlIndex; - /* may indicate which DSP the control is located on */ - u16 wDspIndex; -}; - -union hpi_mixerx_msg { - struct { - u16 wStartingIndex; - u16 wFlags; - u32 dwLengthInBytes; /* length in bytes of pData */ - u32 pData; /* pointer to a data array */ - } gcabi; - struct { - u16 wCommand; - u16 wIndex; - } store; /* for HPI_MIXER_STORE message */ -}; - -union hpi_mixerx_res { - struct { - u32 dwBytesReturned; /* size of items returned */ - u32 pData; /* pointer to data array */ - u16 wMoreToDo; /* indicates if there is more to do */ - } gcabi; -}; - -struct hpi_control_msg { - u32 dwParam1; /* generic parameter 1 */ - u32 dwParam2; /* generic parameter 2 */ - short anLogValue[HPI_MAX_CHANNELS]; - u16 wAttribute; /* control attribute or property */ - u16 wControlIndex; -}; - -struct hpi_control_union_msg { - union { - struct { - u32 dwParam1; /* generic parameter 1 */ - u32 dwParam2; /* generic parameter 2 */ - short anLogValue[HPI_MAX_CHANNELS]; - } old; - union { - u32 dwFrequency; - u32 dwGain; - u32 dwBand; - u32 dwDeemphasis; - u32 dwProgram; - struct { - u32 dwMode; - u32 dwValue; - } mode; - } tuner; - } u; - u16 wAttribute; /* control attribute or property */ - u16 wControlIndex; -}; - -struct hpi_control_res { - /* Could make union. dwParam, anLogValue never used in same response */ - u32 dwParam1; - u32 dwParam2; - short anLogValue[HPI_MAX_CHANNELS]; -}; - -union hpi_control_union_res { - struct { - u32 dwParam1; - u32 dwParam2; - short anLogValue[HPI_MAX_CHANNELS]; - } old; - union { - u32 dwBand; - u32 dwFrequency; - u32 dwGain; - u32 dwLevel; - u32 dwDeemphasis; - struct { - u32 dwData[2]; - u32 dwBLER; - } rds; - } tuner; - struct { - char szData[8]; - u32 dwRemainingChars; - } chars8; -}; - -/* HPI_CONTROLX_STRUCTURES */ - -/* Message */ - -/** Used for all HMI variables where max length <= 8 bytes -*/ -struct hpi_controlx_msg_cobranet_data { - u32 dwHmiAddress; - u32 dwByteCount; - u32 dwData[2]; -}; - -/** Used for string data, and for packet bridge -*/ -struct hpi_controlx_msg_cobranet_bigdata { - u32 dwHmiAddress; - u32 dwByteCount; - u8 *pbData; -#ifndef HPI64BIT - u32 dwPadding; -#endif -}; - -/** Used for PADS control reading of string fields. -*/ -struct hpi_controlx_msg_pad_data { - u32 dwField; - u32 dwByteCount; - u8 *pbData; -#ifndef HPI64BIT - u32 dwPadding; -#endif -}; - -/** Used for generic data -*/ - -struct hpi_controlx_msg_generic { - u32 dwParam1; - u32 dwParam2; -}; - -struct hpi_controlx_msg { - union { - struct hpi_controlx_msg_cobranet_data cobranet_data; - struct hpi_controlx_msg_cobranet_bigdata cobranet_bigdata; - struct hpi_controlx_msg_generic generic; - struct hpi_controlx_msg_pad_data pad_data; - /* nothing extra to send for status read */ - } u; - u16 wControlIndex; - u16 wAttribute; /* control attribute or property */ -}; - -/* Response */ - -/** -*/ -struct hpi_controlx_res_cobranet_data { - u32 dwByteCount; - u32 dwData[2]; -}; - -struct hpi_controlx_res_cobranet_bigdata { - u32 dwByteCount; -}; - -struct hpi_controlx_res_cobranet_status { - u32 dwStatus; - u32 dwReadableSize; - u32 dwWriteableSize; -}; - -struct hpi_controlx_res_generic { - u32 dwParam1; - u32 dwParam2; -}; - -struct hpi_controlx_res { - union { - struct hpi_controlx_res_cobranet_bigdata cobranet_bigdata; - struct hpi_controlx_res_cobranet_data cobranet_data; - struct hpi_controlx_res_cobranet_status cobranet_status; - struct hpi_controlx_res_generic generic; - } u; -}; - -struct hpi_nvmemory_msg { - u16 wIndex; - u16 wData; -}; - -struct hpi_nvmemory_res { - u16 wSizeInBytes; - u16 wData; -}; - -struct hpi_gpio_msg { - u16 wBitIndex; - u16 wBitData; -}; - -struct hpi_gpio_res { - u16 wNumberInputBits; - u16 wNumberOutputBits; - u16 wBitData; - u16 wPadding; -}; - -struct hpi_async_msg { - u32 dwEvents; - u16 wMaximumEvents; - u16 wPadding; -}; - -struct hpi_async_res { - union { - struct { - u16 wCount; - } count; - struct { - u32 dwEvents; - u16 wNumberReturned; - u16 wPadding; - } get; - struct hpi_async_event event; - } u; -}; - -struct hpi_watchdog_msg { - u32 dwTimeMs; -}; - -struct hpi_watchdog_res { - u32 dwTimeMs; -}; - -struct hpi_clock_msg { - u16 wHours; - u16 wMinutes; - u16 wSeconds; - u16 wMilliSeconds; -}; - -struct hpi_clock_res { - u16 wSizeInBytes; - u16 wHours; - u16 wMinutes; - u16 wSeconds; - u16 wMilliSeconds; - u16 wPadding; -}; - -struct hpi_profile_msg { - u16 wIndex; - u16 wPadding; -}; - -struct hpi_profile_res_open { - u16 wMaxProfiles; -}; - -struct hpi_profile_res_time { - u32 dwMicroSeconds; - u32 dwCallCount; - u32 dwMaxMicroSeconds; - u32 dwMinMicroSeconds; - u16 wSeconds; -}; - -struct hpi_profile_res_name { -/* u8 messes up response size for 56301 DSP */ - u16 szName[16]; -}; - -struct hpi_profile_res { - union { - struct hpi_profile_res_open o; - struct hpi_profile_res_time t; - struct hpi_profile_res_name n; - } u; -}; - -struct hpi_message_header { - u16 wSize; - u16 wType; /* HPI_MSG_MESSAGE, HPI_MSG_RESPONSE */ - u16 wObject; /* HPI_OBJ_* */ - u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ - u16 wAdapterIndex; /* the adapter index */ - u16 wDspIndex; /* the dsp index on the adapter */ -}; - -struct hpi_message { - /* following fields must match HPI_MESSAGE_HEADER */ - u16 wSize; - u16 wType; /* HPI_TYPE_MESSAGE, HPI_TYPE_RESPONSE */ - u16 wObject; /* HPI_OBJ_* */ - u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ - u16 wAdapterIndex; /* the adapter index */ - u16 wDspIndex; /* the dsp index on the adapter */ - union { - struct hpi_subsys_msg s; - struct hpi_adapter_msg a; - union hpi_adapterx_msg ax; - struct hpi_stream_msg d; - struct hpi_mixer_msg m; - union hpi_mixerx_msg mx; /* extended mixer; */ - struct hpi_control_msg c; /* mixer control; */ - /* identical to struct hpi_control_msg, - but field naming is improved */ - struct hpi_control_union_msg cu; - struct hpi_controlx_msg cx; /* extended mixer control; */ - struct hpi_nvmemory_msg n; - struct hpi_gpio_msg l; /* digital i/o */ - struct hpi_watchdog_msg w; - struct hpi_clock_msg t; /* dsp time */ - struct hpi_profile_msg p; - struct hpi_async_msg as; - } u; -}; - -#define HPI_MESSAGE_SIZE_BY_OBJECT { \ - sizeof(struct hpi_message_header) , /* Default, no object type 0 */ \ - sizeof(struct hpi_message_header) + sizeof(struct hpi_subsys_msg),\ - sizeof(struct hpi_message_header) + sizeof(union hpi_adapterx_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_stream_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_stream_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_mixer_msg),\ - sizeof(struct hpi_message_header) , /* no node message */ \ - sizeof(struct hpi_message_header) + sizeof(struct hpi_control_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_nvmemory_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_gpio_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_watchdog_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_clock_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_profile_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_controlx_msg),\ - sizeof(struct hpi_message_header) + sizeof(struct hpi_async_msg) \ -} - -struct hpi_response_header { - u16 wSize; - u16 wType; /* HPI_MSG_MESSAGE, HPI_MSG_RESPONSE */ - u16 wObject; /* HPI_OBJ_* */ - u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ - u16 wError; /* HPI_ERROR_xxx */ - u16 wSpecificError; /* Adapter specific error */ -}; - -struct hpi_response { -/* following fields must match HPI_RESPONSE_HEADER */ - u16 wSize; - u16 wType; /* HPI_MSG_MESSAGE, HPI_MSG_RESPONSE */ - u16 wObject; /* HPI_OBJ_* */ - u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ - u16 wError; /* HPI_ERROR_xxx */ - u16 wSpecificError; /* Adapter specific error */ - union { - struct hpi_subsys_res s; - struct hpi_adapter_res a; - union hpi_adapterx_res ax; - struct hpi_stream_res d; - struct hpi_mixer_res m; - union hpi_mixerx_res mx; /* extended mixer; */ - struct hpi_control_res c; /* mixer control; */ - /* identical to hpi_control_res, but field naming is improved */ - union hpi_control_union_res cu; - struct hpi_controlx_res cx; /* extended mixer control; */ - struct hpi_nvmemory_res n; - struct hpi_gpio_res l; /* digital i/o */ - struct hpi_watchdog_res w; - struct hpi_clock_res t; /* dsp time */ - struct hpi_profile_res p; - struct hpi_async_res as; - u8 bytes[52]; - } u; -}; - -#define HPI_RESPONSE_SIZE_BY_OBJECT { \ - sizeof(struct hpi_response_header) ,/* Default, no object type 0 */ \ - sizeof(struct hpi_response_header) + sizeof(struct hpi_subsys_res),\ - sizeof(struct hpi_response_header) + sizeof(union hpi_adapterx_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_stream_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_stream_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_mixer_res),\ - sizeof(struct hpi_response_header) , /* no node response */ \ - sizeof(struct hpi_response_header) + sizeof(struct hpi_control_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_nvmemory_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_gpio_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_watchdog_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_clock_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_profile_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_controlx_res),\ - sizeof(struct hpi_response_header) + sizeof(struct hpi_async_res) \ -} - -/*////////////////////////////////////////////////////////////////////////// */ -/* declarations for compact control calls */ -struct hpi_control_defn { - u8 wType; - u8 wChannels; - u8 wSrcNodeType; - u8 wSrcNodeIndex; - u8 wDestNodeType; - u8 wDestNodeIndex; -}; - -/*////////////////////////////////////////////////////////////////////////// */ -/* declarations for control caching (internal to HPI<->DSP interaction) */ - -/** A compact representation of (part of) a controls state. -Used for efficient transfer of the control state -between DSP and host or across a network -*/ -struct hpi_control_cache_info { - /** one of HPI_CONTROL_* */ - u8 ControlType; - /** The total size of cached information in 32-bit words. */ - u8 nSizeIn32bitWords; - /** The original index of the control on the DSP */ - u16 ControlIndex; -}; - -struct hpi_control_cache_single { - struct hpi_control_cache_info i; - union { - struct { /* volume */ - u16 anLog[2]; - } v; - struct { /* peak meter */ - u16 anLogPeak[2]; - u16 anLogRMS[2]; - } p; - struct { /* channel mode */ - u16 wMode; - } m; - struct { /* multiplexer */ - u16 wSourceNodeType; - u16 wSourceNodeIndex; - } x; - struct { /* level/trim */ - u16 anLog[2]; - } l; - struct { /* tuner - partial caching. - Some attributes go to the DSP. */ - u32 dwFreqInkHz; - u16 wBand; - u16 wLevel; - } t; - struct { /* AESEBU Rx status */ - u32 dwErrorStatus; - u32 dwSource; - } aes3rx; - struct { /* AESEBU Tx */ - u32 dwFormat; - } aes3tx; - struct { /* tone detector */ - u16 wState; - } tone; - struct { /* silence detector */ - u32 dwState; - u32 dwCount; - } silence; - struct { /* sample clock */ - u16 wSource; - u16 wSourceIndex; - u32 dwSampleRate; - } clk; - struct { /* generic control */ - u32 dw1; - u32 dw2; - } g; - } u; -}; - -struct hpi_control_cache_pad { - struct hpi_control_cache_info i; - u32 dwFieldValidFlags; - u8 cChannel[8]; - u8 cArtist[40]; - u8 cTitle[40]; - u8 cComment[200]; - u32 dwPTY; - u32 dwPI; - u32 dwTrafficSupported; - u32 dwTrafficAnouncement; -}; - -/*/////////////////////////////////////////////////////////////////////////// */ -/* declarations for 2^N sized FIFO buffer (internal to HPI<->DSP interaction) */ -struct hpi_fifo_buffer { - u32 dwSize; - u32 dwDSPIndex; - u32 dwHostIndex; -}; - -#ifndef DISABLE_PRAGMA_PACK1 -#pragma pack(pop) -#endif - -/* skip host side function declarations for DSP - compile and documentation extraction */ -void HPI_InitMessage( - struct hpi_message *phm, - u16 wObject, - u16 wFunction -); - -void HPI_InitResponse( - struct hpi_response *phr, - u16 wObject, - u16 wFunction, - u16 wError -); - -char HPI_HandleObject( - const u32 dwHandle -); - -void HPI_HandleToIndexes( - const u32 dwHandle, - u16 *pwAdapterIndex, - u16 *pwObjectIndex, - u16 *pwDspIndex -); - -u32 HPI_IndexesToHandle( - const char cObject, - const u16 wAdapterIndex, - const u16 wObjectIndex, - const u16 wDspIndex -); - -/*////////////////////////////////////////////////////////////////////////// */ - -/* main HPI entry point */ -HPI_HandlerFunc HPI_Message; - -/* UDP message */ -void HPI_MessageUDP( - struct hpi_message *phm, - struct hpi_response *phr, - unsigned int nTimeout -); - -/* used in PnP OS/driver */ -u16 HPI_SubSysCreateAdapter( - struct hpi_hsubsys *phSubSys, - struct hpi_resource *pResource, - u16 *pwAdapterIndex -); - -u16 HPI_SubSysDeleteAdapter( - struct hpi_hsubsys *phSubSys, - u16 wAdapterIndex -); - -void HPI_FormatToMsg( - struct hpi_msg_format *pMF, - struct hpi_format *pF -); -void HPI_StreamResponseToLegacy( - struct hpi_stream_res *pSR -); - -/*////////////////////////////////////////////////////////////////////////// */ -/* declarations for individual HPI entry points */ -HPI_HandlerFunc HPI_6000; -HPI_HandlerFunc HPI_6205; -HPI_HandlerFunc HPI_COMMON; - #ifdef __cplusplus /* *INDENT-OFF* */ } diff --git a/pci/asihpi/hpi_internal.h b/pci/asihpi/hpi_internal.h new file mode 100644 index 0000000..9a1933f --- /dev/null +++ b/pci/asihpi/hpi_internal.h @@ -0,0 +1,1346 @@ +/****************************************************************************** + + AudioScience HPI driver + Copyright (C) 1997-2003 AudioScience Inc. <support(a)audioscience.com> + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +HPI internal definitions + +(C) Copyright AudioScience Inc. 1996-2008 +******************************************************************************/ + +#ifndef _HPI_INTERNAL_H_ +#define _HPI_INTERNAL_H_ + +#include "hpi.h" + +/** maximum number of memory regions mapped to an adapter */ +#define HPI_MAX_ADAPTER_MEM_SPACES (2) + +#include "hpios.h" + +#ifdef __cplusplus +/* *INDENT-OFF* */ +extern "C" { +/* *INDENT-ON* */ +#endif + +/******************************************* CONTROL ATTRIBUTES ****/ +/* (in order of control type ID */ + + /* This allows for 255 control types, 256 unique attributes each */ +#define HPI_CTL_ATTR(ctl, ai) (HPI_CONTROL_##ctl * 0x100 + ai) + +/* Get the sub-index of the attribute for a control type */ +#define HPI_CTL_ATTR_INDEX(i) (i&0xff) + +/* Original 0-based non-unique attributes, might become unique later */ +#define HPI_CTL_ATTR0(ctl, ai) (ai) + +/* Generic control attributes. If a control uses any of these attributes + its other attributes must also be defined using HPI_CTL_ATTR() +*/ + +/** Enable a control. +0=disable, 1=enable +\note generic to all mixer plugins? +*/ +#define HPI_GENERIC_ENABLE HPI_CTL_ATTR(GENERIC, 1) + +/** Enable event generation for a control. +0=disable, 1=enable +\note generic to all controls that can generate events +*/ +#define HPI_GENERIC_EVENT_ENABLE HPI_CTL_ATTR(GENERIC, 2) + +/* Volume Control attributes */ +#define HPI_VOLUME_GAIN HPI_CTL_ATTR0(VOLUME, 1) +#define HPI_VOLUME_AUTOFADE HPI_CTL_ATTR0(VOLUME, 2) + +/** For HPI_ControlQuery() to get the number of channels of a volume control*/ +#define HPI_VOLUME_NUM_CHANNELS HPI_CTL_ATTR0(VOLUME, 6) +#define HPI_VOLUME_RANGE HPI_CTL_ATTR0(VOLUME, 10) + +/** Level Control attributes */ +#define HPI_LEVEL_GAIN HPI_CTL_ATTR0(LEVEL, 1) +#define HPI_LEVEL_RANGE HPI_CTL_ATTR0(LEVEL, 10) + +/* Meter Control attributes */ +/** return RMS signal level */ +#define HPI_METER_RMS HPI_CTL_ATTR0(METER, 1) +/** return peak signal level */ +#define HPI_METER_PEAK HPI_CTL_ATTR0(METER, 2) +/** ballistics for ALL rms meters on adapter */ +#define HPI_METER_RMS_BALLISTICS HPI_CTL_ATTR0(METER, 3) +/** ballistics for ALL peak meters on adapter */ +#define HPI_METER_PEAK_BALLISTICS HPI_CTL_ATTR0(METER, 4) + +/** For HPI_ControlQuery() to get the number of channels of a meter control*/ +#define HPI_METER_NUM_CHANNELS HPI_CTL_ATTR0(METER, 5) + +/* Multiplexer control attributes */ +#define HPI_MULTIPLEXER_SOURCE HPI_CTL_ATTR0(MULTIPLEXER, 1) +#define HPI_MULTIPLEXER_QUERYSOURCE HPI_CTL_ATTR0(MULTIPLEXER, 2) + +/** AES/EBU transmitter control attributes */ +/** AESEBU or SPDIF */ +#define HPI_AESEBUTX_FORMAT HPI_CTL_ATTR0(AESEBUTX, 1) +#define HPI_AESEBUTX_SAMPLERATE HPI_CTL_ATTR0(AESEBUTX, 3) +#define HPI_AESEBUTX_CHANNELSTATUS HPI_CTL_ATTR0(AESEBUTX, 4) +#define HPI_AESEBUTX_USERDATA HPI_CTL_ATTR0(AESEBUTX, 5) + +/** AES/EBU receiver control attributes */ +#define HPI_AESEBURX_FORMAT HPI_CTL_ATTR0(AESEBURX, 1) +#define HPI_AESEBURX_ERRORSTATUS HPI_CTL_ATTR0(AESEBURX, 2) +#define HPI_AESEBURX_SAMPLERATE HPI_CTL_ATTR0(AESEBURX, 3) +#define HPI_AESEBURX_CHANNELSTATUS HPI_CTL_ATTR0(AESEBURX, 4) +#define HPI_AESEBURX_USERDATA HPI_CTL_ATTR0(AESEBURX, 5) + +/** \defgroup tuner_defs Tuners +\{ +*/ +/** \defgroup tuner_attrs Tuner control attributes +\{ +*/ +#define HPI_TUNER_BAND HPI_CTL_ATTR0(TUNER, 1) +#define HPI_TUNER_FREQ HPI_CTL_ATTR0(TUNER, 2) +#define HPI_TUNER_LEVEL HPI_CTL_ATTR0(TUNER, 3) +#define HPI_TUNER_AUDIOMUTE HPI_CTL_ATTR0(TUNER, 4) +/* use TUNER_STATUS instead */ +#define HPI_TUNER_VIDEO_STATUS HPI_CTL_ATTR0(TUNER, 5) +#define HPI_TUNER_GAIN HPI_CTL_ATTR0(TUNER, 6) +#define HPI_TUNER_STATUS HPI_CTL_ATTR0(TUNER, 7) +#define HPI_TUNER_MODE HPI_CTL_ATTR0(TUNER, 8) +/** RDS data. */ +#define HPI_TUNER_RDS HPI_CTL_ATTR0(TUNER, 9) +/** Audio pre-emphasis. */ +#define HPI_TUNER_DEEMPHASIS HPI_CTL_ATTR(TUNER, 10) +/** HD-Radio tuner program control. */ +#define HPI_TUNER_PROGRAM HPI_CTL_ATTR(TUNER, 11) +/** HD-Radio tuner digital signal quality. */ +#define HPI_TUNER_HDRADIO_SIGNAL_QUALITY HPI_CTL_ATTR(TUNER, 12) +/** HD-Radio SDK firmware version. */ +#define HPI_TUNER_HDRADIO_SDK_VERSION HPI_CTL_ATTR(TUNER, 13) +/** HD-Radio DSP firmware version. */ +#define HPI_TUNER_HDRADIO_DSP_VERSION HPI_CTL_ATTR(TUNER, 14) + +/** \} */ + +/** \defgroup pads_attrs Tuner PADs control attributes +\{ +*/ +/** The text string containing the station/channel combination. */ +#define HPI_PAD_CHANNEL_NAME HPI_CTL_ATTR(PAD, 1) +/** The text string containing the artist. */ +#define HPI_PAD_ARTIST HPI_CTL_ATTR(PAD, 2) +/** The text string containing the title. */ +#define HPI_PAD_TITLE HPI_CTL_ATTR(PAD, 3) +/** The text string containing the comment. */ +#define HPI_PAD_COMMENT HPI_CTL_ATTR(PAD, 4) +/** The integer containing the PTY code. */ +#define HPI_PAD_PROGRAM_TYPE HPI_CTL_ATTR(PAD, 5) +/** The integer containing the program identification. */ +#define HPI_PAD_PROGRAM_ID HPI_CTL_ATTR(PAD, 6) +/** The integer containing whether traffic information is supported. +Contains either 1 or 0. */ +#define HPI_PAD_TA_SUPPORT HPI_CTL_ATTR(PAD, 7) +/** The integer containing whether traffic announcement is in progress. +Contains either 1 or 0. */ +#define HPI_PAD_TA_ACTIVE HPI_CTL_ATTR(PAD, 8) +/** \} */ + +/* VOX control attributes */ +#define HPI_VOX_THRESHOLD HPI_CTL_ATTR0(VOX, 1) + +/*?? channel mode used hpi_multiplexer_source attribute == 1 */ +#define HPI_CHANNEL_MODE_MODE HPI_CTL_ATTR0(CHANNEL_MODE, 1) + +/** \defgroup channel_modes Channel Modes +Used for HPI_ChannelModeSet/Get() +\{ +*/ +/** Left channel out = left channel in, Right channel out = right channel in. */ +#define HPI_CHANNEL_MODE_NORMAL 1 +/** Left channel out = right channel in, Right channel out = left channel in. */ +#define HPI_CHANNEL_MODE_SWAP 2 +/** Left channel out = left channel in, Right channel out = left channel in. */ +#define HPI_CHANNEL_MODE_LEFT_TO_STEREO 3 +/** Left channel out = right channel in, Right channel out = right channel in.*/ +#define HPI_CHANNEL_MODE_RIGHT_TO_STEREO 4 +/** Left channel out = (left channel in + right channel in)/2, + Right channel out = mute. */ +#define HPI_CHANNEL_MODE_STEREO_TO_LEFT 5 +/** Left channel out = mute, + Right channel out = (right channel in + left channel in)/2. */ +#define HPI_CHANNEL_MODE_STEREO_TO_RIGHT 6 +#define HPI_CHANNEL_MODE_LAST 6 +/** \} */ + +/* Bitstream control set attributes */ +#define HPI_BITSTREAM_DATA_POLARITY HPI_CTL_ATTR0(BITSTREAM, 1) +#define HPI_BITSTREAM_CLOCK_EDGE HPI_CTL_ATTR0(BITSTREAM, 2) +#define HPI_BITSTREAM_CLOCK_SOURCE HPI_CTL_ATTR0(BITSTREAM, 3) + +#define HPI_POLARITY_POSITIVE 0 +#define HPI_POLARITY_NEGATIVE 1 + +/* Bitstream control get attributes */ +#define HPI_BITSTREAM_ACTIVITY 1 + +/* SampleClock control attributes */ +#define HPI_SAMPLECLOCK_SOURCE HPI_CTL_ATTR0(SAMPLECLOCK, 1) +#define HPI_SAMPLECLOCK_SAMPLERATE HPI_CTL_ATTR0(SAMPLECLOCK, 2) +#define HPI_SAMPLECLOCK_SOURCE_INDEX HPI_CTL_ATTR0(SAMPLECLOCK, 3) +#define HPI_SAMPLECLOCK_LOCAL_SAMPLERATE HPI_CTL_ATTR0(SAMPLECLOCK, 4) +#define HPI_SAMPLECLOCK_AUTO HPI_CTL_ATTR0(SAMPLECLOCK, 5) + +/* Microphone control attributes */ +#define HPI_MICROPHONE_PHANTOM_POWER HPI_CTL_ATTR0(MICROPHONE, 1) + +/** Equalizer control attributes +*/ +/** Used to get number of filters in an EQ. (Can't set) */ +#define HPI_EQUALIZER_NUM_FILTERS HPI_CTL_ATTR0(EQUALIZER, 1) +/** Set/get the filter by type, freq, Q, gain */ +#define HPI_EQUALIZER_FILTER HPI_CTL_ATTR0(EQUALIZER, 2) +/** Get the biquad coefficients */ +#define HPI_EQUALIZER_COEFFICIENTS HPI_CTL_ATTR0(EQUALIZER, 3) + +#define HPI_COMPANDER_PARAMS HPI_CTL_ATTR(COMPANDER, 1) + +/* Cobranet control attributes. + MUST be distinct from all other control attributes. + This is so that host side processing can easily identify a Cobranet control + and apply additional host side operations (like copying data) as required. +*/ +#define HPI_COBRANET_SET HPI_CTL_ATTR(COBRANET, 1) +#define HPI_COBRANET_GET HPI_CTL_ATTR(COBRANET, 2) +#define HPI_COBRANET_SET_DATA HPI_CTL_ATTR(COBRANET, 3) +#define HPI_COBRANET_GET_DATA HPI_CTL_ATTR(COBRANET, 4) +#define HPI_COBRANET_GET_STATUS HPI_CTL_ATTR(COBRANET, 5) +#define HPI_COBRANET_SEND_PACKET HPI_CTL_ATTR(COBRANET, 6) +#define HPI_COBRANET_GET_PACKET HPI_CTL_ATTR(COBRANET, 7) + +/*------------------------------------------------------------ + Cobranet Chip Bridge - copied from HMI.H +------------------------------------------------------------*/ +#define HPI_COBRANET_HMI_cobraBridge 0x20000 +#define HPI_COBRANET_HMI_cobraBridgeTxPktBuf \ + (HPI_COBRANET_HMI_cobraBridge + 0x1000) +#define HPI_COBRANET_HMI_cobraBridgeRxPktBuf \ + (HPI_COBRANET_HMI_cobraBridge + 0x2000) +#define HPI_COBRANET_HMI_cobraIfTable1 0x110000 +#define HPI_COBRANET_HMI_cobraIfPhyAddress \ + (HPI_COBRANET_HMI_cobraIfTable1 + 0xd) +#define HPI_COBRANET_HMI_cobraProtocolIP 0x72000 +#define HPI_COBRANET_HMI_cobraIpMonCurrentIP \ + (HPI_COBRANET_HMI_cobraProtocolIP + 0x0) +#define HPI_COBRANET_HMI_cobraIpMonStaticIP \ + (HPI_COBRANET_HMI_cobraProtocolIP + 0x2) +#define HPI_COBRANET_HMI_cobraSys 0x100000 +#define HPI_COBRANET_HMI_cobraSysDesc \ + (HPI_COBRANET_HMI_cobraSys + 0x0) +#define HPI_COBRANET_HMI_cobraSysObjectID \ + (HPI_COBRANET_HMI_cobraSys + 0x100) +#define HPI_COBRANET_HMI_cobraSysContact \ + (HPI_COBRANET_HMI_cobraSys + 0x200) +#define HPI_COBRANET_HMI_cobraSysName \ + (HPI_COBRANET_HMI_cobraSys + 0x300) +#define HPI_COBRANET_HMI_cobraSysLocation \ + (HPI_COBRANET_HMI_cobraSys + 0x400) + +/*------------------------------------------------------------ + Cobranet Chip Status bits +------------------------------------------------------------*/ +#define HPI_COBRANET_HMI_STATUS_RXPACKET 2 +#define HPI_COBRANET_HMI_STATUS_TXPACKET 3 + +/*------------------------------------------------------------ + Ethernet header size +------------------------------------------------------------*/ +#define HPI_ETHERNET_HEADER_SIZE (16) + +/* These defines are used to fill in protocol information for an Ethernet packet + sent using HMI on CS18102 */ +/** ID supplied by Cirrius for ASI packets. */ +#define HPI_ETHERNET_PACKET_ID 0x85 +/** Simple packet - no special routing required */ +#define HPI_ETHERNET_PACKET_V1 0x01 +/** This packet must make its way to the host across the HPI interface */ +#define HPI_ETHERNET_PACKET_HOSTED_VIA_HMI 0x20 +/** This packet must make its way to the host across the HPI interface */ +#define HPI_ETHERNET_PACKET_HOSTED_VIA_HMI_V1 0x21 +/** This packet must make its way to the host across the HPI interface */ +#define HPI_ETHERNET_PACKET_HOSTED_VIA_HPI 0x40 +/** This packet must make its way to the host across the HPI interface */ +#define HPI_ETHERNET_PACKET_HOSTED_VIA_HPI_V1 0x41 + +#define HPI_ETHERNET_UDP_PORT (44600) /*!< UDP messaging port */ + +/** Base network time out is set to 100 milli-seconds. */ +#define HPI_ETHERNET_TIMEOUT_MS (100) + +/** \defgroup tonedet_attr Tonedetector attributes +\{ +Used by HPI_ToneDetector_Set() and HPI_ToneDetector_Get() +*/ + +/** Set the threshold level of a tonedetector, +Threshold is a -ve number in units of dB/100, +*/ +#define HPI_TONEDETECTOR_THRESHOLD HPI_CTL_ATTR(TONEDETECTOR, 1) + +/** Get the current state of tonedetection +The result is a bitmap of detected tones. pairs of bits represent the left +and right channels, with left channel in LSB. +The lowest frequency detector state is in the LSB +*/ +#define HPI_TONEDETECTOR_STATE HPI_CTL_ATTR(TONEDETECTOR, 2) + +/** Get the frequency of a tonedetector band. +*/ +#define HPI_TONEDETECTOR_FREQUENCY HPI_CTL_ATTR(TONEDETECTOR, 3) + +/**\}*/ + +/** \defgroup silencedet_attr SilenceDetector attributes +\{ +*/ + +/** Get the current state of tonedetection +The result is a bitmap with 1s for silent channels. Left channel is in LSB +*/ +#define HPI_SILENCEDETECTOR_STATE \ + HPI_CTL_ATTR(SILENCEDETECTOR, 2) + +/** Set the threshold level of a SilenceDetector, +Threshold is a -ve number in units of dB/100, +*/ +#define HPI_SILENCEDETECTOR_THRESHOLD \ + HPI_CTL_ATTR(SILENCEDETECTOR, 1) + +/** get/set the silence time before the detector triggers +*/ +#define HPI_SILENCEDETECTOR_DELAY \ + HPI_CTL_ATTR(SILENCEDETECTOR, 3) + +/**\}*/ + +/* Locked memory buffer alloc/free phases */ +/** use one message to allocate or free physical memory */ +#define HPI_BUFFER_CMD_EXTERNAL 0 +/** alloc physical memory */ +#define HPI_BUFFER_CMD_INTERNAL_ALLOC 1 +/** send physical memory address to adapter */ +#define HPI_BUFFER_CMD_INTERNAL_GRANTADAPTER 2 +/** notify adapter to stop using physical buffer */ +#define HPI_BUFFER_CMD_INTERNAL_REVOKEADAPTER 3 +/** free physical buffer */ +#define HPI_BUFFER_CMD_INTERNAL_FREE 4 + +/******************************************* CONTROLX ATTRIBUTES ****/ +/* NOTE: All controlx attributes must be unique, unlike control attributes */ + +/*****************************************************************************/ +/*****************************************************************************/ +/******** HPI LOW LEVEL MESSAGES *******/ +/*****************************************************************************/ +/*****************************************************************************/ +/** Pnp ids */ +/** "ASI" - actual is "ASX" - need to change */ +#define HPI_ID_ISAPNP_AUDIOSCIENCE 0x0669 +/** PCI vendor ID that AudioScience uses */ +#define HPI_PCI_VENDOR_ID_AUDIOSCIENCE 0x175C +/** PCI vendor ID that the DSP56301 has */ +#define HPI_PCI_VENDOR_ID_MOTOROLA 0x1057 +/** PCI vendor ID that TI uses */ +#define HPI_PCI_VENDOR_ID_TI 0x104C + +#define HPI_USB_VENDOR_ID_AUDIOSCIENCE 0x1257 +#define HPI_USB_W2K_TAG 0x57495341 /* "ASIW" */ +#define HPI_USB_LINUX_TAG 0x4C495341 /* "ASIL" */ + +/******************************************* message types */ +#define HPI_TYPE_MESSAGE 1 +#define HPI_TYPE_RESPONSE 2 +#define HPI_TYPE_DATA 3 +#define HPI_TYPE_SSX2BYPASS_MESSAGE 4 + +/******************************************* object types */ +#define HPI_OBJ_SUBSYSTEM 1 +#define HPI_OBJ_ADAPTER 2 +#define HPI_OBJ_OSTREAM 3 +#define HPI_OBJ_ISTREAM 4 +#define HPI_OBJ_MIXER 5 +#define HPI_OBJ_NODE 6 +#define HPI_OBJ_CONTROL 7 +#define HPI_OBJ_NVMEMORY 8 +#define HPI_OBJ_GPIO 9 +#define HPI_OBJ_WATCHDOG 10 +#define HPI_OBJ_CLOCK 11 +#define HPI_OBJ_PROFILE 12 +#define HPI_OBJ_CONTROLEX 13 +#define HPI_OBJ_ASYNCEVENT 14 + +#define HPI_OBJ_MAXINDEX 14 + +/******************************************* methods/functions */ + +#define HPI_OBJ_FUNCTION_SPACING 0x100 +#define HPI_MAKE_INDEX(obj, index) (obj * HPI_OBJ_FUNCTION_SPACING + index) +#define HPI_EXTRACT_INDEX(fn) (fn & 0xff) + +/* SUB-SYSTEM */ +#define HPI_SUBSYS_OPEN HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 1) +#define HPI_SUBSYS_GET_VERSION HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 2) +#define HPI_SUBSYS_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 3) +#define HPI_SUBSYS_FIND_ADAPTERS HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 4) +#define HPI_SUBSYS_CREATE_ADAPTER HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 5) +#define HPI_SUBSYS_CLOSE HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 6) +#define HPI_SUBSYS_DELETE_ADAPTER HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 7) +#define HPI_SUBSYS_DRIVER_LOAD HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 8) +#define HPI_SUBSYS_DRIVER_UNLOAD HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 9) + /*SGT*/ +#define HPI_SUBSYS_READ_PORT_8 HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 10) +#define HPI_SUBSYS_WRITE_PORT_8 HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 11) +#define HPI_SUBSYS_GET_NUM_ADAPTERS HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 12) +#define HPI_SUBSYS_GET_ADAPTER HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 13) +#define HPI_SUBSYS_SET_NETWORK_INTERFACE HPI_MAKE_INDEX(HPI_OBJ_SUBSYSTEM, 14) +/* ADAPTER */ +#define HPI_ADAPTER_OPEN HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 1) +#define HPI_ADAPTER_CLOSE HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 2) +#define HPI_ADAPTER_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 3) +#define HPI_ADAPTER_GET_ASSERT HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 4) +#define HPI_ADAPTER_TEST_ASSERT HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 5) +#define HPI_ADAPTER_SET_MODE HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 6) +#define HPI_ADAPTER_GET_MODE HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 7) +#define HPI_ADAPTER_ENABLE_CAPABILITY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 8) +#define HPI_ADAPTER_SELFTEST HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 9) +#define HPI_ADAPTER_FIND_OBJECT HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 10) +#define HPI_ADAPTER_QUERY_FLASH HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 11) +#define HPI_ADAPTER_START_FLASH HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 12) +#define HPI_ADAPTER_PROGRAM_FLASH HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 13) +#define HPI_ADAPTER_SET_PROPERTY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 14) +#define HPI_ADAPTER_GET_PROPERTY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 15) +#define HPI_ADAPTER_ENUM_PROPERTY HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 16) +#define HPI_ADAPTER_MODULE_INFO HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 17) +#define HPI_ADAPTER_DEBUG_READ HPI_MAKE_INDEX(HPI_OBJ_ADAPTER, 18) +#define HPI_ADAPTER_FUNCTION_COUNT 18 +/* OUTPUT STREAM */ +#define HPI_OSTREAM_OPEN HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 1) +#define HPI_OSTREAM_CLOSE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 2) +#define HPI_OSTREAM_WRITE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 3) +#define HPI_OSTREAM_START HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 4) +#define HPI_OSTREAM_STOP HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 5) +#define HPI_OSTREAM_RESET HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 6) +#define HPI_OSTREAM_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 7) +#define HPI_OSTREAM_QUERY_FORMAT HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 8) +#define HPI_OSTREAM_DATA HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 9) +#define HPI_OSTREAM_SET_VELOCITY HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 10) +#define HPI_OSTREAM_SET_PUNCHINOUT HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 11) +#define HPI_OSTREAM_SINEGEN HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 12) +#define HPI_OSTREAM_ANC_RESET HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 13) +#define HPI_OSTREAM_ANC_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 14) +#define HPI_OSTREAM_ANC_READ HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 15) +#define HPI_OSTREAM_SET_TIMESCALE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 16) +#define HPI_OSTREAM_SET_FORMAT HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 17) +#define HPI_OSTREAM_HOSTBUFFER_ALLOC HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 18) +#define HPI_OSTREAM_HOSTBUFFER_FREE HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 19) +#define HPI_OSTREAM_GROUP_ADD HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 20) +#define HPI_OSTREAM_GROUP_GETMAP HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 21) +#define HPI_OSTREAM_GROUP_RESET HPI_MAKE_INDEX(HPI_OBJ_OSTREAM, 22) +#define HPI_OSTREAM_FUNCTION_COUNT (22) +/* INPUT STREAM */ +#define HPI_ISTREAM_OPEN HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 1) +#define HPI_ISTREAM_CLOSE HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 2) +#define HPI_ISTREAM_SET_FORMAT HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 3) +#define HPI_ISTREAM_READ HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 4) +#define HPI_ISTREAM_START HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 5) +#define HPI_ISTREAM_STOP HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 6) +#define HPI_ISTREAM_RESET HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 7) +#define HPI_ISTREAM_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 8) +#define HPI_ISTREAM_QUERY_FORMAT HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 9) +#define HPI_ISTREAM_ANC_RESET HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 10) +#define HPI_ISTREAM_ANC_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 11) +#define HPI_ISTREAM_ANC_WRITE HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 12) +#define HPI_ISTREAM_HOSTBUFFER_ALLOC HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 13) +#define HPI_ISTREAM_HOSTBUFFER_FREE HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 14) +#define HPI_ISTREAM_GROUP_ADD HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 15) +#define HPI_ISTREAM_GROUP_GETMAP HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 16) +#define HPI_ISTREAM_GROUP_RESET HPI_MAKE_INDEX(HPI_OBJ_ISTREAM, 17) +#define HPI_ISTREAM_FUNCTION_COUNT (17) +/* MIXER */ +/* NOTE: + GET_NODE_INFO, SET_CONNECTION, GET_CONNECTIONS are not currently used */ +#define HPI_MIXER_OPEN HPI_MAKE_INDEX(HPI_OBJ_MIXER, 1) +#define HPI_MIXER_CLOSE HPI_MAKE_INDEX(HPI_OBJ_MIXER, 2) +#define HPI_MIXER_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_MIXER, 3) +#define HPI_MIXER_GET_NODE_INFO HPI_MAKE_INDEX(HPI_OBJ_MIXER, 4) +#define HPI_MIXER_GET_CONTROL HPI_MAKE_INDEX(HPI_OBJ_MIXER, 5) +#define HPI_MIXER_SET_CONNECTION HPI_MAKE_INDEX(HPI_OBJ_MIXER, 6) +#define HPI_MIXER_GET_CONNECTIONS HPI_MAKE_INDEX(HPI_OBJ_MIXER, 7) +#define HPI_MIXER_GET_CONTROL_BY_INDEX HPI_MAKE_INDEX(HPI_OBJ_MIXER, 8) +#define HPI_MIXER_GET_CONTROL_ARRAY_BY_INDEX HPI_MAKE_INDEX(HPI_OBJ_MIXER, 9) +#define HPI_MIXER_GET_CONTROL_MULTIPLE_VALUES HPI_MAKE_INDEX(HPI_OBJ_MIXER, 10) +#define HPI_MIXER_STORE HPI_MAKE_INDEX(HPI_OBJ_MIXER, 11) +#define HPI_MIXER_FUNCTION_COUNT 11 +/* MIXER CONTROLS */ +#define HPI_CONTROL_GET_INFO HPI_MAKE_INDEX(HPI_OBJ_CONTROL, 1) +#define HPI_CONTROL_GET_STATE HPI_MAKE_INDEX(HPI_OBJ_CONTROL, 2) +#define HPI_CONTROL_SET_STATE HPI_MAKE_INDEX(HPI_OBJ_CONTROL, 3) +#define HPI_CONTROL_FUNCTION_COUNT 3 +/* NONVOL MEMORY */ +#define HPI_NVMEMORY_OPEN HPI_MAKE_INDEX(HPI_OBJ_NVMEMORY, 1) +#define HPI_NVMEMORY_READ_BYTE HPI_MAKE_INDEX(HPI_OBJ_NVMEMORY, 2) +#define HPI_NVMEMORY_WRITE_BYTE HPI_MAKE_INDEX(HPI_OBJ_NVMEMORY, 3) +#define HPI_NVMEMORY_FUNCTION_COUNT 3 +/* GPIO */ +#define HPI_GPIO_OPEN HPI_MAKE_INDEX(HPI_OBJ_GPIO, 1) +#define HPI_GPIO_READ_BIT HPI_MAKE_INDEX(HPI_OBJ_GPIO, 2) +#define HPI_GPIO_WRITE_BIT HPI_MAKE_INDEX(HPI_OBJ_GPIO, 3) +#define HPI_GPIO_READ_ALL HPI_MAKE_INDEX(HPI_OBJ_GPIO, 4) +#define HPI_GPIO_WRITE_STATUS HPI_MAKE_INDEX(HPI_OBJ_GPIO, 5) +#define HPI_GPIO_FUNCTION_COUNT 5 +/* ASYNC EVENT */ +#define HPI_ASYNCEVENT_OPEN HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 1) +#define HPI_ASYNCEVENT_CLOSE HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 2) +#define HPI_ASYNCEVENT_WAIT HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 3) +#define HPI_ASYNCEVENT_GETCOUNT HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 4) +#define HPI_ASYNCEVENT_GET HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 5) +#define HPI_ASYNCEVENT_SENDEVENTS HPI_MAKE_INDEX(HPI_OBJ_ASYNCEVENT, 6) +#define HPI_ASYNCEVENT_FUNCTION_COUNT 6 +/* WATCH-DOG */ +#define HPI_WATCHDOG_OPEN HPI_MAKE_INDEX(HPI_OBJ_WATCHDOG, 1) +#define HPI_WATCHDOG_SET_TIME HPI_MAKE_INDEX(HPI_OBJ_WATCHDOG, 2) +#define HPI_WATCHDOG_PING HPI_MAKE_INDEX(HPI_OBJ_WATCHDOG, 3) +/* CLOCK */ +#define HPI_CLOCK_OPEN HPI_MAKE_INDEX(HPI_OBJ_CLOCK, 1) +#define HPI_CLOCK_SET_TIME HPI_MAKE_INDEX(HPI_OBJ_CLOCK, 2) +#define HPI_CLOCK_GET_TIME HPI_MAKE_INDEX(HPI_OBJ_CLOCK, 3) +/* PROFILE */ +#define HPI_PROFILE_OPEN_ALL HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 1) +#define HPI_PROFILE_START_ALL HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 2) +#define HPI_PROFILE_STOP_ALL HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 3) +#define HPI_PROFILE_GET HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 4) +#define HPI_PROFILE_GET_IDLECOUNT HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 5) +#define HPI_PROFILE_GET_NAME HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 6) +#define HPI_PROFILE_GET_UTILIZATION HPI_MAKE_INDEX(HPI_OBJ_PROFILE, 7) +#define HPI_PROFILE_FUNCTION_COUNT 7 +/* ////////////////////////////////////////////////////////////////////// */ +/* PRIVATE ATTRIBUTES */ +/* ////////////////////////////////////////////////////////////////////// */ +/* STRUCTURES */ +#ifndef DISABLE_PRAGMA_PACK1 +#pragma pack(push, 1) +#endif +/** PCI bus resource */ + struct hpi_pci { + u32 __iomem *apMemBase[HPI_MAX_ADAPTER_MEM_SPACES]; + struct pci_dev *pOsData; + +#ifndef HPI64BIT /* keep structure size constant */ + u32 dwPadding[HPI_MAX_ADAPTER_MEM_SPACES + 1]; +#endif + u16 wVendorId; + u16 wDeviceId; + u16 wSubSysVendorId; + u16 wSubSysDeviceId; + u16 wBusNumber; + u16 wDeviceNumber; + u32 wInterrupt; +}; + +struct hpi_resource { + union { + struct hpi_pci *Pci; + char *net_if; + } r; +#ifndef HPI64BIT /* keep structure size constant */ + u32 dwPadTo64; +#endif + u16 wBusType; /* HPI_BUS_PNPISA, _PCI, _USB etc */ + u16 wPadding; + +}; + +/** Format info used inside struct hpi_message + Not the same as public API struct hpi_format */ +struct hpi_msg_format { + u32 dwSampleRate; + /**< 11025, 32000, 44100 ... */ + u32 dwBitRate; /**< for MPEG */ + u32 dwAttributes; + /**< Stereo/JointStereo/Mono */ + u16 wChannels; /**< 1,2..., (or ancillary mode or idle bit */ + u16 wFormat; /**< HPI_FORMAT_PCM16, _MPEG etc. see \ref formats. */ +}; + +/** Buffer+format structure. + Must be kept 7 * 32 bits to match public struct hpi_datastruct */ +struct hpi_msg_data { + struct hpi_msg_format Format; + u8 *pbData; +#ifndef HPI64BIT + u32 dwPadding; +#endif + u32 dwDataSize; +}; + +/** struct hpi_datastructure used up to 3.04 driver */ +struct hpi_data_legacy32 { + struct hpi_format Format; + u32 pbData; + u32 dwDataSize; +}; + +#ifdef HPI64BIT +/* Compatibility version of struct hpi_data*/ +struct hpi_data_compat32 { + struct hpi_msg_format Format; + u32 pbData; + u32 dwPadding; + u32 dwDataSize; +}; +#endif + +struct hpi_buffer { + /** placehoder for backward compatability (see dwBufferSize) */ + struct hpi_msg_format reserved; + u32 dwCommand; /**< HPI_BUFFER_CMD_xxx*/ + u32 dwPciAddress; /**< PCI physical address of buffer for DSP DMA */ + u32 dwBufferSize; /**< must line up with dwDataSize of HPI_DATA*/ +}; + +struct hpi_streamid { + u16 wObjectType; + /**< Type of object, HPI_OBJ_OSTREAM or HPI_OBJ_ISTREAM. */ + u16 wStreamIndex; /**< OStream or IStream index. */ +}; + +struct hpi_punchinout { + u32 dwPunchInSample; + u32 dwPunchOutSample; +}; + +struct hpi_subsys_msg { + struct hpi_resource Resource; +}; + +struct hpi_subsys_res { + u32 dwVersion; + u32 dwData; /* used to return extended version */ + u16 wNumAdapters; /* number of adapters */ + u16 wAdapterIndex; + u16 awAdapterList[HPI_MAX_ADAPTERS]; +}; + +struct hpi_adapter_msg { + u32 dwAdapterMode; /* adapter mode */ + u16 wAssertId; /* assert number for "test assert" call + wObjectIndex for find object call + wQueryOrSet for HPI_AdapterSetModeEx() */ + u16 wObjectType; /* for adapter find object call */ +}; + +union hpi_adapterx_msg { + struct hpi_adapter_msg adapter; + struct { + u32 dwOffset; + } query_flash; + struct { + u32 dwOffset; + u32 dwLength; + u32 dwKey; + } start_flash; + struct { + u32 dwChecksum; + u16 wSequence; + u16 wLength; + } program_flash; + struct { + u16 wProperty; + u16 wParameter1; + u16 wParameter2; + } property_set; + struct { + u16 wIndex; + u16 wWhat; + u16 wPropertyIndex; + } property_enum; + struct { + u16 index; + } module_info; + struct { + u32 dwDspAddress; + u32 dwCountBytes; + } debug_read; +}; + +struct hpi_adapter_res { + u32 dwSerialNumber; + u16 wAdapterType; + u16 wAdapterIndex; /* Is this needed? also used for wDspIndex */ + u16 wNumIStreams; + u16 wNumOStreams; + u16 wNumMixers; + u16 wVersion; + u8 szAdapterAssert[HPI_STRING_LEN]; +}; + +union hpi_adapterx_res { + struct hpi_adapter_res adapter; + struct { + u32 dwChecksum; + u32 dwLength; + u32 dwVersion; + } query_flash; + struct { + u16 wSequence; + } program_flash; + struct { + u16 wParameter1; + u16 wParameter2; + } property_get; +}; + +struct hpi_stream_msg { + union { + struct hpi_msg_data Data; + struct hpi_data_legacy32 Data32; + u16 wVelocity; + struct hpi_punchinout Pio; + u32 dwTimeScale; + struct hpi_buffer Buffer; + struct hpi_streamid Stream; + } u; + u16 wStreamIndex; + u16 wIStreamIndex; +}; + +struct hpi_stream_res { + union { + struct { + /* size of hardware buffer */ + u32 dwBufferSize; + /* OutStream - data to play, + InStream - data recorded */ + u32 dwDataAvailable; + /* OutStream - samples played, + InStream - samples recorded */ + u32 dwSamplesTransferred; + /* Adapter - OutStream - data to play, + InStream - data recorded */ + u32 dwAuxiliaryDataAvailable; + u16 wState; /* HPI_STATE_PLAYING, _STATE_STOPPED */ + u16 wPadding; + } stream_info; + struct { + u32 dwBufferSize; + u32 dwDataAvailable; + u32 dwSamplesTransfered; + u16 wState; + u16 wOStreamIndex; + u16 wIStreamIndex; + u16 wPadding; + u32 dwAuxiliaryDataAvailable; + } legacy_stream_info; + struct { + /* bitmap of grouped OutStreams */ + u32 dwOutStreamGroupMap; + /* bitmap of grouped InStreams */ + u32 dwInStreamGroupMap; + } group_info; + } u; +}; + +struct hpi_mixer_msg { + u16 wControlIndex; + u16 wControlType; /* = HPI_CONTROL_METER _VOLUME etc */ + u16 wPadding1; /* Maintain alignment of subsequent fields */ + u16 wNodeType1; /* = HPI_SOURCENODE_LINEIN etc */ + u16 wNodeIndex1; /* = 0..N */ + u16 wNodeType2; + u16 wNodeIndex2; + u16 wPadding2; /* round to 4 bytes */ +}; + +struct hpi_mixer_res { + u16 wSrcNodeType; /* = HPI_SOURCENODE_LINEIN etc */ + u16 wSrcNodeIndex; /* = 0..N */ + u16 wDstNodeType; + u16 wDstNodeIndex; + /* Also controlType for MixerGetControlByIndex */ + u16 wControlIndex; + /* may indicate which DSP the control is located on */ + u16 wDspIndex; +}; + +union hpi_mixerx_msg { + struct { + u16 wStartingIndex; + u16 wFlags; + u32 dwLengthInBytes; /* length in bytes of pData */ + u32 pData; /* pointer to a data array */ + } gcabi; + struct { + u16 wCommand; + u16 wIndex; + } store; /* for HPI_MIXER_STORE message */ +}; + +union hpi_mixerx_res { + struct { + u32 dwBytesReturned; /* size of items returned */ + u32 pData; /* pointer to data array */ + u16 wMoreToDo; /* indicates if there is more to do */ + } gcabi; +}; + +struct hpi_control_msg { + u32 dwParam1; /* generic parameter 1 */ + u32 dwParam2; /* generic parameter 2 */ + short anLogValue[HPI_MAX_CHANNELS]; + u16 wAttribute; /* control attribute or property */ + u16 wControlIndex; +}; + +struct hpi_control_union_msg { + union { + struct { + u32 dwParam1; /* generic parameter 1 */ + u32 dwParam2; /* generic parameter 2 */ + short anLogValue[HPI_MAX_CHANNELS]; + } old; + union { + u32 dwFrequency; + u32 dwGain; + u32 dwBand; + u32 dwDeemphasis; + u32 dwProgram; + struct { + u32 dwMode; + u32 dwValue; + } mode; + } tuner; + } u; + u16 wAttribute; /* control attribute or property */ + u16 wControlIndex; +}; + +struct hpi_control_res { + /* Could make union. dwParam, anLogValue never used in same response */ + u32 dwParam1; + u32 dwParam2; + short anLogValue[HPI_MAX_CHANNELS]; +}; + +union hpi_control_union_res { + struct { + u32 dwParam1; + u32 dwParam2; + short anLogValue[HPI_MAX_CHANNELS]; + } old; + union { + u32 dwBand; + u32 dwFrequency; + u32 dwGain; + u32 dwLevel; + u32 dwDeemphasis; + struct { + u32 dwData[2]; + u32 dwBLER; + } rds; + } tuner; + struct { + char szData[8]; + u32 dwRemainingChars; + } chars8; +}; + +/* HPI_CONTROLX_STRUCTURES */ + +/* Message */ + +/** Used for all HMI variables where max length <= 8 bytes +*/ +struct hpi_controlx_msg_cobranet_data { + u32 dwHmiAddress; + u32 dwByteCount; + u32 dwData[2]; +}; + +/** Used for string data, and for packet bridge +*/ +struct hpi_controlx_msg_cobranet_bigdata { + u32 dwHmiAddress; + u32 dwByteCount; + u8 *pbData; +#ifndef HPI64BIT + u32 dwPadding; +#endif +}; + +/** Used for PADS control reading of string fields. +*/ +struct hpi_controlx_msg_pad_data { + u32 dwField; + u32 dwByteCount; + u8 *pbData; +#ifndef HPI64BIT + u32 dwPadding; +#endif +}; + +/** Used for generic data +*/ + +struct hpi_controlx_msg_generic { + u32 dwParam1; + u32 dwParam2; +}; + +struct hpi_controlx_msg { + union { + struct hpi_controlx_msg_cobranet_data cobranet_data; + struct hpi_controlx_msg_cobranet_bigdata cobranet_bigdata; + struct hpi_controlx_msg_generic generic; + struct hpi_controlx_msg_pad_data pad_data; + /* nothing extra to send for status read */ + } u; + u16 wControlIndex; + u16 wAttribute; /* control attribute or property */ +}; + +/* Response */ + +/** +*/ +struct hpi_controlx_res_cobranet_data { + u32 dwByteCount; + u32 dwData[2]; +}; + +struct hpi_controlx_res_cobranet_bigdata { + u32 dwByteCount; +}; + +struct hpi_controlx_res_cobranet_status { + u32 dwStatus; + u32 dwReadableSize; + u32 dwWriteableSize; +}; + +struct hpi_controlx_res_generic { + u32 dwParam1; + u32 dwParam2; +}; + +struct hpi_controlx_res { + union { + struct hpi_controlx_res_cobranet_bigdata cobranet_bigdata; + struct hpi_controlx_res_cobranet_data cobranet_data; + struct hpi_controlx_res_cobranet_status cobranet_status; + struct hpi_controlx_res_generic generic; + } u; +}; + +struct hpi_nvmemory_msg { + u16 wIndex; + u16 wData; +}; + +struct hpi_nvmemory_res { + u16 wSizeInBytes; + u16 wData; +}; + +struct hpi_gpio_msg { + u16 wBitIndex; + u16 wBitData; +}; + +struct hpi_gpio_res { + u16 wNumberInputBits; + u16 wNumberOutputBits; + u16 wBitData[4]; +}; + +struct hpi_async_msg { + u32 dwEvents; + u16 wMaximumEvents; + u16 wPadding; +}; + +struct hpi_async_res { + union { + struct { + u16 wCount; + } count; + struct { + u32 dwEvents; + u16 wNumberReturned; + u16 wPadding; + } get; + struct hpi_async_event event; + } u; +}; + +struct hpi_watchdog_msg { + u32 dwTimeMs; +}; + +struct hpi_watchdog_res { + u32 dwTimeMs; +}; + +struct hpi_clock_msg { + u16 wHours; + u16 wMinutes; + u16 wSeconds; + u16 wMilliSeconds; +}; + +struct hpi_clock_res { + u16 wSizeInBytes; + u16 wHours; + u16 wMinutes; + u16 wSeconds; + u16 wMilliSeconds; + u16 wPadding; +}; + +struct hpi_profile_msg { + u16 wIndex; + u16 wPadding; +}; + +struct hpi_profile_res_open { + u16 wMaxProfiles; +}; + +struct hpi_profile_res_time { + u32 dwMicroSeconds; + u32 dwCallCount; + u32 dwMaxMicroSeconds; + u32 dwMinMicroSeconds; + u16 wSeconds; +}; + +struct hpi_profile_res_name { +/* u8 messes up response size for 56301 DSP */ + u16 szName[16]; +}; + +struct hpi_profile_res { + union { + struct hpi_profile_res_open o; + struct hpi_profile_res_time t; + struct hpi_profile_res_name n; + } u; +}; + +struct hpi_message_header { + u16 wSize; + u16 wType; /* HPI_MSG_MESSAGE, HPI_MSG_RESPONSE */ + u16 wObject; /* HPI_OBJ_* */ + u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ + u16 wAdapterIndex; /* the adapter index */ + u16 wDspIndex; /* the dsp index on the adapter */ +}; + +struct hpi_message { + /* following fields must match HPI_MESSAGE_HEADER */ + u16 wSize; + u16 wType; /* HPI_TYPE_MESSAGE, HPI_TYPE_RESPONSE */ + u16 wObject; /* HPI_OBJ_* */ + u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ + u16 wAdapterIndex; /* the adapter index */ + u16 wDspIndex; /* the dsp index on the adapter */ + union { + struct hpi_subsys_msg s; + struct hpi_adapter_msg a; + union hpi_adapterx_msg ax; + struct hpi_stream_msg d; + struct hpi_mixer_msg m; + union hpi_mixerx_msg mx; /* extended mixer; */ + struct hpi_control_msg c; /* mixer control; */ + /* identical to struct hpi_control_msg, + but field naming is improved */ + struct hpi_control_union_msg cu; + struct hpi_controlx_msg cx; /* extended mixer control; */ + struct hpi_nvmemory_msg n; + struct hpi_gpio_msg l; /* digital i/o */ + struct hpi_watchdog_msg w; + struct hpi_clock_msg t; /* dsp time */ + struct hpi_profile_msg p; + struct hpi_async_msg as; + } u; +}; + +#define HPI_MESSAGE_SIZE_BY_OBJECT { \ + sizeof(struct hpi_message_header) , /* Default, no object type 0 */ \ + sizeof(struct hpi_message_header) + sizeof(struct hpi_subsys_msg),\ + sizeof(struct hpi_message_header) + sizeof(union hpi_adapterx_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_stream_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_stream_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_mixer_msg),\ + sizeof(struct hpi_message_header) , /* no node message */ \ + sizeof(struct hpi_message_header) + sizeof(struct hpi_control_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_nvmemory_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_gpio_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_watchdog_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_clock_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_profile_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_controlx_msg),\ + sizeof(struct hpi_message_header) + sizeof(struct hpi_async_msg) \ +} + +struct hpi_response_header { + u16 wSize; + u16 wType; /* HPI_MSG_MESSAGE, HPI_MSG_RESPONSE */ + u16 wObject; /* HPI_OBJ_* */ + u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ + u16 wError; /* HPI_ERROR_xxx */ + u16 wSpecificError; /* Adapter specific error */ +}; + +struct hpi_response { +/* following fields must match HPI_RESPONSE_HEADER */ + u16 wSize; + u16 wType; /* HPI_MSG_MESSAGE, HPI_MSG_RESPONSE */ + u16 wObject; /* HPI_OBJ_* */ + u16 wFunction; /* HPI_SUBSYS_xxx, HPI_ADAPTER_xxx */ + u16 wError; /* HPI_ERROR_xxx */ + u16 wSpecificError; /* Adapter specific error */ + union { + struct hpi_subsys_res s; + struct hpi_adapter_res a; + union hpi_adapterx_res ax; + struct hpi_stream_res d; + struct hpi_mixer_res m; + union hpi_mixerx_res mx; /* extended mixer; */ + struct hpi_control_res c; /* mixer control; */ + /* identical to hpi_control_res, but field naming is improved */ + union hpi_control_union_res cu; + struct hpi_controlx_res cx; /* extended mixer control; */ + struct hpi_nvmemory_res n; + struct hpi_gpio_res l; /* digital i/o */ + struct hpi_watchdog_res w; + struct hpi_clock_res t; /* dsp time */ + struct hpi_profile_res p; + struct hpi_async_res as; + u8 bytes[52]; + } u; +}; + +#define HPI_RESPONSE_SIZE_BY_OBJECT { \ + sizeof(struct hpi_response_header) ,/* Default, no object type 0 */ \ + sizeof(struct hpi_response_header) + sizeof(struct hpi_subsys_res),\ + sizeof(struct hpi_response_header) + sizeof(union hpi_adapterx_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_stream_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_stream_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_mixer_res),\ + sizeof(struct hpi_response_header) , /* no node response */ \ + sizeof(struct hpi_response_header) + sizeof(struct hpi_control_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_nvmemory_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_gpio_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_watchdog_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_clock_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_profile_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_controlx_res),\ + sizeof(struct hpi_response_header) + sizeof(struct hpi_async_res) \ +} + +/*////////////////////////////////////////////////////////////////////////// */ +/* declarations for compact control calls */ +struct hpi_control_defn { + u8 wType; + u8 wChannels; + u8 wSrcNodeType; + u8 wSrcNodeIndex; + u8 wDestNodeType; + u8 wDestNodeIndex; +}; + +/*////////////////////////////////////////////////////////////////////////// */ +/* declarations for control caching (internal to HPI<->DSP interaction) */ + +/** A compact representation of (part of) a controls state. +Used for efficient transfer of the control state +between DSP and host or across a network +*/ +struct hpi_control_cache_info { + /** one of HPI_CONTROL_* */ + u8 ControlType; + /** The total size of cached information in 32-bit words. */ + u8 nSizeIn32bitWords; + /** The original index of the control on the DSP */ + u16 ControlIndex; +}; + +struct hpi_control_cache_single { + struct hpi_control_cache_info i; + union { + struct { /* volume */ + u16 anLog[2]; + } v; + struct { /* peak meter */ + u16 anLogPeak[2]; + u16 anLogRMS[2]; + } p; + struct { /* channel mode */ + u16 wMode; + } m; + struct { /* multiplexer */ + u16 wSourceNodeType; + u16 wSourceNodeIndex; + } x; + struct { /* level/trim */ + u16 anLog[2]; + } l; + struct { /* tuner - partial caching. + Some attributes go to the DSP. */ + u32 dwFreqInkHz; + u16 wBand; + u16 wLevel; + } t; + struct { /* AESEBU Rx status */ + u32 dwErrorStatus; + u32 dwSource; + } aes3rx; + struct { /* AESEBU Tx */ + u32 dwFormat; + } aes3tx; + struct { /* tone detector */ + u16 wState; + } tone; + struct { /* silence detector */ + u32 dwState; + u32 dwCount; + } silence; + struct { /* sample clock */ + u16 wSource; + u16 wSourceIndex; + u32 dwSampleRate; + } clk; + struct { /* generic control */ + u32 dw1; + u32 dw2; + } g; + } u; +}; + +struct hpi_control_cache_pad { + struct hpi_control_cache_info i; + u32 dwFieldValidFlags; + u8 cChannel[8]; + u8 cArtist[40]; + u8 cTitle[40]; + u8 cComment[200]; + u32 dwPTY; + u32 dwPI; + u32 dwTrafficSupported; + u32 dwTrafficAnouncement; +}; + +/*/////////////////////////////////////////////////////////////////////////// */ +/* declarations for 2^N sized FIFO buffer (internal to HPI<->DSP interaction) */ +struct hpi_fifo_buffer { + u32 dwSize; + u32 dwDSPIndex; + u32 dwHostIndex; +}; + +#ifndef DISABLE_PRAGMA_PACK1 +#pragma pack(pop) +#endif + +/* skip host side function declarations for DSP + compile and documentation extraction */ +void HPI_InitMessage( + struct hpi_message *phm, + u16 wObject, + u16 wFunction +); + +void HPI_InitResponse( + struct hpi_response *phr, + u16 wObject, + u16 wFunction, + u16 wError +); + +char HPI_HandleObject( + const u32 dwHandle +); + +void HPI_HandleToIndexes( + const u32 dwHandle, + u16 *pwAdapterIndex, + u16 *pwObjectIndex, + u16 *pwDspIndex +); + +u32 HPI_IndexesToHandle( + const char cObject, + const u16 wAdapterIndex, + const u16 wObjectIndex, + const u16 wDspIndex +); + +/*////////////////////////////////////////////////////////////////////////// */ + +/* main HPI entry point */ +HPI_HandlerFunc HPI_Message; + +/* UDP message */ +void HPI_MessageUDP( + struct hpi_message *phm, + struct hpi_response *phr, + unsigned int nTimeout +); + +/* used in PnP OS/driver */ +u16 HPI_SubSysCreateAdapter( + struct hpi_hsubsys *phSubSys, + struct hpi_resource *pResource, + u16 *pwAdapterIndex +); + +u16 HPI_SubSysDeleteAdapter( + struct hpi_hsubsys *phSubSys, + u16 wAdapterIndex +); + +void HPI_FormatToMsg( + struct hpi_msg_format *pMF, + struct hpi_format *pF +); +void HPI_StreamResponseToLegacy( + struct hpi_stream_res *pSR +); + +/*////////////////////////////////////////////////////////////////////////// */ +/* declarations for individual HPI entry points */ +HPI_HandlerFunc HPI_6000; +HPI_HandlerFunc HPI_6205; +HPI_HandlerFunc HPI_COMMON; + +#ifdef __cplusplus +/* *INDENT-OFF* */ +} +/* *INDENT-ON* */ +#endif +#endif /* _HPI_INTERNAL_H_ */ -- 1.5.4.3

2 6

[alsa-devel] ASoC widgets questions
by Lopez Cruz, Misael 29 Jul '08

29 Jul '08

Hi all, I started to read about ASoC, but I didn't find much information apart from the kernel documentation files. Is there any other documents? Below are some questions mainly related to DAPM for codec driver. How should the interconnections map be created? What is the difference between using "SND_SOC_DAPM_PGA" and "SND_SOC_DAPM_MICBIAS" for an input? When to use each one? Why in some codec drivers there is a distinction between left and right ADC/DACs when creating DAPM widgets and in others not? How should be handled digital gains and mixing (after ADCs/before DACs)? Regards, Misael

2 1

[alsa-devel] app-internal volume control
by Wolfgang Rosenauer 29 Jul '08

29 Jul '08

Hi, might be a dumb question for people knowing the externals of Alsa but I'm not sure :-( What is the best practice to control the volume of a Linux application? I could control Alsa'x mixer for Master or PCM but that looks like a bad thing to do since it changes the volume for every other app as well. Is there another mixer I could use which is application specific? Or do I need to implement an application internal filter to control the volume by modifying the audio stream? Thanks, Wolfgang

1 0

[alsa-devel] Exposing control of my device
by Jon Smirl 29 Jul '08

29 Jul '08

My device has 220 registers, some of these are quite long (32 bytes) and I have split them up into multiple 32b pseudo registers. That results in about 300 registers for a TAS5504 and 450 for a TAS5508. A large number of these registers are for BiQuads, DRC, loudness, etc. They're not general purpose ALSA functions. What is an efficient way to expose several hundred registers of this class? Making snd_kcontrols for all of them seems pretty inefficient. Is there some generic way to do register access from user space that I haven't found yet? Or should I make two special snd_kcontrols, one for read register and write register? -- Jon Smirl jonsmirl(a)gmail.com

2 2

[alsa-devel] [PATCH] asoc: restrict sample rate and size in Freescale MPC8610 sound drivers
by Timur Tabi 29 Jul '08

29 Jul '08

The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Please apply this patch for 2.6.26.1 and for 2.6.27-rc1. Without this patch, recording on the MPC8610 is just plain broken, so it is a must-fix for 2.6.26. sound/soc/fsl/fsl_dma.c | 7 ++++- sound/soc/fsl/fsl_ssi.c | 66 +++++++++++++++++++++++++++++++++++++++------- 2 files changed, 62 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc59..7ceea2b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33..da71876 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,7 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: stream ID of first stream to program hardware * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +80,7 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -283,6 +285,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) return ret; } + ssi_private->first_stream = substream; + /* * Section 16.5 of the MPC8610 reference manual says that the * SSI needs to be disabled before updating the registers we set @@ -340,6 +344,44 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) * this is bad is because at this point, the PCM driver has not * finished initializing the DMA controller. */ + + } else { + /* This is not the first stream open, so we need to impose + * sample rate and maybe sample size constraints. Note that + * this can cause a race condition if the second stream is + * opened before the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -371,18 +413,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +447,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -462,6 +506,8 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); free_irq(ssi_private->irq, ssi_private); + + ssi_private->first_stream = NULL; } } -- 1.5.5

1 1

[alsa-devel] [PATCH 1/2] ALSA: Add jack reporting API
by Mark Brown 28 Jul '08

28 Jul '08

Currently very few systems provide information about jack status to user space, even though many have hardware facilities to do detection. Those systems that do use an input device with the existing SW_HEADPHONE_INSERT switch type to do so, often independently of ALSA. This patch introduces a standard method for representing jacks to user space into ALSA. It allows drivers to register jacks for a sound card with the input subsystem, binding the input device to the card to help user space associate the input devices with their sound cards. The created input devices are named in the form "card longname jack" where jack is provided by the driver when allocating a jack. By default the parent for the input device is the sound card but this can be overridden by the card driver. The existing user space API with SW_HEADPHONE_INSERT is preserved. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/core.h | 1 + include/sound/jack.h | 75 +++++++++++++++++++++++ sound/core/jack.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 include/sound/jack.h create mode 100644 sound/core/jack.c diff --git a/include/sound/core.h b/include/sound/core.h index 558b962..1a4ff0b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -63,6 +63,7 @@ typedef int __bitwise snd_device_type_t; #define SNDRV_DEV_INFO ((__force snd_device_type_t) 0x1006) #define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007) #define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008) +#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009) #define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000) typedef int __bitwise snd_device_state_t; diff --git a/include/sound/jack.h b/include/sound/jack.h new file mode 100644 index 0000000..b1b2b8b --- /dev/null +++ b/include/sound/jack.h @@ -0,0 +1,75 @@ +#ifndef __SOUND_JACK_H +#define __SOUND_JACK_H + +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics plc + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <sound/core.h> + +struct input_dev; + +/** + * Jack types which can be reported. These values are used as a + * bitmask. + */ +enum snd_jack_types { + SND_JACK_HEADPHONE = 0x0001, + SND_JACK_MICROPHONE = 0x0002, + SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, +}; + +struct snd_jack { + struct input_dev *input_dev; + int registered; + int type; + const char *id; + char name[100]; +}; + +#ifdef CONFIG_SND_JACK + +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack); +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent); + +void snd_jack_report(struct snd_jack *jack, int status); + +#else + +static inline int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack) +{ + return 0; +} + +static inline void snd_jack_set_parent(struct snd_jack *jack, + struct device *parent) +{ +} + +static inline void snd_jack_report(struct snd_jack *jack, int status) +{ +} + +#endif + +#endif diff --git a/sound/core/jack.c b/sound/core/jack.c new file mode 100644 index 0000000..8133a2b --- /dev/null +++ b/sound/core/jack.c @@ -0,0 +1,163 @@ +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/input.h> +#include <sound/jack.h> +#include <sound/core.h> + +static int snd_jack_dev_free(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + + /* If the input device is registered with the input subsystem + * then we need to use a different deallocator. */ + if (jack->registered) + input_unregister_device(jack->input_dev); + else + input_free_device(jack->input_dev); + + kfree(jack); + + return 0; +} + +static int snd_jack_dev_register(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + int err; + + snprintf(jack->name, sizeof(jack->name), "%s %s", + card->longname, jack->id); + jack->input_dev->name = jack->name; + + /* Default to the sound card device. */ + if (!jack->input_dev->dev.parent) + jack->input_dev->dev.parent = card->dev; + + err = input_register_device(jack->input_dev); + if (err == 0) + jack->registered = 1; + + return err; +} + +/** + * snd_jack_new - Create a new jack + * @card: the card instance + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jjack: Used to provide the allocated jack object to the caller. + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jjack will be initialised. + */ +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jjack) +{ + struct snd_jack *jack; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_jack_dev_free, + .dev_register = snd_jack_dev_register, + }; + + jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL); + if (jack == NULL) + return -ENOMEM; + + jack->id = id; + + jack->input_dev = input_allocate_device(); + if (jack->input_dev == NULL) { + err = -ENOMEM; + goto fail_input; + } + + jack->input_dev->phys = "ALSA"; + + jack->type = type; + + if (type & SND_JACK_HEADPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_HEADPHONE_INSERT); + if (type & SND_JACK_MICROPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_MICROPHONE_INSERT); + + err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); + if (err < 0) + goto fail_input; + + *jjack = jack; + + return 0; + +fail_input: + input_free_device(jack->input_dev); + kfree(jack); + return err; +} +EXPORT_SYMBOL(snd_jack_new); + +/** + * snd_jack_set_parent - Set the parent device for a jack + * + * @jack: The jack to configure + * @parent: The device to set as parent for the jack. + * + * Set the parent for the jack input device in the device tree. This + * function is only valid prior to registration of the jack. If no + * parent is configured then the parent device will be the sound card. + */ +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) +{ + WARN_ON(jack->registered); + + jack->input_dev->dev.parent = parent; +} +EXPORT_SYMBOL(snd_jack_set_parent); + +/** + * snd_jack_report - Report the current status of a jack + * + * @jack: The jack to report status for + * @status: The current status of the jack + */ +void snd_jack_report(struct snd_jack *jack, int status) +{ + if (jack->type & SND_JACK_HEADPHONE) + input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, + status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_MICROPHONE) + input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, + status & SND_JACK_MICROPHONE); + + input_sync(jack->input_dev); +} +EXPORT_SYMBOL(snd_jack_report); + +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("Jack detection support for ALSA"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

1 1

[alsa-devel] [PATCH 1/2] ALSA: Add jack reporting API
by Mark Brown 28 Jul '08

28 Jul '08

Currently very few systems provide information about jack status to user space, even though many have hardware facilities to do detection. Those systems that do use an input device with the existing SW_HEADPHONE_INSERT switch type to do so, often independently of ALSA. This patch introduces a standard method for representing jacks to user space into ALSA. It allows drivers to register jacks for a sound card with the input subsystem, binding the input device to the card to help user space associate the input devices with their sound cards. The created input devices are named in the form "card longname jack" where jack is provided by the driver when allocating a jack. By default the parent for the input device is the sound card but this can be overridden by the card driver. The existing user space API with SW_HEADPHONE_INSERT is preserved. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/core.h | 1 + include/sound/jack.h | 75 +++++++++++++++++++++++ sound/core/jack.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 include/sound/jack.h create mode 100644 sound/core/jack.c diff --git a/include/sound/core.h b/include/sound/core.h index 558b962..1a4ff0b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -63,6 +63,7 @@ typedef int __bitwise snd_device_type_t; #define SNDRV_DEV_INFO ((__force snd_device_type_t) 0x1006) #define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007) #define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008) +#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009) #define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000) typedef int __bitwise snd_device_state_t; diff --git a/include/sound/jack.h b/include/sound/jack.h new file mode 100644 index 0000000..16521a3 --- /dev/null +++ b/include/sound/jack.h @@ -0,0 +1,75 @@ +#ifndef __SOUND_JACK_H +#define __SOUND_JACK_H + +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics plc + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <sound/core.h> + +struct input_dev; + +/** + * Jack types which can be reported. These values are used as a + * bitmask. + */ +enum snd_jack_types { + SND_JACK_HEADPHONE = 0x0001, + SND_JACK_MICROPHONE = 0x0002, + SND_JACK_HEADSET = 0x0003, /* Both microphone and headphone */ +}; + +struct snd_jack { + struct input_dev *input_dev; + int registered; + int type; + const char *id; + char name[100]; +}; + +#ifdef CONFIG_SND_JACK + +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack); +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent); + +void snd_jack_report(struct snd_jack *jack, int status); + +#else + +static inline int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack) +{ + return 0; +} + +static inline void snd_jack_set_parent(struct snd_jack *jack, + struct device *parent) +{ +} + +static inline void snd_jack_report(struct snd_jack *jack, int status) +{ +} + +#endif + +#endif diff --git a/sound/core/jack.c b/sound/core/jack.c new file mode 100644 index 0000000..9c05ec8 --- /dev/null +++ b/sound/core/jack.c @@ -0,0 +1,163 @@ +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/input.h> +#include <sound/jack.h> +#include <sound/core.h> + +static int snd_jack_dev_free(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + + /* If the input device is registered with the input subsystem + * then we need to use a different deallocator. */ + if (jack->registered) + input_unregister_device(jack->input_dev); + else + input_free_device(jack->input_dev); + + kfree(jack); + + return 0; +} + +static int snd_jack_dev_register(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + int err; + + snprintf(jack->name, sizeof(jack->name), "%s %s", + card->longname, jack->id); + jack->input_dev->name = jack->name; + + /* Default to the sound card device. */ + if (!jack->input_dev->dev.parent) + jack->input_dev->dev.parent = card->dev; + + err = input_register_device(jack->input_dev); + if (err == 0) + jack->registered = 1; + + return err; +} + +/** + * snd_jack_new - Create a new jack + * @card: the card instance + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jjack: Used to provide the allocated jack object to the caller. + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jjack will be initialised. + */ +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jjack) +{ + struct snd_jack *jack; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_jack_dev_free, + .dev_register = snd_jack_dev_register, + }; + + jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL); + if (jack == NULL) + return -ENOMEM; + + jack->id = id; + + jack->input_dev = input_allocate_device(); + if (jack->input_dev == NULL) { + err = -ENOMEM; + goto fail_input; + } + + jack->input_dev->phys = "ALSA"; + + jack->type = type; + + if (type & SND_JACK_HEADPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_HEADPHONE_INSERT); + if (type & SND_JACK_MICROPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_MICROPHONE_INSERT); + + err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); + if (err < 0) + goto fail_input; + + *jjack = jack; + + return 0; + +fail_input: + input_free_device(jack->input_dev); + kfree(jack); + return err; +} +EXPORT_SYMBOL(snd_jack_new); + +/** + * snd_jack_set_parent - Set the parent device for a jack + * + * @jack: The jack to configure + * @parent: The device to set as parent for the jack. + * + * Set the parent for the jack input device in the device tree. This + * function is only valid prior to registration of the jack. If no + * parent is configured then the parent device will be the sound card. + */ +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) +{ + BUG_ON(jack->registered); + + jack->input_dev->dev.parent = parent; +} +EXPORT_SYMBOL(snd_jack_set_parent); + +/** + * snd_jack_report - Report the current status of a jack + * + * @jack: The jack to report status for + * @status: The current status of the jack + */ +void snd_jack_report(struct snd_jack *jack, int status) +{ + if (jack->type & SND_JACK_HEADPHONE) + input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, + status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_MICROPHONE) + input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, + status & SND_JACK_MICROPHONE); + + input_sync(jack->input_dev); +} +EXPORT_SYMBOL(snd_jack_report); + +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("Jack detection support for ALSA"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 3

[alsa-devel] underrun!!! (at least 1786051083.613 ms long)
by Timur Tabi 28 Jul '08

28 Jul '08

I have a driver that supports simultaneous playback and capture, but only if the same size and sample rate are the same. I use these commands to play and capture audio: aplay -D hw:0,0 -t raw -f S24_BE -r 48000 -c 2 test2.raw arecord -D hw:0,0 -t raw -f S24_BE -r 48000 -c 2 test.raw If I start capture first and playback second, then everything works. If I start playback first and capture second, I get this: underrun!!! (at least 1786051083.587 ms long) underrun!!! (at least 1786051083.586 ms long) underrun!!! (at least 1786051083.585 ms long) underrun!!! (at least 1786051083.585 ms long) ... This repeats indefinitely. Can someone explain how this is even possible? 1786051083.585ms is longer than 20 days! How could ALSA be so wrong about the extend of the underrun? -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] sound/soc/pxa/tosa.c: removed duplicated include
by Huang Weiyi 28 Jul '08

28 Jul '08

Removed duplicated include <asm/arch/tosa.h> in sound/soc/pxa/tosa.c. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fe6cca9..22971a0 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -33,7 +33,6 @@ #include <asm/arch/pxa-regs.h> #include <asm/arch/hardware.h> #include <asm/arch/audio.h> -#include <asm/arch/tosa.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h"

3 4

[alsa-devel] [PATCH 0/2] ALSA jack reporting API
by Mark Brown 28 Jul '08

28 Jul '08

This patch series is a repost of the jack reporting API previously discussed on this list. Patch 1 is a cherry pick of the commit from the input subsystem (available from Linus' current tree) adding a switch type for reporting microphone insertion which could be skipped if Linus' current tree were merged in to ALSA.

2 7

Re: [alsa-devel] [PATCH] cs4232: fix crash during chip PNP detection
by krzysztof.h1＠poczta.fm 28 Jul '08

28 Jul '08

Rene Herman napisał(a): > On 23-07-08 07:48, Krzysztof Helt wrote: > > > Looking back, it seems this was broken at least as far back as 2.6.20 > which would nicely show how much testing that code path gets. Do I > understand correctly from the backtrace that you have the PNPBIOS > hardware? If so, good... > Yes, I borrowed old laptop with PNP bios but no ISA PnP devices. I will test my wss_lib on it. > > --- linux-2.6.26/sound/isa/cs423x/cs4236.c~ 2008-07-23 > 07:40:16.129637645 +0200 > > +++ linux-2.6.26/sound/isa/cs423x/cs4236.c 2008-07-23 07:40:32.905641806 > +0200 > > @@ -325,6 +325,7 @@ static int __devinit snd_cs423x_pnp_init > > static int __devinit snd_card_cs4232_pnp(int dev, struct > snd_card_cs4236 *acard, > > struct pnp_dev *pdev) > > { > > + acard->wss = pdev; > > if (snd_cs423x_pnp_init_wss(dev, acard->wss) > 0) > > return -EBUSY; > > cport[dev] = -1; > > > > Acked-by: Rene Herman >rene.herman(a)gmail.com> > > (I believe I noticed earlier that struct snd_cs4236->wss, ctrl and mpu > really needn't/shouldn't be kept around in the first place and you could > just pass pdev directly as well but that's for another cleanup; this has > it look most similar to the card variant) > Feel free to changed it but I prefer to have the wss_lib patches merged first to avoid conflicts and redoing the patches. Krzysztof ---------------------------------------------------------------------- Galeria absurdow. zobacz >>> http://link.interia.pl/f1e5e

4 4

[alsa-devel] [PATCH 1/2] Support internal I2S clock sources on MPC5200
by Jon Smirl 27 Jul '08

27 Jul '08

Support internal I2S clock sources on MPC5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/mpc5200_psc_i2s.c | 58 ++++++++++++++++++++++++++++++++++----- sound/soc/fsl/mpc5200_psc_i2s.h | 13 +++++++++ 2 files changed, 64 insertions(+), 7 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 8692329..f028f61 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -23,8 +23,12 @@ #include <sysdev/bestcomm/bestcomm.h> #include <sysdev/bestcomm/gen_bd.h> +#include <asm/time.h> +#include <asm/mpc52xx.h> #include <asm/mpc52xx_psc.h> +#include "mpc5200_psc_i2s.h" + MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); MODULE_LICENSE("GPL"); @@ -93,6 +97,7 @@ struct psc_i2s { struct snd_soc_dai dai; spinlock_t lock; u32 sicr; + uint sysclk; /* per-stream data */ struct psc_i2s_stream playback; @@ -224,6 +229,7 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + uint bits, framesync, bitclk, value; u32 mode; dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i" @@ -235,15 +241,19 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: mode = MPC52xx_PSC_SICR_SIM_CODEC_8; + bits = 8; break; case SNDRV_PCM_FORMAT_S16_BE: mode = MPC52xx_PSC_SICR_SIM_CODEC_16; + bits = 16; break; case SNDRV_PCM_FORMAT_S24_BE: mode = MPC52xx_PSC_SICR_SIM_CODEC_24; + bits = 24; break; case SNDRV_PCM_FORMAT_S32_BE: mode = MPC52xx_PSC_SICR_SIM_CODEC_32; + bits = 32; break; default: dev_dbg(psc_i2s->dev, "invalid format\n"); @@ -251,7 +261,24 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, } out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode); - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + if (psc_i2s->sysclk) { + framesync = bits * 2; + bitclk = (psc_i2s->sysclk) / (params_rate(params) * framesync); + + /* bitclk field is byte swapped due to mpc5200/b compatibility */ + value = ((framesync - 1) << 24) | + (((bitclk - 1) & 0xFF) << 16) | ((bitclk - 1) & 0xFF00); + + dev_dbg(psc_i2s->dev, "%s(substream=%p) rate=%i sysclk=%i" + " framesync=%i bitclk=%i reg=%X\n", + __FUNCTION__, substream, params_rate(params), psc_i2s->sysclk, + framesync, bitclk, value); + + out_be32(&psc_i2s->psc_regs->ccr, value); + out_8(&psc_i2s->psc_regs->ctur, bits - 1); + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } @@ -429,9 +456,29 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct psc_i2s *psc_i2s = cpu_dai->private_data; - dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", - cpu_dai, dir); - return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; + int clkdiv, err; + dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, freq=%u, dir=%i)\n", + cpu_dai, freq, dir); + if (dir == SND_SOC_CLOCK_OUT) { + psc_i2s->sysclk = freq; + if (clk_id == MPC52xx_CLK_CELLSLAVE) { + psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | MPC52xx_PSC_SICR_GENCLK; + } else { /* MPC52xx_CLK_INTERNAL */ + psc_i2s->sicr &= ~MPC52xx_PSC_SICR_CELLSLAVE; + psc_i2s->sicr |= MPC52xx_PSC_SICR_GENCLK; + + clkdiv = ppc_proc_freq / freq; + err = ppc_proc_freq % freq; + if (err > freq / 2) + clkdiv++; + + dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(clkdiv %d freq error=%ldHz)\n", + clkdiv, (ppc_proc_freq / clkdiv - freq)); + + return mpc52xx_set_psc_clkdiv(psc_i2s->dai.id + 1, clkdiv); + } + } + return 0; } /** @@ -784,9 +831,6 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, /* Configure the serial interface mode; defaulting to CODEC8 mode */ psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL; - if (of_get_property(op->node, "fsl,cellslave", NULL)) - psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | - MPC52xx_PSC_SICR_GENCLK; out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h new file mode 100644 index 0000000..0e0a84e --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.h @@ -0,0 +1,13 @@ +/* + * Freescale MPC5200 PSC in I2S mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ + +#define MPC52xx_CLK_INTERNAL 0 +#define MPC52xx_CLK_CELLSLAVE 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */

5 16

[alsa-devel] [2.6 patch] sound/pci/azt3328.h: no variables for enums
by Adrian Bunk 27 Jul '08

27 Jul '08

AZF_FREQUENCIES and AZF_GAME_CONFIGS were variables, and this doesn't seem to have been intended. Signed-off-by: Adrian Bunk <bunk(a)kernel.org> --- d73b516dc7770a7141b6015b24c806fc697f066d diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 7e3e894..974e051 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -94,7 +94,7 @@ enum azf_freq_t { AZF_FREQ(48000), AZF_FREQ(66200), #undef AZF_FREQ -} AZF_FREQUENCIES; +}; /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ @@ -210,7 +210,7 @@ enum azf_freq_t { enum { AZF_GAME_LEGACY_IO_PORT = 0x200 -} AZF_GAME_CONFIGS; +}; #define IDX_GAME_LEGACY_COMPATIBLE 0x00 /* in some operation mode, writing anything to this port

3 2

[alsa-devel] [PATCH] Allow to force model to intel-mac-v3 in snd_hda_intel (sigmatel).
by Nicolas Boichat 27 Jul '08

27 Jul '08

Hi all, Currently, even if you pass model=intel-mac-v3 as a module parameter to snd_hda_intel, the function patch_stac922x (patch_sigmatel.c) will still try to auto-detect the model type. This is a problem on my MacBook Pro 1st generation, which needs intel-mac-v3, but sometimes incorrectly reports 0x00000100 as subsystem id, which causes the switch in patch_stac922x to select intel-mac-v4. To fix this, I added a new model called intel-mac-auto, so in case no module parameter is passed, and an Intel Mac board is detected, the model will be automatically detected, while no detection will be done if the model is forced to intel-mac-v3. This problem has been around for quite a while, and I used to fix it by moving the case statement for 0x00000100 in patch_stac922x so that intel-mac-v3 is chosen. Another way to fix the problem would be to check if a module parameter was set directly in patch_stac922x, using something like this: if (spec->board_config == STAC_INTEL_MAC_V3 && !codec->bus->modelname) { But I think it is less elegant (if you prefer that way, I can prepare a patch). This applies against 2.6.26, and should apply without any problem on the latest git. Thanks, Best regards, Nicolas Signed-off-by: Nicolas Boichat <nicolas(a)boichat.ch> --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_sigmatel.c | 14 +++++++++++--- 2 files changed, 12 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 0bbee38..705ca75 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1017,6 +1017,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. intel-mac-v3 Intel Mac Type 3 intel-mac-v4 Intel Mac Type 4 intel-mac-v5 Intel Mac Type 5 + intel-mac-auto Intel Mac (detect type according to subsystem id) macmini Intel Mac Mini (equivalent with type 3) macbook Intel Mac Book (eq. type 5) macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a0..8f71b48 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -94,6 +94,9 @@ enum { STAC_INTEL_MAC_V3, STAC_INTEL_MAC_V4, STAC_INTEL_MAC_V5, + STAC_INTEL_MAC_AUTO, /* This model is selected if no module parameter + * is given, one of the above models will be + * chosen according to the subsystem id. */ /* for backward compatibility */ STAC_MACMINI, STAC_MACBOOK, @@ -1473,6 +1476,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, [STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs, [STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs, + [STAC_INTEL_MAC_AUTO] = intel_mac_v3_pin_configs, /* for backward compatibility */ [STAC_MACMINI] = intel_mac_v3_pin_configs, [STAC_MACBOOK] = intel_mac_v5_pin_configs, @@ -1495,6 +1499,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_INTEL_MAC_V3] = "intel-mac-v3", [STAC_INTEL_MAC_V4] = "intel-mac-v4", [STAC_INTEL_MAC_V5] = "intel-mac-v5", + [STAC_INTEL_MAC_AUTO] = "intel-mac-auto", /* for backward compatibility */ [STAC_MACMINI] = "macmini", [STAC_MACBOOK] = "macbook", @@ -1566,9 +1571,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), /* other systems */ - /* Apple Mac Mini (early 2006) */ + /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, - "Mac Mini", STAC_INTEL_MAC_V3), + "Mac", STAC_INTEL_MAC_AUTO), /* Dell systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a7, "unknown Dell", STAC_922X_DELL_D81), @@ -3709,7 +3714,7 @@ static int patch_stac922x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, stac922x_cfg_tbl); - if (spec->board_config == STAC_INTEL_MAC_V3) { + if (spec->board_config == STAC_INTEL_MAC_AUTO) { spec->gpio_mask = spec->gpio_dir = 0x03; spec->gpio_data = 0x03; /* Intel Macs have all same PCI SSID, so we need to check @@ -3741,6 +3746,9 @@ static int patch_stac922x(struct hda_codec *codec) case 0x106b2200: spec->board_config = STAC_INTEL_MAC_V5; break; + default: + spec->board_config = STAC_INTEL_MAC_V3; + break; } }

2 1

[alsa-devel] [PATCH] hda: added 92HD81/83 support
by Matthew Ranostay 27 Jul '08

27 Jul '08

Added support for 92HD81/83 family of codecs. This also includes a pwr_mapping array for pins that have more than one amp to power down. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/hda_codec.h b/pci/hda/hda_codec.h index a8b7757..21d305d 100644 --- a/pci/hda/hda_codec.h +++ b/pci/hda/hda_codec.h @@ -121,6 +121,7 @@ enum { #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f +#define AC_VERB_SET_EAPD 0x788 #define AC_VERB_SET_CODEC_RESET 0x7ff /* diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index d11fd1a..90ecb2d 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -72,6 +72,11 @@ enum { }; enum { + STAC_92HD83XXX_REF, + STAC_92HD83XXX_MODELS +}; + +enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, @@ -142,6 +147,7 @@ struct sigmatel_spec { /* power management */ unsigned int num_pwrs; + unsigned int *pwr_mapping; hda_nid_t *pwr_nids; hda_nid_t *dac_list; @@ -237,6 +243,33 @@ static hda_nid_t stac92hd73xx_dmux_nids[2] = { 0x20, 0x21, }; +#define STAC92HD83XXX_NUM_DMICS 2 +static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { + 0x11, 0x12, 0 +}; + +#define STAC92HD81_DAC_COUNT 2 +#define STAC92HD83_DAC_COUNT 3 +static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = { + 0x13, 0x14, 0x22, +}; + +static hda_nid_t stac92hd83xxx_dmux_nids[2] = { + 0x17, 0x18, +}; + +static hda_nid_t stac92hd83xxx_adc_nids[2] = { + 0x15, 0x16, +}; + +static hda_nid_t stac92hd83xxx_pwr_nids[4] = { + 0xa, 0xb, 0xd, 0xe, +}; + +static unsigned int stac92hd83xxx_pwr_mapping[4] = { + 0x03, 0x0c, 0x10, 0x40, +}; + static hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f }; @@ -350,6 +383,11 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x1e, 0x22 }; +static hda_nid_t stac92hd83xxx_pin_nids[14] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, 0x13, + 0x1d, 0x1e, 0x1f, 0x20 +}; static hda_nid_t stac92hd71bxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, @@ -627,6 +665,19 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { {} }; +static struct hda_verb stac92hd83xxx_core_init[] = { + /* start of config #1 */ + { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3}, + + /* start of config #2 */ + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* power state controls amps */ + { 0x01, AC_VERB_SET_EAPD, 1 << 2}, +}; + static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -697,6 +748,8 @@ static struct hda_verb stac927x_core_init[] = { static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, {} }; @@ -819,6 +872,33 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { { } /* end */ }; + +static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT), + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT), + + /* + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT), + */ + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), @@ -1329,6 +1409,27 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; +static unsigned int ref92hd83xxx_pin_configs[14] = { + 0x02214030, 0x02211010, 0x02a19020, 0x02170130, + 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, + 0x01451160, 0x98560170, +}; + +static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, +}; + +static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD71BXX_REF), +}; + static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, @@ -2581,8 +2682,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, } /* labels for mono mux outputs */ -static const char *stac92xx_mono_labels[3] = { - "DAC0", "DAC1", "Mixer" +static const char *stac92xx_mono_labels[4] = { + "DAC0", "DAC1", "Mixer", "DAC2" }; /* create mono mux for mono out on capable codecs */ @@ -2686,16 +2787,19 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, } continue; found: - wcaps = get_wcaps(codec, nid); + wcaps = get_wcaps(codec, nid) & + (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); - if (wcaps & AC_WCAP_OUT_AMP) { + if (wcaps) { sprintf(name, "%s Capture Volume", stac92xx_dmic_labels[dimux->num_items]); err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + (wcaps & AC_WCAP_OUT_AMP) ? + HDA_OUTPUT : HDA_INPUT)); if (err < 0) return err; } @@ -2819,8 +2923,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out hp_speaker_swap = 1; } if (spec->autocfg.mono_out_pin) { - int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin) - & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & + (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); u32 caps = query_amp_caps(codec, spec->autocfg.mono_out_pin, dir); hda_nid_t conn_list[1]; @@ -2842,21 +2946,26 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out !(wcaps & AC_WCAP_LR_SWAP)) spec->mono_nid = conn_list[0]; } - /* all mono outs have a least a mute/unmute switch */ - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "Mono Playback Switch", - HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, - 1, 0, dir)); - if (err < 0) - return err; - /* check to see if there is volume support for the amp */ - if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "Mono Playback Volume", - HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, - 1, 0, dir)); + if (dir) { + hda_nid_t nid = spec->autocfg.mono_out_pin; + + /* most mono outs have a least a mute/unmute switch */ + dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); if (err < 0) return err; + /* check for volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) + >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, + "Mono Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); + if (err < 0) + return err; + } } stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, @@ -2936,7 +3045,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = dig_out; - if (spec->autocfg.dig_in_pin) + if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; if (spec->kctl_alloc) @@ -3332,7 +3441,12 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx) val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0x000000ff; presence = get_hp_pin_presence(codec, nid); - idx = 1 << idx; + + /* several codecs have two power down bits */ + if (spec->pwr_mapping) + idx = spec->pwr_mapping[idx]; + else + idx = 1 << idx; if (presence) val &= ~idx; @@ -3668,6 +3782,94 @@ again: return 0; } +static struct hda_input_mux stac92hd83xxx_dmux = { + .num_items = 3, + .items = { + { "Analog Inputs", 0x03 }, + { "Digital Mic 1", 0x04 }, + { "Digital Mic 2", 0x05 }, + } +}; + +static int patch_stac92hd83xxx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->mono_nid = 0x19; + spec->digbeep_nid = 0x21; + spec->dmic_nids = stac92hd83xxx_dmic_nids; + spec->dmux_nids = stac92hd83xxx_dmux_nids; + spec->adc_nids = stac92hd83xxx_adc_nids; + spec->pwr_nids = stac92hd83xxx_pwr_nids; + spec->pwr_mapping = stac92hd83xxx_pwr_mapping; + spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); + spec->multiout.dac_nids = stac92hd83xxx_dac_nids; + + spec->init = stac92hd83xxx_core_init; + switch (codec->vendor_id) { + case 0x111d7605: + spec->multiout.num_dacs = STAC92HD81_DAC_COUNT; + break; + default: + spec->num_pwrs--; + spec->init++; /* switch to config #2 */ + spec->multiout.num_dacs = STAC92HD83_DAC_COUNT; + } + + spec->mixer = stac92hd83xxx_mixer; + spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); + spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); + spec->num_dmics = STAC92HD83XXX_NUM_DMICS; + spec->dinput_mux = &stac92hd83xxx_dmux; + spec->pin_nids = stac92hd83xxx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD83XXX_MODELS, + stac92hd83xxx_models, + stac92hd83xxx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD83XXX, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + err = stac92xx_parse_auto_config(codec, 0x1d, 0); + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD83XXX_REF; + goto again; + } + err = -EINVAL; + } + + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -4386,6 +4588,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },

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Re: [alsa-devel] Patch - ESI Maya44 driver
by midragius 27 Jul '08

27 Jul '08

Dnia 9 maja 2008 12:51 Takashi Iwai <tiwai(a)suse.de> napisał(a): > At Fri, 9 May 2008 09:22:13 +0200, > Piotr Zaryk wrote: > > > > Hi, > > > > Is there anybody who could integrate the MAYA44 driver with ALSA? Rainer > > does not respond and I don't know what's going on. Maya44 driver should be > > integrated as soon as possible because of the improvements in the ice-1724 > > code. Anybody interested? > > Such a big patch cannot be merged to the upstream without a proper > sign-off. I mailed Rainer again but no response yet, unfortunately. > > > Takashi Rainer doesn't reply to email for a long time, is there any way to somebody else sign-off his driver? Maybe he has some trouble, it will be big waste if his driver will be foredoomed. I made simply update of his patch, so it can be merge with alsa-driver-1.0.17 Patryk Krawczyk

2 2

Re: [alsa-devel] Alsa-devel Digest, Vol 17, Issue 128
by Peter Dolding 27 Jul '08

27 Jul '08

> Peter, > > Two assumptions you make are problematic: > > 1. You assume that one FOSS project can "kill" another. This is false. > For a project to continue existing, the only thing it needs is people > working on the code to maintain it so that it works in the > ever-changing environment around it (the compiler, the kernel, core > libraries, etc.) > To be correct a FOSS project can kill another buy taking its developers. Its way simpler than you could think. If ALSA has all the features of PulseAudio including its interfacing API's why will you add PulseAudio. This is exactly how PulseAudio is killing off ESD. So in time not a single person is going to use ESD other than threw PulseAudio. Now PulseAudio killing Alsa is a lot harder and equally possible. So its a shark in the water. > 2. You assume that by creating synergy and prominence of a single > piece of software, that no one will try to write a competing piece of > software. This is false. You absolutely cannot stop people from > releasing alternative choices that have merits which draw users to > them. There is no such thing as "de-fragmenting" the FOSS space. You > can't make it simpler; you can only make it more complex. The number > of FOSS projects of any given category grows infinitely with time, > regardless of how good existing solutions are. This is true at every > level of the software stack, but some categories have more > alternatives than others. > > Sound traditionally happens to have a large amount of > actively-maintained, moderately-popular alternatives to the point of > being annoying, but there is no remedy that will make the others > irrelevant. As long as there is some moderately-popular application > being shipped with distributions that uses XYZ sound API / sound > server, XYZ will have a user base. This leads to the unfortunate fact > that, as far as I know, there does not currently exist _any_ > arrangement of nested sound servers and APIs that will: > > 1. Work for _every_ existing sound-using application (i.e. nobody > crashes, few or no drop-outs, no assertions or incompatibilities or > limitations, etc.) > 2. Permit concurrent access to the soundcard by multiple applications > at the same time > 3. Allow all the features of every API to work at the same time > Missed the worst avoid lag 2 sound servers like dmix and pulse operating on top of each other equals lag. Goal has to be to get to 1 sound server only. Remember ALSA name clearly. Advanced Linux Sound Architecture This is a Architecture single API s not a Architecture. If all the needed features to run ESD Pulse and so on there API can be directly wrapped down to ALSA removing the need for sound servers other than ALSA's. Those API's just become part of the Architecture and can be deprecated out of existence over time. We also get broader ideas and more developers focused in a single place. Its a simple case why install pulse if its features already work. Note Pulseaudio is defragmenting. The more API's it supports the more it gets threw its central path. Less important those engines until they are no longer maintained and dead. Why are you going to put something else functional than Pulse on a system. This method of defragmentation has happened many times over. It works. Its a myth that FOSS projects have to keep on growing in numbers in any give category. If that was so we would have a extra clone to ALSA by now. To be correct if we don't watch it we will. Its only a matter of time before pulse guys just want to go straight past the alsa-lib so they get faster interfacing with hardware. Question is how long. Note its imposable to get nested sound server to work perfectly. But its more than possible to get a single sound server to do all the jobs of all the other sound servers so giving perfect API coverage. The issue is we have to reduce to 1 sound server. Many wrapping api's. Its less than 10 API's total. Not many sound servers providing single api's. Only way is for 1 sound server to have all the features of all the other sound servers. Perfectively with direct calls to hardware. Alsa fits the direct calls to hardware bit. Features and wrappers a little behind. First project to 100 percent coverage wins. Having a single sound server also reduces travel to hardware down. You don't have to go sound server to sound server to get to hardware. Only way forward is to end the peace. Every project targeting to be last 1 standing. Even doing merges with each other to be last 1 standing. Peter Dolding

2 2

[alsa-devel] soc.h macros
by Jon Smirl 26 Jul '08

26 Jul '08

Can you help me document these macros? For example what does TLV stand for? I know it supplies a decibel range. R really mean two registers. What does EXT do? How about S8? #define SOC_SINGLE(xname, reg, shift, max, invert) #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) #define SOC_DOUBLE(xname, xreg, shift_left, shift_right, xmax, xinvert) #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) #define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) #define SOC_ENUM(xname, xenum) #define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert, xhandler_get, xhandler_put) #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert, xhandler_get, xhandler_put, tlv_array) #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) #define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) SOC_ENUM("3D Function", wm8753_enum[5]), SOC_ENUM seems really inefficient tracking those enum numbers. Could you declare the enum in place? #define SOC_ENUM_SINGLE(xname, xreg, xshift, xmax, xtexts) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ .info = snd_soc_info_enum_double, \ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \ .private_value = (unsigned long) &(struct soc_enum) \ .reg = xreg, .shift_l = xshift, .shift_r = xshift, \ .max = xmax, .texts = xtexts } Is it possible to turn all of these macros into inline functions? Can an inline initialize a structure? Making them inlines would get us syntax checking without expansion. I'm having some success getting 32b wide controls working. -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] alsa-lib has anyone ever looked at taking it cross platform?
by Peter Dolding 26 Jul '08

26 Jul '08

> Peter Dolding wrote: >> Pulseaudio and all other Sound Servers since its a extra service adds >> a extra point of complete sound failure to the Linux sound stack or >> what ever stack they happen to be operating on top of. It more of a >> hack around the limitations of the alsa framework. ALSA was really >> designed to avoid the need of such extras. The existence of >> Pulseaudio is really sign of failure. Reason Sound Servers existed at >> first due to the failures of OSS to provide a multi application sound >> support. ALSA was really designed to do away for the need of sound >> servers completely. Now alsa is failing to provide per application >> sound control and a sound server is back for another pass. > > I really don't see why the existence of sound servers is a failure of > ALSA. ALSA does it's job very well by providing the necessary hardware > drivers and an interface to them. The kind of things needed by a modern > desktop sound system, however, goes way beyond the 1:1 hardware interface. > > At the most basic level, sound servers can be reduced to just software > mixers. The name "sound server" has a very bad reputation that (these > days) is not deserved. To say that pulseaudio's existence is a failure > of ALSA is doing ALSA a massive disservice. Sure, the software mixing > can be done by Alsa's DMIX but DMIX is just a "sound server" anyway, so > what specifically makes DMIX better or worse than PA in this regard? > Because DMIX is started automatically? Because it's in kernel space? > None of these arguments hold up when you get involved at the level of a > distribution who packages things together correctly. > > Pulseaudio is just a name and the kind of things it does are very much > different to the core goals of ALSA (with the exception of the obvious > overlap between DMIX and PA). The two fit very well together IMO. Same line OSS used just with a few words changed Pulseaudio with ESD and ALSA with OSS. You are repeating history and don't even know it. > > If it makes you feel any better, just call pulseaudio "ALSA Desktop > Services". Like I say it's only a name! > Its a failure renaming failure does not change it. Over time you get more and more competing sound servers. How long before people of PA look at ALSA and go why in hell are we using that nasty thing and just go straight around alsa-lib and straight to alsa drivers so we don't have the performance bottle neck. This is exactly what PA is trying to do now you will talk straight to us so you must run PA. One everything is talking to PA they can go direct to hardware. No matter if it causes trouble or not. DMIX is userspace and was alway designed that ALSA could be expanded to take on any task. The difference is 1 sound server. Total 1. No more does OS need. ALSA was designed to be Linux's Sound Server and all the other low level sound parts. As soon as you have more you have trouble. PA is a second sound server that just provides a second point for sound failure. Basically PA should not exist. If alsa was interesting in being just hardware only DMIX did not have to be created we could have just linked to ESD and other sound servers of the time. DMIX is a clear sign that ALSA is ment to be far more. PA is a intruder everything PA is ALSA should be as well. PA is taking the wrong method to fixing the the MIX problem DMIX should have been upgraded. > >> Now if you want to keep on thinking along that line has ALSA got to >> the point it should be dumped like OSS has been dumped because of its >> limitations? > > Of course ALSA should not be dumped! ALSA does it's job very well, and > PA does it's very well too. I really don't see how you got to that > statement from what I said. PA doesn't try and reimplement ALSA, it just > implements the higher level audio stack. > >> alsa-lib to freebsd or somewhere with oss most likely could be done >> really quickly since we already have a oss output option. >> >> Its also like saying the Linux Standard Base is only used by Linux. >> Its not its also used by a few non Linux's. Driver section of ALSA >> most likely will always be Linux only. But the wrapper layer that >> hides the hardware from the software really does not have to stay >> Linux only. Alsa is going up to be part of the Linux Standard Base >> its about time we all start looking at how it can be made most useful. >> Provide all the features it should and work for stability. > > I've not programmed much with alsa lib but I do know it's not all that > simple. That's there are several other layers on top that exist. There > API is complex and powerful but it also open to a lot of abuse. There > are many ALSA client apps out there that are poorly written. That's > simply because the API is complex and it is easy to make mistakes that > do not surface with one setup but do on another. Making apps work nicely > with DMIX was a pain and as soon as you remove the hardware aspect of > the underlying system (e.g. to replace it with a plugin like bluez or > PA) lots of interesting issues come to the fore. > So it is like xlib. A wrapper lib that has got too complex for its own good. xlib also caused strange failures when connected to different brand X11 servers. xcb was all about simplifying a very complex process. Sound servers are also a way of avoiding have to fix these issues. Just like toolkits on X11 have been. If PA or ESD or any of the other sound servers can provide a simple usable API so should ALSA. Difference in operational complexity is 100 percent a error in ALSA. Of course more complex options can remain. Existence of sound servers is pointing to failures. I have not got onto the nasty bit of 3d sound not being simple inside alsa. Creative wanted to go straight past ALSA with openal. Somehow I think they should have been let destroy alsa completely if people here are not prepared to pull head out sand. Tidy direct hardware interface api is needed. ALSA-LIB need to be feature rich. It need to go cross platform so developers of PA, liboa and others get pulled in here. Or this will just rot and rot until the day comes that ALSA-LIB will have to be dumped because its no longer anything more than a processing black hole. One option if PA is so many time better. Is to complete save a lot of time and completely do way with ALSA-LIB and say to the PA guys here write a new direct interface to the drivers. Because that is what will happen in time if this project does not step up. The question is how many years. It is really a simple selection kill/let ALSA-LIB slowly die off or step up be a architecture and take PA and other sound servers on head on. If alsa-lib is not going to step up we might as well save the resources and focus them better. Really we do want all the sound server/driver/basic wrapper developers here. So there are more developer to deal with ALSA glitches. I think some people here have got way way too friendly with the completion. PA is not a save ALSA camp it is a long term kill bits of ALSA camp. Question is how far do they go before alsa starts protecting itself. Going cross platform is part of the protection. No reason to keep the sound system of linux segmented any more. ALSA was always ment to kill of the segmentation. Its currently failing nicely at that task. Peter Dolding

2 1

[alsa-devel] Requiring the same sample rate for playback and record?
by Timur Tabi 25 Jul '08

25 Jul '08

In my driver, how do I tell ALSA that the sample rate for playback must be the same as for capture? I know I can have my driver remember the sample rate for the first stream open, and then if the second stream doesn't specify the same sample rate, it can reject it, but this is inelegant. I'm hoping there's a more straightforward way. -- Timur Tabi Linux kernel developer at Freescale

3 11

[alsa-devel] alsa-lib has anyone ever looked at taking it cross platform?
by Peter Dolding 25 Jul '08

25 Jul '08

This would kinda make logical sence. Alsa-lib plugin system in theory should be able to take any kind sound output under it. Simplify development of all applications on top of it. If alsa was cross platform why would you use something else as a middle body wrapper when you port. Windows and Mac support could be done as plugins. Or is there something that is in the alsa-lib that is a problem? Its all about reducing distance between application and hardware and reduce amount of coding. Peter Dolding

3 2

Re: [alsa-devel] [PATCH 10/10] wss_lib: use wss detection code instead of ad1848 one
by krzysztof.h1＠poczta.fm 25 Jul '08

25 Jul '08

Rene Herman napisał(a): > >>>> Is this now done in generic code? Not necessary anymore? Was no > comment > >>>> in the changelog. > >>> The MODE_RUNNING is not used at all in the cs4231 code. I wonder what > the purpose of it. > >> It was used by the AD1848 code though; snd_ad1848_trigger() set/reset > it > >> on start/stop and it was then tested by snd_ad1848_interrupt() to > decide > >> whether or not to call snd_pcm_period_elapsed(). > > > > It is not used by the cs4231. The only difference is that ad1848 does > not > > call the snd_pcm_period_elapsed() after calling the snd_ad1848_open() > > but before calling the snd_ad1848_trigger() (and similar restriction > > after the snd_ad1848_trigger() and snd_ad1848_close(). > > > > The cs4231 does not use such restriction. I decided it does not really > matter. > > The interrupts are not enabled before and after the > snd_ad1848_trigger(). > > If the cs4231 driver needed this it would be already causing problems. > > It seems we are talking at cross purposes. I'm not talking about cs4231. > I see this MODE_RUNNING thing disappear from ad1848_lib and, unless I've > missed it, not resurface in wss_lib -- that library that after these > patches is used to drive AD1848 chips that used to be driven by > ad1848_lib. > > The MODE_RUNNING looks as something someone will have once added upon > seeing spurious IRQs and as such, testing that it isn't needed on > chip/card A, B and C doesn't tell us much. The problem may have been > observed on type D, E or F and/or under condition foo or bar. > > Looking at snd_wss_interrupt() I dom't seem to be seeing a similar guard > after these patches. > > If it's not needed, all for slashing it but please be convincing. Note > that by now you should know a lot more about the innards of this code > than I do, so please be as verbose as needed. > I cannot be convincing as I am not convinced myself. I understand your reasoning. My situation was that I saw the guard in the ad1848 which was missing in the cs4231 while they should look like twins (in the area when MODE_RUNNING existed). I had two choices: 1. Add MODE_RUNNING to the cs4231 2. Remove MODE_RUNNING The first approach is a bloat to me as I tested on ad1848 and it was not needed. I don't know the conditions you called the "condition foo or bar" and I don't know if it ever exists. I have checked git history on the file, but this one was added earlier. Maybe there was a kind of approach in the old days then dropped/replaced later (like mutex_open which is not needed now). The second approach is more dangerous but with enough testing should be justified and provides us with cleaner code. Regards, Krzysztof ---------------------------------------------------------------------- Te newsy nakreca Cie na caly dzien! Sprawdz >>> http://link.interia.pl/f1e94

1 0

Re: [alsa-devel] [PATCH] [alsa-utils] Fix build with library
by Bryan Hanks 25 Jul '08

25 Jul '08

Takashi, Here is the patch for alsa-utils not being able to compile against alsa-lib if certain features were disabled in the library. It uses AC_CHECK_HEADERS as you suggested. It disables midi or seq and alsactl (if either are disabled, since it makes use of rawmidi.h and seq.h) from the build. Bryan

1 0

[alsa-devel] Help
by Sudeept Prusti 24 Jul '08

24 Jul '08

Hi all, As i am working on ALSA for last few weeks.Can anyone guide where can i get moe informative things on ALSA internals (Internally how alsa works). Could anyone suggest what are validation scenario on alsa driver flow. Thanks a lot. Awaiting for response soon. Regards, Sudeept The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. Contact your Administrator for further information.

2 1

[alsa-devel] [Fwd: Filling "Struct snd_dma_buffer" manually]
by dinesh 24 Jul '08

24 Jul '08

-------- Forwarded Message -------- From: dinesh <dinesh.dua(a)coraltele.com> To: alsa-devel <alsa-devel(a)alsa-project.org> Subject: [alsa-devel] Filling "Struct snd_dma_buffer" manually Date: Thu, 24 Jul 2008 18:29:59 +0530 Hello, I am stuck with a problem please help me. I have allocated a buffer in ram and i want that if i playback something from user end then alsa library exchanges data with this buffer using memory mapping but for this i have to fill a structure "snd_dma_buffer" in driver code but as i have told i dont want to use dma so i have to fill it manually with address of ram buffer. Please tell me if it is possible if yes then help me. Thanks in advance.. Regards, Dinesh Dua _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

Re: [alsa-devel] [PATCH 2/2] Update Poodle to current ASoC API
by Mark Brown 24 Jul '08

24 Jul '08

On Thu, Jul 24, 2008 at 04:54:48PM +0400, Dmitry Baryshkov wrote: > Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> > Cc: Richard Purdie <rpurdie(a)rpsys.net> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > This should probably go into 2.6.27, if possible I've queued this for Takashi, thanks (he's on holiday this week). I had thought that at least the second hunk was already fixed but it seems not. Please CC all ALSA patches to alsa-devel(a)alsa-project.org.

3 6

[alsa-devel] Filling "Struct snd_dma_buffer" manually
by dinesh 24 Jul '08

24 Jul '08

Hello, I am stuck with a problem please help me. I have allocated a buffer in ram and i want that if i playback something from user end then alsa library exchanges data with this buffer using memory mapping but for this i have to fill a structure "snd_dma_buffer" in driver code but as i have told i dont want to use dma so i have to fill it manually with address of ram buffer. Please tell me if it is possible if yes then help me. Thanks in advance.. Regards, Dinesh Dua

1 0

[alsa-devel] [PATCH] ASOC, rename mask to max where appropriate
by Jon Smirl 24 Jul '08

24 Jul '08

Where appropriate, rename ASOC mask variables to max to reflect their use Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- include/sound/soc.h | 22 +++++++++++----------- sound/soc/soc-core.c | 20 ++++++++++---------- sound/soc/soc-dapm.c | 14 +++++++------- 3 files changed, 28 insertions(+), 28 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index cc64b04..412cd83 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -88,25 +88,25 @@ .put = snd_soc_put_volsw_s8, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .min = xmin, .max = xmax} } -#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ +#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .texts = xtexts } -#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \ - SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts) -#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \ -{ .mask = xmask, .texts = xtexts } + .max = xmax, .texts = xtexts } +#define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \ + SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts) +#define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \ +{ .max = xmax, .texts = xtexts } #define SOC_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ .info = snd_soc_info_enum_double, \ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \ .private_value = (unsigned long)&xenum } -#define SOC_SINGLE_EXT(xname, xreg, xshift, xmask, xinvert,\ +#define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } -#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -114,7 +114,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ @@ -534,7 +534,7 @@ struct soc_enum { unsigned short reg2; unsigned char shift_l; unsigned char shift_r; - unsigned int mask; + unsigned int max; const char **texts; void *dapm; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7c59665..c0e6ab4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1296,10 +1296,10 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->mask; + uinfo->value.enumerated.items = e->max; - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; + if (uinfo->value.enumerated.item > e->max - 1) + uinfo->value.enumerated.item = e->max - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); return 0; @@ -1322,7 +1322,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] @@ -1352,14 +1352,14 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, unsigned short val; unsigned short mask, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; - if (ucontrol->value.enumerated.item[0] > e->mask - 1) + if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; mask = (bitmask - 1) << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->mask - 1) + if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= (bitmask - 1) << e->shift_r; @@ -1386,10 +1386,10 @@ int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = e->mask; + uinfo->value.enumerated.items = e->max; - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; + if (uinfo->value.enumerated.item > e->max - 1) + uinfo->value.enumerated.item = e->max - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); return 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b956771..424e4e8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -125,13 +125,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; int val, item, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; val = snd_soc_read(w->codec, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; - for (i = 0; i < e->mask; i++) { + for (i = 0; i < e->max; i++) { if (!(strcmp(p->name, e->texts[i])) && item == i) p->connect = 1; } @@ -168,7 +168,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int i; - for (i = 0; i < e->mask; i++) { + for (i = 0; i < e->max; i++) { if (!(strcmp(control_name, e->texts[i]))) { list_add(&path->list, &codec->dapm_paths); list_add(&path->list_sink, &dest->sources); @@ -1264,7 +1264,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; val = snd_soc_read(widget->codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); @@ -1294,15 +1294,15 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned short mask, bitmask; int ret = 0; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; - if (ucontrol->value.enumerated.item[0] > e->mask - 1) + if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; mask = (bitmask - 1) << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->mask - 1) + if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= (bitmask - 1) << e->shift_r;

2 1

[alsa-devel] [PATCH] Clean up ASOC functions to support 32b values instead of 16b
by Jon Smirl 24 Jul '08

24 Jul '08

Clean up ASOC functions to support 32b values instead of 16b Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- include/sound/soc.h | 22 +++++++++++----------- sound/soc/soc-core.c | 35 +++++++++++++++++------------------ sound/soc/soc-dapm.c | 32 ++++++++++++++++---------------- 3 files changed, 44 insertions(+), 45 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 412cd83..ac59506 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -253,10 +253,10 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, #define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); +int snd_soc_update_bits(struct snd_soc_codec *codec, uint reg, + uint mask, uint value); +int snd_soc_test_bits(struct snd_soc_codec *codec, uint reg, + uint mask, uint value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); @@ -419,8 +419,8 @@ struct snd_soc_codec { hw_write_t hw_write; hw_read_t hw_read; void *reg_cache; - short reg_cache_size; - short reg_cache_step; + uint reg_cache_size; + uint reg_cache_step; /* dapm */ struct list_head dapm_widgets; @@ -530,11 +530,11 @@ struct soc_mixer_control { /* enumerated kcontrol */ struct soc_enum { - unsigned short reg; - unsigned short reg2; - unsigned char shift_l; - unsigned char shift_r; - unsigned int max; + uint reg; + uint reg2; + uint shift_l; + uint shift_r; + uint max; const char **texts; void *dapm; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c0e6ab4..358aa95 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -961,7 +961,7 @@ static ssize_t codec_reg_show(struct device *dev, { struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; - int i, step = 1, count = 0; + uint i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; @@ -1039,11 +1039,11 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); * * Returns 1 for change else 0. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) +int snd_soc_update_bits(struct snd_soc_codec *codec, u32 reg, + u32 mask, u32 value) { int change; - unsigned short old, new; + u32 old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1069,11 +1069,11 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * * Returns 1 for change else 0. */ -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) +int snd_soc_test_bits(struct snd_soc_codec *codec, u32 reg, + u32 mask, u32 value) { int change; - unsigned short old, new; + u32 old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1320,7 +1320,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + u32 val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1349,8 +1349,7 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask, bitmask; + u32 val, mask, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1512,7 +1511,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; uint mask = (1 << fls(max)) - 1; uint invert = mc->invert; - unsigned short val, val2, val_mask; + u32 val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -1618,7 +1617,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, uint mask = (1 << fls(max)) - 1; uint invert = mc->invert; int err; - unsigned short val, val2, val_mask; + u32 val, val2, val_mask; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] & mask); @@ -1683,12 +1682,12 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); uint reg = mc->reg; int min = mc->min; - int val = snd_soc_read(codec, reg); + uint val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; + ((signed char)(val & 0xff)) - min; ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; + ((signed char)((val >> 8) & 0xff)) - min; return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); @@ -1710,10 +1709,10 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); uint reg = mc->reg; int min = mc->min; - unsigned short val; + uint val; - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + val = (ucontrol->value.integer.value[0] + min) & 0xff; + val |= ((ucontrol->value.integer.value[1] + min) & 0xff) << 8; return snd_soc_update_bits(codec, reg, 0xffff, val); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 424e4e8..d88b21a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -103,7 +103,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, switch (w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: { - int val; + uint val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrols[i].private_value; uint reg = mc->reg; @@ -123,7 +123,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, break; case snd_soc_dapm_mux: { struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; - int val, item, bitmask; + uint val, item, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -206,8 +206,8 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, /* update dapm codec register bits */ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) { - int change, power; - unsigned short old, new; + uint change, power; + uint old, new; struct snd_soc_codec *codec = widget->codec; /* check for valid widgets */ @@ -221,7 +221,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) power = widget->power; if (widget->invert) - power = (power ? 0:1); + power = (power ? 0 : 1); old = snd_soc_read(codec, widget->reg); new = (old & ~(0x1 << widget->shift)) | (power << widget->shift); @@ -464,7 +464,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - unsigned int val; + uint val; if (SND_SOC_DAPM_EVENT_ON(event)) val = w->on_val; @@ -695,8 +695,8 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + struct snd_kcontrol *kcontrol, uint mask, + uint val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -733,8 +733,8 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, /* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int reg, - int val_mask, int val, int invert) + struct snd_kcontrol *kcontrol, uint reg, + uint val_mask, uint val, uint invert) { struct snd_soc_dapm_path *path; int found = 0; @@ -755,10 +755,10 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, found = 1; if (val) /* new connection */ - path->connect = invert ? 0:1; + path->connect = invert ? 0 : 1; else /* old connection must be powered down */ - path->connect = invert ? 1:0; + path->connect = invert ? 1 : 0; break; } @@ -1198,7 +1198,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; uint mask = (1 << fls(max)) - 1; uint invert = mc->invert; - unsigned short val, val2, val_mask; + uint val, val2, val_mask; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1262,7 +1262,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + uint val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1290,8 +1290,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask, bitmask; + uint val, mux; + uint mask, bitmask; int ret = 0; for (bitmask = 1; bitmask < e->max; bitmask <<= 1)

3 6

[alsa-devel] Roland/Edirol M-16DX
by Lasse Kärkkäinen 24 Jul '08

24 Jul '08

Repost, as I got no responses to the original. This device doesn't seem to be supported yet. Does Roland make the specifications available, etc? The device is not compatible with USB Audio, but rather uses Vendor Specific Class. lsusb -vvv is attached below and if you need more information, I can debug the device on Linux and on Vista. Bus 007 Device 025: ID 0582:00c4 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x00c4 bcdDevice 0.00 iManufacturer 1 EDIROL iProduct 2 M-16DX iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 12 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x01f8 1x 504 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1

2 1

[alsa-devel] Setting format to SND_PCM_FORMAT_MU_LAW does not let me apply hardware parameters
by Mitul Sen (misen) 24 Jul '08

24 Jul '08

I am a newbie to ALSA and any help will be much appreciated. I have an application which sets up the signaling between an IP phone and my desktop and then sets up the audio path between the two. On my desktop application, I am able to receive RTP packets from IP phone. I use an RTP stack to parse the data and after going through the RTP stack, I try to play back the audio using ALSA. When I use the ALSA code to play back (in real time) using sound card of my device, there is only noise, I cannot hear the audio that I speak into the IP phone. However, if I take the raw data coming from the RTP stack and write it to a file, I can play back the file successfully. Since my data from IP phone is G.711 encoded, I have set the sampling rate within ALSA to 8000. Also I am using one source (mono) and non-interleaved data option for preparing ALSA for playback. When I set the format to SND_PCM_FORMAT_MU_LAW, at runtime it lets me set that format ie. snd_pcm_hw_params_set_format (for SND_PCM_FORMAT_MU_LAW) is successful. However I get a runtime error while applying the hardware parameters using snd_pcm_hw_params(..) if the format set earlier is SND_PCM_FORMAT_MU_LAW. Using any other format like SND_PCM_FORMAT_U8 or SND_PCM_FORMAT_S8, lets me apply the hardware parameters but it gives the problem of the noise (no "audible" voice) that I described earlier. This is what my ALSA code looks like ***** beginning of code snippet ***************** int err; snd_pcm_hw_params_t *hw_params; // Try to open the default device err = snd_pcm_open( &_soundDevice, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0 ); // Check for error on open. if( err < 0 ) { printf("Init: cannot open audio device\n"); return -1; } // Allocate the hardware parameter structure. if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { printf("Init: cannot allocate hardware parameter structure\n"); return -1; } if ((err = snd_pcm_hw_params_any (_soundDevice, hw_params)) < 0) { printf("Init: cannot initialize hardware parameter structure\n"); return -1; } // Set access to RW interleaved. if ((err = snd_pcm_hw_params_set_access (_soundDevice, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED)) < 0) { printf("Init: cannot set access type\n"); return -1; } if ((err = snd_pcm_hw_params_set_format (_soundDevice, hw_params, SND_PCM_FORMAT_MU_LAW)) < 0) { return -1; } // Set channels to mono (1) if ((err = snd_pcm_hw_params_set_channels (_soundDevice, hw_params, 1)) < 0) { return -1; } // Set sample rate. unsigned int actualRate = 8000; if ((err = snd_pcm_hw_params_set_rate_near (_soundDevice, hw_params, &actualRate, 0)) < 0) { return -1; } // Apply the hardware parameters that we've set. if ((err = snd_pcm_hw_params (_soundDevice, hw_params)) < 0) { printf("Audio device parameters have not been set successfully.(%s)(%s)\n", strerror (err), snd_strerror (err)); return -1; } // Free the hardware parameters now that we're done with them. snd_pcm_hw_params_free (hw_params); /* Prepare interface for use. if ((err = snd_pcm_prepare (_soundDevice)) < 0) { printf("Audio device not prepared for use\n"); return -1; } // If the frames are received... /* find out how much space is available for playback data if ((frames_to_deliver = snd_pcm_avail_update (_soundDevice)) < 0) { printf("Error returned by snd_pcm_avail_update\n"); } frames_to_deliver = frames_to_deliver > 4096 ? 4096 : frames_to_deliver; printf("frames to deliver is %d\n",(int)frames_to_deliver); */ frames_to_deliver = 172; bufs[1] = (void*)buffer; if ((err = snd_pcm_writen( _soundDevice, bufs, frames_to_deliver)) < 0) { printf ("write failed (%s)\n", snd_strerror (err)); } ****************** end of code snippet ********************************** The error I get is as follows:- Audio device parameters have not been set successfully.(Unknown error 4294967274)(Invalid argument) Any pointers on what I could be doing wrong? Thanks and regards, Mitul

3 14

[alsa-devel] soc_enum has issues
by Jon Smirl 24 Jul '08

24 Jul '08

It is treating mask like a max: dapm_connect_mux... struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int i; for (i = 0; i < e->mask; i++) { if (!(strcmp(control_name, e->texts[i]))) { This looks like an error. I don't think there is a requirement that the number of elements in an enum be a power of two. -- Jon Smirl jonsmirl(a)gmail.com

1 1

[alsa-devel] dapm_reg_event, etc
by Jon Smirl 24 Jul '08

24 Jul '08

I modified Kconfig to build all of the codecs. That exposed a couple more errors. tlv320aic3x won't build . dapm_reg_event() is missing from soc_dapm.h: /* generic register modifier widget */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ { .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \ .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); and there is another error in my patch: diff --git a/include/sound/soc.h b/include/sound/soc.h index 41e73fc..cc64b04 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -67,7 +67,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = shift_left, .rshift = right_shift, \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ .max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ And the AC97 codec won't link, not sure why. sound/built-in.o: In function `ac97_read': ac97.c:(.text+0x2513a): undefined reference to `soc_ac97_ops' ac97.c:(.text+0x25146): undefined reference to `soc_ac97_ops' sound/built-in.o: In function `ac97_write': ac97.c:(.text+0x2516e): undefined reference to `soc_ac97_ops' ac97.c:(.text+0x2517e): undefined reference to `soc_ac97_ops' sound/built-in.o: In function `ac97_soc_probe': ac97.c:(.text+0x252f6): undefined reference to `soc_ac97_ops' sound/built-in.o:ac97.c:(.text+0x25302): more undefined references to `soc_ac97_ops' follow make: *** [.tmp_vmlinux1] Error 1 jonsmirl@terra:~/fs$ make I'll make a new version and put these fixes into it. I'll also make a patch for a Kconfig option to force all of the codecs to build. -- Jon Smirl jonsmirl(a)gmail.com

2 1

[alsa-devel] [PATCH] V3, Convert bitfields in ASOC into full int width
by Jon Smirl 24 Jul '08

24 Jul '08

Convert bitfields in ASOC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Missing header for dapm_reg_event added. All codecs can build. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- include/sound/soc-dapm.h | 2 + include/sound/soc.h | 50 +++++++++++++++++---------- sound/soc/soc-core.c | 86 ++++++++++++++++++++++++++++------------------ sound/soc/soc-dapm.c | 52 +++++++++++++++++----------- 4 files changed, 116 insertions(+), 74 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3030fdc..0b0eff2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -226,6 +226,8 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, const struct snd_soc_dapm_route *route, int num); /* dapm events */ +int dapm_reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, diff --git a/include/sound/soc.h b/include/sound/soc.h index 1890d87..cc64b04 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -26,10 +26,12 @@ /* * Convenience kcontrol builders */ -#define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\ - ((shift) << 12) | ((max) << 16) | ((invert) << 24)) -#define SOC_SINGLE_VALUE_EXT(reg, max, invert) ((reg) | ((max) << 16) |\ - ((invert) << 31)) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .max = xmax, .invert = xinvert}) +#define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -43,45 +45,49 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } -#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ +#define SOC_DOUBLE(xname, xreg, shift_left, shift_right, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, max, invert) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ .put = snd_soc_put_volsw_s8, \ - .private_value = (reg) | (((signed char)max) << 16) | \ - (((signed char)min) << 24) } + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .min = xmin, .max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -516,6 +522,12 @@ struct snd_soc_pcm_runtime { struct snd_soc_device *socdev; }; +/* mixer control */ +struct soc_mixer_control { + int min, max; + uint reg, rreg, shift, rshift, invert; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83f1190..7c59665 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1434,9 +1434,11 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 16) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + uint shift = mc->min; + uint rshift = mc->rshift; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1462,13 +1464,15 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1499,13 +1503,15 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; unsigned short val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); @@ -1537,7 +1543,9 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw); int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 12) & 0xff; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1563,13 +1571,15 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int max = (kcontrol->private_value >> 12) & 0xff; - int mask = (1<<fls(max))-1; - int invert = (kcontrol->private_value >> 20) & 0x01; + uint reg = mc->reg; + uint reg2 = mc->rreg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1<<fls(max))-1; + uint invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1598,13 +1608,15 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int max = (kcontrol->private_value >> 12) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 20) & 0x01; + uint reg = mc->reg; + uint reg2 = mc->rreg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; int err; unsigned short val, val2, val_mask; @@ -1641,8 +1653,10 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (signed char)((kcontrol->private_value >> 16) & 0xff); - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -1664,9 +1678,11 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + uint reg = mc->reg; + int min = mc->min; int val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = @@ -1689,9 +1705,11 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + uint reg = mc->reg; + int min = mc->min; unsigned short val; val = (ucontrol->value.integer.value[0]+min) & 0xff; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..b956771 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -104,10 +104,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: { int val; - int reg = w->kcontrols[i].private_value & 0xff; - int shift = (w->kcontrols[i].private_value >> 8) & 0x0f; - int mask = (w->kcontrols[i].private_value >> 16) & 0xff; - int invert = (w->kcontrols[i].private_value >> 24) & 0x01; + struct soc_mixer_control *mc = (struct soc_mixer_control *) + w->kcontrols[i].private_value; + uint reg = mc->reg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; val = snd_soc_read(w->codec, reg); val = (val >> shift) & mask; @@ -247,16 +250,19 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) return 0; if (widget->num_kcontrols && k) { - int reg = k->private_value & 0xff; - int shift = (k->private_value >> 8) & 0x0f; - int mask = (k->private_value >> 16) & 0xff; - int invert = (k->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)k->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; if (power) { int i; /* power up has happended, increase volume to last level */ if (invert) { - for (i = mask; i > widget->saved_value; i--) + for (i = max; i > widget->saved_value; i--) snd_soc_update_bits(widget->codec, reg, mask, i); } else { for (i = 0; i < widget->saved_value; i++) @@ -1139,12 +1145,14 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; - int mask = (1 << fls(max)) - 1; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint invert = mc->invert; + uint mask = (1 << fls(max)) - 1; /* return the saved value if we are powered down */ if (widget->id == snd_soc_dapm_pga && !widget->power) { @@ -1182,12 +1190,14 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; unsigned short val, val2, val_mask; int ret;

2 2

[alsa-devel] [PATCH] Add a Kconfig option to force all ASOC codecs to build
by Jon Smirl 23 Jul '08

23 Jul '08

--- sound/soc/Kconfig | 7 +++++++ sound/soc/codecs/Kconfig | 15 ++++++++++++++- 2 files changed, 21 insertions(+), 1 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index f743530..ec5d036 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -35,5 +35,12 @@ source "sound/soc/omap/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" +config SND_SOC_ALL_CODECS + bool "Force all codecs to build" + default N + ---help--- + + Select this to force all codecs to build to assist with testing. + endif # SND_SOC diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..6d82c6d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,37 +1,48 @@ config SND_SOC_AC97_CODEC tristate + default y if SND_SOC_ALL_CODECS select SND_AC97_CODEC config SND_SOC_AK4535 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_UDA1380 - tristate + tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM8510 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM8731 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM8750 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM8753 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM8990 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM9712 tristate + default y if SND_SOC_ALL_CODECS config SND_SOC_WM9713 tristate + default y if SND_SOC_ALL_CODECS # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate + default y if SND_SOC_ALL_CODECS # Cirrus Logic CS4270 Codec Hardware Mute Support # Select if you have external muting circuitry attached to your CS4270. @@ -50,3 +61,5 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C + default y if SND_SOC_ALL_CODECS + \ No newline at end of file

2 1

[alsa-devel] [PATCH] V2, Convert bitfields in ASOC into full int width
by Jon Smirl 23 Jul '08

23 Jul '08

Convert bitfields in ASOC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- include/sound/soc.h | 50 ++++++++++++++++++----------- sound/soc/soc-core.c | 86 ++++++++++++++++++++++++++++++-------------------- sound/soc/soc-dapm.c | 52 ++++++++++++++++++------------ 3 files changed, 114 insertions(+), 74 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1890d87..d2a6531 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -26,10 +26,12 @@ /* * Convenience kcontrol builders */ -#define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\ - ((shift) << 12) | ((max) << 16) | ((invert) << 24)) -#define SOC_SINGLE_VALUE_EXT(reg, max, invert) ((reg) | ((max) << 16) |\ - ((invert) << 31)) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .max = xmax, .invert = xinvert}) +#define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -43,45 +45,49 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } -#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ +#define SOC_DOUBLE(xname, xreg, shift_left, shift_right, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, max, invert) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = right_shift, + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ .put = snd_soc_put_volsw_s8, \ - .private_value = (reg) | (((signed char)max) << 16) | \ - (((signed char)min) << 24) } + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .min = xmin, .max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -516,6 +522,12 @@ struct snd_soc_pcm_runtime { struct snd_soc_device *socdev; }; +/* mixer control */ +struct soc_mixer_control { + int min, max; + uint reg, rreg, shift, rshift, invert; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83f1190..7c59665 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1434,9 +1434,11 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 16) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + uint shift = mc->min; + uint rshift = mc->rshift; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1462,13 +1464,15 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1499,13 +1503,15 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; unsigned short val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); @@ -1537,7 +1543,9 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw); int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 12) & 0xff; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1563,13 +1571,15 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int max = (kcontrol->private_value >> 12) & 0xff; - int mask = (1<<fls(max))-1; - int invert = (kcontrol->private_value >> 20) & 0x01; + uint reg = mc->reg; + uint reg2 = mc->rreg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1<<fls(max))-1; + uint invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1598,13 +1608,15 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int max = (kcontrol->private_value >> 12) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 20) & 0x01; + uint reg = mc->reg; + uint reg2 = mc->rreg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; int err; unsigned short val, val2, val_mask; @@ -1641,8 +1653,10 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (signed char)((kcontrol->private_value >> 16) & 0xff); - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -1664,9 +1678,11 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + uint reg = mc->reg; + int min = mc->min; int val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = @@ -1689,9 +1705,11 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + uint reg = mc->reg; + int min = mc->min; unsigned short val; val = (ucontrol->value.integer.value[0]+min) & 0xff; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..b956771 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -104,10 +104,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: { int val; - int reg = w->kcontrols[i].private_value & 0xff; - int shift = (w->kcontrols[i].private_value >> 8) & 0x0f; - int mask = (w->kcontrols[i].private_value >> 16) & 0xff; - int invert = (w->kcontrols[i].private_value >> 24) & 0x01; + struct soc_mixer_control *mc = (struct soc_mixer_control *) + w->kcontrols[i].private_value; + uint reg = mc->reg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; val = snd_soc_read(w->codec, reg); val = (val >> shift) & mask; @@ -247,16 +250,19 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) return 0; if (widget->num_kcontrols && k) { - int reg = k->private_value & 0xff; - int shift = (k->private_value >> 8) & 0x0f; - int mask = (k->private_value >> 16) & 0xff; - int invert = (k->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)k->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; if (power) { int i; /* power up has happended, increase volume to last level */ if (invert) { - for (i = mask; i > widget->saved_value; i--) + for (i = max; i > widget->saved_value; i--) snd_soc_update_bits(widget->codec, reg, mask, i); } else { for (i = 0; i < widget->saved_value; i++) @@ -1139,12 +1145,14 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; - int mask = (1 << fls(max)) - 1; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint invert = mc->invert; + uint mask = (1 << fls(max)) - 1; /* return the saved value if we are powered down */ if (widget->id == snd_soc_dapm_pga && !widget->power) { @@ -1182,12 +1190,14 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; unsigned short val, val2, val_mask; int ret;

2 3

[alsa-devel] [PATCH] Add automatic model setting for the Acer Aspire 5920G laptop
by Travis Place 23 Jul '08

23 Jul '08

Changelog: Make the Acer Aspire 5920G (1025:0121) select ALC883_ACER_ASPIRE by default Patch attached. Signed-off-by: Travis Place <wishie(a)wishie.net>

2 1

[alsa-devel] [PATCH] Convert bitfields in ASOC into full int width
by Jon Smirl 23 Jul '08

23 Jul '08

Convert bitfields in ASOC into full int width. This needs testing, I don't have sample controls for all of the different configurations. This is a simple mechanical conversion, behavior should not be changed. I did find two places in the DAPM code where max was being used as a mask. I fixed them to convert max to a mask, was this right? Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- include/sound/soc.h | 46 ++++++++++++++++++------------ sound/soc/soc-core.c | 77 ++++++++++++++++++++++++++++---------------------- sound/soc/soc-dapm.c | 46 +++++++++++++++++------------- 3 files changed, 96 insertions(+), 73 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1890d87..8d16394 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -26,10 +26,12 @@ /* * Convenience kcontrol builders */ -#define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\ - ((shift) << 12) | ((max) << 16) | ((invert) << 24)) -#define SOC_SINGLE_VALUE_EXT(reg, max, invert) ((reg) | ((max) << 16) |\ - ((invert) << 31)) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ + (unsigned long)&(struct mixer_control) \ + {.reg=xreg, .shift=xshift, .max=xmax, .invert=xinvert} +#define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ + (unsigned long)&(struct mixer_control) \ + {.reg=xreg, .max=xmax, .invert=xinvert} #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -43,45 +45,45 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } -#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ +#define SOC_DOUBLE(xname, xreg, shift_left, shift_right, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, max, invert) \ + .private_value = (unsigned long)&(struct mixer_control) \ + {.reg=xreg, .shift=shift_left, .rshift=shift_right, .max=xmax, .invert=xinvert} +#define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ + .private_value = (unsigned long)&(struct mixer_control) \ + {.reg=reg_left, .rreg=reg_right, .shift=xshift, .max=xmax, .invert=xinvert} +#define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ - .private_value = (reg) | ((shift_left) << 8) | \ - ((shift_right) << 12) | ((max) << 16) | ((invert) << 24) } -#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ + .private_value = (unsigned long)&(struct mixer_control) \ + {.reg=xreg, .shift=shift_left, .rshift=right_shift, .max=xmax, .invert=xinvert} +#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ + .private_value = (unsigned long)&(struct mixer_control) \ + {.reg=reg_left, .rreg=reg_right, .shift=xshift, .max=xmax, .invert=xinvert} +#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ .put = snd_soc_put_volsw_s8, \ - .private_value = (reg) | (((signed char)max) << 16) | \ - (((signed char)min) << 24) } + .private_value = (unsigned long)&(struct mixer_control) \ + {.reg=xreg, .min=xmin, .max=xmax} #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -516,6 +518,12 @@ struct snd_soc_pcm_runtime { struct snd_soc_device *socdev; }; +/* mixer control */ +struct mixer_control { + int min, max; + uint reg, rreg, shift, rshift, invert; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83f1190..3c36b4d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1434,9 +1434,10 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 16) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; + int max = mc->max; + uint shift = mc->min; + uint rshift = mc->rshift; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1462,13 +1463,14 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1499,13 +1501,14 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; unsigned short val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); @@ -1537,7 +1540,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw); int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 12) & 0xff; + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; + int max = mc->max; if (max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1563,13 +1567,14 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int max = (kcontrol->private_value >> 12) & 0xff; - int mask = (1<<fls(max))-1; - int invert = (kcontrol->private_value >> 20) & 0x01; + uint reg = mc->reg; + uint reg2 = mc->rreg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1<<fls(max))-1; + uint invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -1598,13 +1603,14 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int max = (kcontrol->private_value >> 12) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 20) & 0x01; + uint reg = mc->reg; + uint reg2 = mc->rreg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; int err; unsigned short val, val2, val_mask; @@ -1641,8 +1647,9 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (signed char)((kcontrol->private_value >> 16) & 0xff); - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -1664,9 +1671,10 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + uint reg = mc->reg; + int min = mc->min; int val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = @@ -1689,9 +1697,10 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + uint reg = mc->reg; + int min = mc->min; unsigned short val; val = (ucontrol->value.integer.value[0]+min) & 0xff; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..490a486 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -104,10 +104,12 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: { int val; - int reg = w->kcontrols[i].private_value & 0xff; - int shift = (w->kcontrols[i].private_value >> 8) & 0x0f; - int mask = (w->kcontrols[i].private_value >> 16) & 0xff; - int invert = (w->kcontrols[i].private_value >> 24) & 0x01; + struct mixer_control *mc = (struct mixer_control *)w->kcontrols[i].private_value; + uint reg = mc->reg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; val = snd_soc_read(w->codec, reg); val = (val >> shift) & mask; @@ -247,10 +249,12 @@ static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) return 0; if (widget->num_kcontrols && k) { - int reg = k->private_value & 0xff; - int shift = (k->private_value >> 8) & 0x0f; - int mask = (k->private_value >> 16) & 0xff; - int invert = (k->private_value >> 24) & 0x01; + struct mixer_control *mc = (struct mixer_control *)k->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; if (power) { int i; @@ -1139,12 +1143,13 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; - int mask = (1 << fls(max)) - 1; + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint invert = mc->invert; + uint mask = (1 << fls(max)) - 1; /* return the saved value if we are powered down */ if (widget->id == snd_soc_dapm_pga && !widget->power) { @@ -1182,12 +1187,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int rshift = (kcontrol->private_value >> 12) & 0x0f; - int max = (kcontrol->private_value >> 16) & 0xff; - int mask = (1 << fls(max)) - 1; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct mixer_control *mc = (struct mixer_control *)kcontrol->private_value; + uint reg = mc->reg; + uint shift = mc->shift; + uint rshift = mc->rshift; + int max = mc->max; + uint mask = (1 << fls(max)) - 1; + uint invert = mc->invert; unsigned short val, val2, val_mask; int ret;

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[alsa-devel] Linux Plumbers Conference CFP Extended!
by Lennart Poettering 23 Jul '08

23 Jul '08

Heya! The Call For Papers for the Linux Plumbers Conference has been extended until July 31st! Anyone doing (audio) infrastructure work on Linux? If so, please make sure to join us at the Linux Plumbers Conference in Portland/Oregon in September. I will be doing a track about "Audio" there and would like to make sure that everyone who does audio infrastructure work will submit a paper, or at least attend, or maybe just knows about this conference! It's a conference about the kernel-userspace interfacing, about the lower levels in our software stack. For further details, checkout the conference website: http://www.linuxplumbersconf.org/ (Sorry for this spam!) Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] cs4232: fix crash during chip PNP detection
by Krzysztof Helt 23 Jul '08

23 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The acard->wss pointer is uninitialized in this function which leads to crash during chip PNP detection. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This bug was found in the 2.6.26-git9 kernel. This is the second version of the patch previously called: "cs4236: add missing pnp_request_card_device()" The crash log (if needed): BUG: unable to handle kernel NULL pointer dereference at 00000158 IP: [<c02ba3f8>] pnp_activate_dev+0x5/0x37 *pde = 00000000 Oops: 0000 [#1] Modules linked in: snd_cs4232(+) snd_opl3_lib snd_hwdep snd_cs4231_lib snd_pcm snd_timer snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device parport_pc parport Pid: 668, comm: modprobe Not tainted (2.6.26-git9 #1) EIP: 0060:[<c02ba3f8>] EFLAGS: 00010246 CPU: 0 EIP is at pnp_activate_dev+0x5/0x37 EAX: 00000000 EBX: 00000000 ECX: c1c46800 EDX: 00000000 ESI: 00000000 EDI: c1c46800 EBP: c02d6635 ESP: c1db3ef0 DS: 007b ES: 007b FS: 0000 GS: 0033 SS: 0068 Process modprobe (pid: 668, ti=c1db2000 task=c064d3e0 task.ti=c1db2000) Stack: 00000000 c2a710ed 00000000 c1c46800 c2a712e7 c1e35600 c2a71702 c1c46800 c1c46800 c2a74120 c02b93ca c1c46800 00000000 c2a7413c c02d650d c2a7413c c1c46800 00000000 c02d65ca c1c46800 c1c468a8 c2a7413c c02d666c c1db3f5c Call Trace: [<c2a710ed>] snd_cs423x_pnp_init_wss+0xd/0x10c [snd_cs4232] [<c2a712e7>] snd_card_cs4232_pnp+0xb/0x25 [snd_cs4232] [<c2a71702>] snd_cs4232_pnpbios_detect+0x7b/0xcf [snd_cs4232] [<c02b93ca>] pnp_device_probe+0x5d/0x7a [<c02d650d>] really_probe+0x70/0xea [<c02d65ca>] driver_probe_device+0x34/0x3c [<c02d666c>] __driver_attach+0x37/0x55 [<c02d5aa2>] bus_for_each_dev+0x36/0x5a [<c02998d2>] kobject_init_and_add+0x23/0x25 [<c02d669b>] driver_attach+0x11/0x13 [<c02d6635>] __driver_attach+0x0/0x55 [<c02d5f9f>] bus_add_driver+0x8a/0x132 [<c02d69c2>] driver_register+0x68/0x88 [<c2a43027>] alsa_card_cs423x_init+0x27/0x6e [snd_cs4232] [<c022eb96>] sys_init_module+0x84/0x173 [<c02028c2>] syscall_call+0x7/0xb ======================= Code: e8 10 b3 f5 ff b8 fb ff ff ff eb 15 50 89 d8 e8 eb 9e 01 00 50 68 21 c3 3d c0 e8 f6 b2 f5 ff 31 c0 83 c4 0c 5b c3 53 31 d2 89 c3 <83> b8 58 01 00 00 00 75 25 e8 a3 fe ff ff ba f0 ff ff ff 85 c0 EIP: [<c02ba3f8>] pnp_activate_dev+0x5/0x37 SS:ESP 0068:c1db3ef0 ---[ end trace ff15a9f65b38124f ]--- --- linux-2.6.26/sound/isa/cs423x/cs4236.c~ 2008-07-23 07:40:16.129637645 +0200 +++ linux-2.6.26/sound/isa/cs423x/cs4236.c 2008-07-23 07:40:32.905641806 +0200 @@ -325,6 +325,7 @@ static int __devinit snd_cs423x_pnp_init static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, struct pnp_dev *pdev) { + acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; cport[dev] = -1; ---------------------------------------------------------------------- Galeria absurdow. zobacz >>> http://link.interia.pl/f1e5e

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[alsa-devel] [PATCH] cs4236: add missing pnp_request_card_device()
by Krzysztof Helt 23 Jul '08

23 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is a missing pnp_request_card_device() call for cs4232 card. The acard->wss pointer is left uninitialized which leads to crash during detection. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This bug was found in the 2.6.26-git9 kernel. Please compare the snd_card_cs4232_pnp() and the snd_card_cs423x_pnpc(). --- linux-2.6.26/sound/isa/cs423x/cs4236.c~ 2008-07-13 23:51:29.000000000 +0200 +++ linux-2.6.26/sound/isa/cs423x/cs4236.c 2008-07-22 23:41:56.010942922 +0200 @@ -325,6 +325,9 @@ static int __devinit snd_cs423x_pnp_init static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, struct pnp_dev *pdev) { + acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL); + if (acard->wss == NULL) + return -EBUSY; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; cport[dev] = -1; ---------------------------------------------------------------------- Czy Twoj mezczyzna gotuje dla Ciebie? Wygraj nagrody >>> http://link.interia.pl/f1e86

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[alsa-devel] [PATCH v3 0/3] ALSA SoC: MPC5200 audio driver
by Grant Likely 23 Jul '08

23 Jul '08

Hi all, Here is the latest series for adding MPC5200 I2S and TI AIC26 codec support to ALSA SoC. I believe all the comments are addressed and I hope that this series is now ready to be merged. I would really like to see these ones land in 2.6.27. Thanks, g. include/sound/soc-of-simple.h | 21 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic26.c | 519 +++++++++++++++++++++++ sound/soc/codecs/tlv320aic26.h | 93 ++++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/mpc5200_psc_i2s.c | 884 +++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/soc-of-simple.c | 171 ++++++++ 9 files changed, 1709 insertions(+), 0 deletions(-) Cheers, g. On Tue, Jul 22, 2008 at 12:53 AM, Grant Likely <grant.likely(a)secretlab.ca> wrote: > From: Grant Likely <grant.likely(a)secretlab.ca> > > Simple utility layer for creating ASoC machine instances based on data > in the OpenFirmware device tree. OF aware platform drivers and codec > drivers register themselves with this framework and the framework > automatically instantiates a machine driver. At the moment, the driver > is not very capable and it is expected to be extended as more features > are needed for specifying the configuration in the device tree. > > This is most likely temporary glue code to work around limitations in > the ASoC v1 framework. When v2 is merged, most of this driver will > need to be reworked. > > Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> > --- > > include/sound/soc-of-simple.h | 21 +++++ > sound/soc/fsl/Kconfig | 3 + > sound/soc/fsl/Makefile | 3 + > sound/soc/fsl/soc-of-simple.c | 171 +++++++++++++++++++++++++++++++++++++++++ > 4 files changed, 198 insertions(+), 0 deletions(-) > > diff --git a/include/sound/soc-of-simple.h b/include/sound/soc-of-simple.h > new file mode 100644 > index 0000000..696fc51 > --- /dev/null > +++ b/include/sound/soc-of-simple.h > @@ -0,0 +1,21 @@ > +/* > + * OF helpers for ALSA SoC > + * > + * Copyright (C) 2008, Secret Lab Technologies Ltd. > + */ > + > +#ifndef _INCLUDE_SOC_OF_H_ > +#define _INCLUDE_SOC_OF_H_ > + > +#include <linux/of.h> > +#include <sound/soc.h> > + > +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, > + void *codec_data, struct snd_soc_dai *dai, > + struct device_node *node); > + > +int of_snd_soc_register_platform(struct snd_soc_platform *platform, > + struct device_node *node, > + struct snd_soc_dai *cpu_dai); > + > +#endif /* _INCLUDE_SOC_OF_H_ */ > diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig > index 3368ace..398f002 100644 > --- a/sound/soc/fsl/Kconfig > +++ b/sound/soc/fsl/Kconfig > @@ -1,3 +1,6 @@ > +config SND_SOC_OF_SIMPLE > + tristate > + > config SND_SOC_MPC8610 > bool "ALSA SoC support for the MPC8610 SOC" > depends on MPC8610_HPCD > diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile > index 62f680a..aa2100b 100644 > --- a/sound/soc/fsl/Makefile > +++ b/sound/soc/fsl/Makefile > @@ -1,3 +1,6 @@ > +# Simple machine driver that extracts configuration from the OF device tree > +obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o > + > # MPC8610 HPCD Machine Support > obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o > > diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c > new file mode 100644 > index 0000000..0382fda > --- /dev/null > +++ b/sound/soc/fsl/soc-of-simple.c > @@ -0,0 +1,171 @@ > +/* > + * OF helpers for ALSA SoC Layer > + * > + * Copyright (C) 2008, Secret Lab Technologies Ltd. > + */ > + > +#include <linux/module.h> > +#include <linux/moduleparam.h> > +#include <linux/init.h> > +#include <linux/delay.h> > +#include <linux/pm.h> > +#include <linux/bitops.h> > +#include <linux/platform_device.h> > +#include <linux/of.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > +#include <sound/soc-of-simple.h> > +#include <sound/initval.h> > + > +MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); > +MODULE_LICENSE("GPL"); > +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings"); > + > +static DEFINE_MUTEX(of_snd_soc_mutex); > +static LIST_HEAD(of_snd_soc_device_list); > +static int of_snd_soc_next_index; > + > +struct of_snd_soc_device { > + int id; > + struct list_head list; > + struct snd_soc_device device; > + struct snd_soc_machine machine; > + struct snd_soc_dai_link dai_link; > + struct platform_device *pdev; > + struct device_node *platform_node; > + struct device_node *codec_node; > +}; > + > +static struct snd_soc_ops of_snd_soc_ops = { > +}; > + > +static struct of_snd_soc_device * > +of_snd_soc_get_device(struct device_node *codec_node) > +{ > + struct of_snd_soc_device *of_soc; > + > + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { > + if (of_soc->codec_node == codec_node) > + return of_soc; > + } > + > + of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL); > + if (!of_soc) > + return NULL; > + > + /* Initialize the structure and add it to the global list */ > + of_soc->codec_node = codec_node; > + of_soc->id = of_snd_soc_next_index++; > + of_soc->machine.dai_link = &of_soc->dai_link; > + of_soc->machine.num_links = 1; > + of_soc->device.machine = &of_soc->machine; > + of_soc->dai_link.ops = &of_snd_soc_ops; > + list_add(&of_soc->list, &of_snd_soc_device_list); > + > + return of_soc; > +} > + > +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) > +{ > + struct platform_device *pdev; > + int rc; > + > + /* Only register the device if both the codec and platform have > + * been registered */ > + if ((!of_soc->device.codec_data) || (!of_soc->platform_node)) > + return; > + > + pr_info("platform<-->codec match achieved; registering machine\n"); > + > + pdev = platform_device_alloc("soc-audio", of_soc->id); > + if (!pdev) { > + pr_err("of_soc: platform_device_alloc() failed\n"); > + return; > + } > + > + pdev->dev.platform_data = of_soc; > + platform_set_drvdata(pdev, &of_soc->device); > + of_soc->device.dev = &pdev->dev; > + > + /* The ASoC device is complete; register it */ > + rc = platform_device_add(pdev); > + if (rc) { > + pr_err("of_soc: platform_device_add() failed\n"); > + return; > + } > + > +} > + > +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, > + void *codec_data, struct snd_soc_dai *dai, > + struct device_node *node) > +{ > + struct of_snd_soc_device *of_soc; > + int rc = 0; > + > + pr_info("registering ASoC codec driver: %s\n", node->full_name); > + > + mutex_lock(&of_snd_soc_mutex); > + of_soc = of_snd_soc_get_device(node); > + if (!of_soc) { > + rc = -ENOMEM; > + goto out; > + } > + > + /* Store the codec data */ > + of_soc->device.codec_data = codec_data; > + of_soc->device.codec_dev = codec_dev; > + of_soc->dai_link.name = (char *)node->name; > + of_soc->dai_link.stream_name = (char *)node->name; > + of_soc->dai_link.codec_dai = dai; > + > + /* Now try to register the SoC device */ > + of_snd_soc_register_device(of_soc); > + > + out: > + mutex_unlock(&of_snd_soc_mutex); > + return rc; > +} > +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec); > + > +int of_snd_soc_register_platform(struct snd_soc_platform *platform, > + struct device_node *node, > + struct snd_soc_dai *cpu_dai) > +{ > + struct of_snd_soc_device *of_soc; > + struct device_node *codec_node; > + const phandle *handle; > + int len, rc = 0; > + > + pr_info("registering ASoC platform driver: %s\n", node->full_name); > + > + handle = of_get_property(node, "codec-handle", &len); > + if (!handle || len < sizeof(handle)) > + return -ENODEV; > + codec_node = of_find_node_by_phandle(*handle); > + if (!codec_node) > + return -ENODEV; > + pr_info("looking for codec: %s\n", codec_node->full_name); > + > + mutex_lock(&of_snd_soc_mutex); > + of_soc = of_snd_soc_get_device(codec_node); > + if (!of_soc) { > + rc = -ENOMEM; > + goto out; > + } > + > + of_soc->platform_node = node; > + of_soc->dai_link.cpu_dai = cpu_dai; > + of_soc->device.platform = platform; > + of_soc->machine.name = of_soc->dai_link.cpu_dai->name; > + > + /* Now try to register the SoC device */ > + of_snd_soc_register_device(of_soc); > + > + out: > + mutex_unlock(&of_snd_soc_mutex); > + return rc; > +} > +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform); > > -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

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[alsa-devel] [PATCH v3 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers
by Grant Likely 23 Jul '08

23 Jul '08

From: Grant Likely <grant.likely(a)secretlab.ca> Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver. At the moment, the driver is not very capable and it is expected to be extended as more features are needed for specifying the configuration in the device tree. This is most likely temporary glue code to work around limitations in the ASoC v1 framework. When v2 is merged, most of this driver will need to be reworked. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- include/sound/soc-of-simple.h | 21 +++++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/soc-of-simple.c | 171 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 198 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-of-simple.h b/include/sound/soc-of-simple.h new file mode 100644 index 0000000..696fc51 --- /dev/null +++ b/include/sound/soc-of-simple.h @@ -0,0 +1,21 @@ +/* + * OF helpers for ALSA SoC + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#ifndef _INCLUDE_SOC_OF_H_ +#define _INCLUDE_SOC_OF_H_ + +#include <linux/of.h> +#include <sound/soc.h> + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_dai *dai, + struct device_node *node); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_dai *cpu_dai); + +#endif /* _INCLUDE_SOC_OF_H_ */ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3368ace..398f002 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,6 @@ +config SND_SOC_OF_SIMPLE + tristate + config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" depends on MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..aa2100b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,3 +1,6 @@ +# Simple machine driver that extracts configuration from the OF device tree +obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o + # MPC8610 HPCD Machine Support obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c new file mode 100644 index 0000000..0382fda --- /dev/null +++ b/sound/soc/fsl/soc-of-simple.c @@ -0,0 +1,171 @@ +/* + * OF helpers for ALSA SoC Layer + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-of-simple.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings"); + +static DEFINE_MUTEX(of_snd_soc_mutex); +static LIST_HEAD(of_snd_soc_device_list); +static int of_snd_soc_next_index; + +struct of_snd_soc_device { + int id; + struct list_head list; + struct snd_soc_device device; + struct snd_soc_machine machine; + struct snd_soc_dai_link dai_link; + struct platform_device *pdev; + struct device_node *platform_node; + struct device_node *codec_node; +}; + +static struct snd_soc_ops of_snd_soc_ops = { +}; + +static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{ + struct of_snd_soc_device *of_soc; + + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { + if (of_soc->codec_node == codec_node) + return of_soc; + } + + of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL); + if (!of_soc) + return NULL; + + /* Initialize the structure and add it to the global list */ + of_soc->codec_node = codec_node; + of_soc->id = of_snd_soc_next_index++; + of_soc->machine.dai_link = &of_soc->dai_link; + of_soc->machine.num_links = 1; + of_soc->device.machine = &of_soc->machine; + of_soc->dai_link.ops = &of_snd_soc_ops; + list_add(&of_soc->list, &of_snd_soc_device_list); + + return of_soc; +} + +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) +{ + struct platform_device *pdev; + int rc; + + /* Only register the device if both the codec and platform have + * been registered */ + if ((!of_soc->device.codec_data) || (!of_soc->platform_node)) + return; + + pr_info("platform<-->codec match achieved; registering machine\n"); + + pdev = platform_device_alloc("soc-audio", of_soc->id); + if (!pdev) { + pr_err("of_soc: platform_device_alloc() failed\n"); + return; + } + + pdev->dev.platform_data = of_soc; + platform_set_drvdata(pdev, &of_soc->device); + of_soc->device.dev = &pdev->dev; + + /* The ASoC device is complete; register it */ + rc = platform_device_add(pdev); + if (rc) { + pr_err("of_soc: platform_device_add() failed\n"); + return; + } + +} + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_dai *dai, + struct device_node *node) +{ + struct of_snd_soc_device *of_soc; + int rc = 0; + + pr_info("registering ASoC codec driver: %s\n", node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + /* Store the codec data */ + of_soc->device.codec_data = codec_data; + of_soc->device.codec_dev = codec_dev; + of_soc->dai_link.name = (char *)node->name; + of_soc->dai_link.stream_name = (char *)node->name; + of_soc->dai_link.codec_dai = dai; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_dai *cpu_dai) +{ + struct of_snd_soc_device *of_soc; + struct device_node *codec_node; + const phandle *handle; + int len, rc = 0; + + pr_info("registering ASoC platform driver: %s\n", node->full_name); + + handle = of_get_property(node, "codec-handle", &len); + if (!handle || len < sizeof(handle)) + return -ENODEV; + codec_node = of_find_node_by_phandle(*handle); + if (!codec_node) + return -ENODEV; + pr_info("looking for codec: %s\n", codec_node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(codec_node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + of_soc->platform_node = node; + of_soc->dai_link.cpu_dai = cpu_dai; + of_soc->device.platform = platform; + of_soc->machine.name = of_soc->dai_link.cpu_dai->name; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);

3 10

[alsa-devel] RFC: reworking ASOC for large registers
by Jon Smirl 22 Jul '08

22 Jul '08

The current read/write routines: unsigned int read(struct snd_soc_codec *codec, unsigned int reg); int write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); How about making them work on an array of bytes? unsigned int read(struct snd_soc_codec *codec, unsigned int reg, u8 *value); int write(struct snd_soc_codec *codec, unsigned int reg, u8* value); If read is sucessful, it returns the number of bytes read. Reading a non-existent register returns -ENODEV, or you can get other errors like -EIO when the IO fails. Write can return bytes written or a negative error code. Instead of u8 I've been using a union in my internal code. We can come up with better names for the members. typedef struct { union { u16 s16; u8 byte[32]; u16 value[16]; u32 data[8]; }; } register_t; It is easy to change the existing drivers to assign to the var parameter rather than return from the function. After the low level is changed, the ASOC core needs to be updated to allow 16 bit shifts and masks up to 64 bits. The encoding into 32 bit in the private field would need to be broken out to use a structure. .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } I think this can be changed inside the macros via a C99 initializer without effecting existing source. For a transition the snd_soc_write/snd_soc_read macros could see if the driver has the old or new form of read/write and use whichever is defined. -- Jon Smirl jonsmirl(a)gmail.com

2 9

[alsa-devel] [2.6 patch] make snd_ac97_add_vmaster() static
by Adrian Bunk 22 Jul '08

22 Jul '08

This patch makes the needlessly global snd_ac97_add_vmaster() static. Signed-off-by: Adrian Bunk <bunk(a)kernel.org> --- 8ca0ca6ab50a53faf6c7af44b67408e32d11f69a diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 0746e9c..f4fbc79 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3381,8 +3381,8 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, } /* create a virtual master control and add slaves */ -int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, - const unsigned int *tlv, const char **slaves) +static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) { struct snd_kcontrol *kctl; const char **s;

1 0

[alsa-devel] logging stream creation
by Nick Thompson 21 Jul '08

21 Jul '08

Hello, I've just started looking at ALSA, and I have a couple questions that I hope will be pretty easy to answer. I'd like to log stream creation as streams are brought up and shut down. Should I use an I/O plugin for this? Is there code I should start to look at for doing this. Ideally I'd like to get a notification once streaming starts on a PCM and when it stops. If I wanted to manipulate DMIX settings in response to streams being started and stopped, is it best to do this via a plugin (possibly the same one that I'd write for the stream monitoring, or from an external app that has a connection to the monitoring plug). Finally is it possible to change the physical ALSA sink for a running PCM (for example if I have two devices, I start streaming on the first and decide in response to a user even that I need to stream to another sink). Thank you!! Nick

1 0

[alsa-devel] CS4279 and hw_read_t/hw_write_t
by Jon Smirl 21 Jul '08

21 Jul '08

_From soc/codecs/cs4270.c... * * Note that we don't use the hw_write function pointer of snd_soc_codec. * That's because it's too clunky: the hw_write_t prototype does not match * i2c_smbus_write_byte_data(), and it's just another layer of overhead. */ The prototype doesn't match because you are using i2c_smbus_write_byte_data() instead of i2c_master_send()/i2c_master_recv(). -- Jon Smirl jonsmirl(a)gmail.com

2 4

[alsa-devel] [PATCH] ALSA: soc - fsl_ssi.c fix audio capture
by Anton Vorontsov 21 Jul '08

21 Jul '08

Since we're using SSI in synchronous mode, the STCCR register controls both the receive and transmit sections. So, when we're trying to record anything, stccr register does not get initialized, thus the output file filled with the white noise. Fix this by initializing the STCCR for both playback and capture. If TX/RX widths don't match, return that we're busy at the moment. Signed-off-by: Anton Vorontsov <avorontsov(a)ru.mvista.com> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f588545..31f689e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -372,16 +372,24 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct ccsr_ssi __iomem *ssi = ssi_private->ssi; u32 wl; + u32 cur_wl; wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + /* + * MPC8610 spec says: "The STCCR register is dedicated to the transmit + * section, and the SRCCR register is dedicated to the receive section + * except in Synchronous mode, in which the STCCR register controls + * both the receive and transmit sections." + * So, the width for TX and RX should match, otherwise we're busy with + * either TX or RX. + */ + cur_wl = in_be32(&ssi->stccr) & CCSR_SSI_SxCCR_WL_MASK; + if (cur_wl && cur_wl != wl) + return -EBUSY; + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); return 0; @@ -460,6 +468,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, 0); free_irq(ssi_private->irq, ssi_private); } -- 1.5.5.4

4 12

[alsa-devel] GIT: something wrong with sound-2.6/devel?
by Rene Herman 21 Jul '08

21 Jul '08

Hi Takashi. Is there something wrong with your devel branch or is it me? rene@7ixe4:~/src/linux/7ixe4$ git fetch tiwai remote: Counting objects: 1695, done. remote: Compressing objects: 100% (418/418), done. remote: Total 1418 (delta 1182), reused 1222 (delta 1000) Receiving objects: 100% (1418/1418), 275.35 KiB, done. Resolving deltas: 100% (1182/1182), completed with 170 local objects. From git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 ! [rejected] devel -> tiwai/devel (non fast forward) ! [rejected] dma-fix -> tiwai/dma-fix (non fast forward) 3e14b50..9432484 for-linus -> tiwai/for-linus ! [rejected] master -> tiwai/master (non fast forward) 481c534..50515af upstream -> tiwai/upstream rene@7ixe4:~/src/linux/7ixe4$ git checkout -b head v2.6.26 Switched to a new branch "head" rene@7ixe4:~/src/linux/7ixe4$ git merge tiwai/devel Auto-merged sound/pci/ac97/ac97_patch.c Auto-merged sound/pci/emu10k1/emu10k1_main.c Auto-merged sound/pci/hda/patch_analog.c Auto-merged sound/pci/hda/patch_realtek.c CONFLICT (content): Merge conflict in sound/pci/hda/patch_realtek.c Auto-merged sound/pci/hda/patch_sigmatel.c Automatic merge failed; fix conflicts and then commit the result. It's no doubt trivial to fixup, but I believe this should've merged cleanly? (the master branch from where I'm branching off head is an undisturbed linus tree). Rene.

3 7

[alsa-devel] cross compile errors
by SunZhibin 21 Jul '08

21 Jul '08

Hello, I want to cross compile alsa-driver for my arm9 custom board , when I build it, I get the following errors: CC [M] /home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.oIn file included from /home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:25:/home/zb/alsa/alsa-driver-1.0.17/include/adriver.h:828: error: static declaration of 'jiffies_to_msecs' follows non-static declarationinclude/linux/jiffies.h:264: error: previous declaration of 'jiffies_to_msecs' was here/home/zb/alsa/alsa-driver-1.0.17/include/adriver.h:847: error: static declaration of 'msecs_to_jiffies' follows non-static declarationinclude/linux/jiffies.h:266: error: previous declaration of 'msecs_to_jiffies' was hereIn file included from /home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:25:/home/zb/alsa/alsa-driver-1.0.17/include/adriver.h:1552:1: warning: "page_to_pfn" redefinedIn file included from include/asm/memory.h:329, from include/asm/system.h:6, from include/linux/list.h:9, from include/linux/module.h:9, from /home/zb/alsa/alsa-driver-1.0.17/include/adriver.h:26, from /home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:25:include/asm-generic/memory_model.h:79:1: warning: this is the location of the previous definitionIn file included from /home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:26:include/linux/mm.h: In function 'lowmem_page_address':include/linux/mm.h:538: error: implicit declaration of function 'page_to_pfn'/home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c: In function 'snd_compat_vmalloc_to_page':/home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:44: error: implicit declaration of function 'VMALLOC_VMADDR'/home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:50: error: implicit declaration of function 'pte_offset'/home/zb/alsa/alsa-driver-1.0.17/acore/memory_wrapper.c:50: warning: assignment makes pointer from integer without a cast I use the following configure options: export $PREFIX=/opt/alsaexport ARCH=armexport CROSS_COMPILE=/opt/arm-linux/bin/arm-linux- ./configure --prefix=$PREFIX --host=arm-linux --enable-shared --enable-static \ --with-cross=$CROSS_COMPILE \ --with-kernel=/home/zb/embedded/linux-2.6.24.7 \ --with-debug=verbose \ --with-isapnp=no \ --with-cards=s3c2410,soc,s3c24xx-soc,s3c24xx-soc-neo1973-wm8753,s3c24xx-soc-smdk2443-wm9710,s3c24xx-soc-ln2440sbc-alc650 how can is solve this problem ? Thanks _________________________________________________________________ 这里好多好玩的视频，用鼠标点到视频看看，有惊喜！ http://cnweb.search.live.com/video/results.aspx?q=%E5%A5%A5%E8%BF%90%E5%9C%…

1 0

[alsa-devel] ASOC and 12 bit volume control
by Jon Smirl 21 Jul '08

21 Jul '08

How do I define a volume control that is the bottom 12 bits of a 32 bit register? #define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\ ((shift) << 12) | ((max) << 16) | ((invert) << 24)) This looks like it only support an 8 bit max. Is there another way to do it? -- Jon Smirl jonsmirl(a)gmail.com

2 8

[alsa-devel] 1.0.17
by Elimar Riesebieter 21 Jul '08

21 Jul '08

For the first step alsa-lib is upgraded in svn. I am very busy at the moment and will see what I can do nest week. My goal: Upload latest in week 31. Elimar -- Numeric stability is probably not all that important when you're guessing;-)

1 1

[alsa-devel] [PATCH] wss: fix capture formats limitations
by Krzysztof Helt 20 Jul '08

20 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix copy-paste error with Interwave instead of ad1848 chip condition. Limit opti93x cards capture formats to only linear ones. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Thank you Rene for collecting, testing and reviewing my patches. I found one mistake in the series and this patch fixes it. I have also limited capture formats for opti93x chips because the opti93x manual states that it supports mu-law, a-law and ADPCM decompression. The word "decompression" probably suggest that compression (recording) is not possible as its manufacturer could just state the card supported mu-law, a-law and ADPCM. Regards, Krzysztof diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-18 22:47:02.056943647 +0200 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-07-18 22:56:38.200949874 +0200 @@ -1460,14 +1460,16 @@ static int snd_wss_capture_open(struct s runtime->hw = snd_wss_capture; /* hardware limitation of older chipsets */ - if (chip->hardware == WSS_HW_INTERWAVE && chip->dma1 > 3) + if (chip->hardware & WSS_HW_AD1848_MASK) runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM | SNDRV_PCM_FMTBIT_S16_BE); /* hardware limitation of cheap chips */ if (chip->hardware == WSS_HW_CS4235 || - chip->hardware == WSS_HW_CS4239) - runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; + chip->hardware == WSS_HW_CS4239 || + chip->hardware == WSS_HW_OPTI93X) + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE; snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); ---------------------------------------------------------------------- W kosciele tez zdarzaja sie wpadki! Smieszny filmik >>> http://link.interia.pl/f1e61

2 2

[alsa-devel] [0/2] Jack reporting
by Mark Brown 20 Jul '08

20 Jul '08

This is a repost of the the jack reporting API I posted recently. There are no chances since the last submission - as discussed in that thread the existing users and the need to support buttons implemented in jacks pushes me towards use of an input device to represent jacks.

3 11

[alsa-devel] Repost: snd-hda-intel[ALC883] scratching and stuttering sound on MCP78S board
by Hans-Frieder Vogt 20 Jul '08

20 Jul '08

Hi List, This is just an updated repost of my request from a month ago. So far I have not achieved to solve the problem myselves, therefore I would very much appreciate ANY hint as to what (and even more important: where) to look for. Here is a summary of my problem and some related info: I am experiencing severely scratching and stuttering sound on my nVidia MCP78S (GeForce 8200) based Asus M3N78-EMH HDMI mainboard, which has an ALC883 on board, running with an Athlon X2 4850e dual-core CPU. It sounds, as if the sound buffers are not filled in time and are therefore repeated many times (2 to 4 times) per second. The problem seems to be kernel / alsa-driver related, because I experience the problem also when running runlevel 1 with only a minimum of active processes (e.g. without graphics). Strange enough, the sound is OK when I do a larger compilation in parallel (kernel make with make -j 2), but the sound is not OK when I run burnK7 (two instances to sufficiently load both CPU cores) => What is different in CPU loading or scheduling between these two load cases? Also, the sound is OK when I copy large files in parallel. Could it be related, that lspci gives me for all devices a latency of 0, even though the kernel message tell me that it was set to 64? I cannot change the PCI latency to another value via setpci. I am using Message Signal Interrupts for HD Audio, although switching it off also makes no audible difference. My Interrupt distribution is: CPU0 CPU1 0: 39 1 IO-APIC-edge timer 1: 61 27962 IO-APIC-edge i8042 6: 0 5 IO-APIC-edge floppy 7: 1 0 IO-APIC-edge 8: 0 1 IO-APIC-edge rtc0 9: 0 0 IO-APIC-fasteoi acpi 20: 826 748020 IO-APIC-fasteoi nvidia 22: 0 3 IO-APIC-fasteoi ehci_hcd:usb2, ehci_hcd:usb4 23: 437 175388 IO-APIC-fasteoi ohci_hcd:usb1, ohci_hcd:usb3 314: 2891 1330645 PCI-MSI-edge eth1 315: 0 0 PCI-MSI-edge cx23885[0] 316: 258 115865 PCI-MSI-edge HDA Intel 317: 286 118802 PCI-MSI-edge ahci NMI: 0 0 Non-maskable interrupts LOC: 977113 1020994 Local timer interrupts RES: 327021 99771 Rescheduling interrupts CAL: 1408 972 function call interrupts TLB: 11275 4795 TLB shootdowns TRM: 0 0 Thermal event interrupts THR: 0 0 Threshold APIC interrupts SPU: 0 0 Spurious interrupts ERR: 1 I tried several things so far: Debian kernel image 2.6.25-2-amd64 (2.6.25-5 and 2.6.25-7) Ubuntu 8.04 with kernel 2.6.24-16 self-compiled 2.6.25.x (with x in nothing to 5), with and without -rt patches self-compiled 2.6.26-rc5 to 2.6.26 final latest alsa-driver 1.0.17rc2, 1.0.17-rc3 and 1.0.17 alsa-library versions 1.0.16 and 1.0.17 all these versions give the similar distorted output. please find the alsa-info output here: http://pastebin.ca/1077615 (previously posted info: http://pastebin.ca/1052359 ) attached is my latest kernel setup (2.6.26, alsa driver modules now replaced by version 1.0.17). Please tell me your suggestions what the problem may be or how I could debug the problem. Thanks very much. -- -- Hans-Frieder Vogt e-mail: hfvogt <at> gmx .dot. net

1 0

[alsa-devel] [PATCH] wm9712 mono mixer
by Marek Vasut 20 Jul '08

20 Jul '08

Hi, this fixes typo in wm9712 codec which prevents it from registering all audio routes (and thus working correctly). Please consider applying. (Tested and works on palmtx, palmld and palmt5)

3 2

[alsa-devel] softvol and snd_pcm_rewind() is broken
by Lennart Poettering 20 Jul '08

20 Jul '08

Heya! With 1.0.17rc3 snd_pcm_rewind() is broken for softvol as it seems: - Sometimes the sound becomes heavily distorted after such a seek: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008860.html - And snd_pcm_rewind() might return a value that is higher than was passed in, which as far as I understood should never happen: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-April/007308.html These two issues might be caused by the same error. Takashi, Jaroslav, how can I bribe you into fixing this? I'd love to release my new PulseAudio version soon which heavily relies on snd_pcm_rewind(), but unfortunately the most important driver (hda with softvol) makes the most problems with it. :-( Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] No sound with ice_1724 when I restart the system since I upgrade to 2.6.26 (+workaround)
by Amaury De Ganseman 20 Jul '08

20 Jul '08

Hi all, I think it's better to send here ;-) Additional infos: Kernel 2.6.26 from kernel.org /proc/asound/cards: 0 [Juli ]: ICE1724 - ESI Juli@ ESI Juli@ at 0xb400, irq 18 + /var/lib/alsa/asound.state in attach (after stop alsasound, rm asound.state and then restart alsa --> so sound is working) ---------- Forwarded message ---------- From: Amaury De Ganseman <amaury.deganseman(a)gmail.com> Date: Tue, Jul 15, 2008 at 9:50 PM Subject: No sound when I restart the system since I upgrade to 2.6.26 (+workaround) To: alsa-user(a)lists.sourceforge.net Hi all, Here is the description and workaround to have sound. It's related to mixer..but I don't know why it saves wrong mixer settings. Thanks for help #I have this problem (no sound when I start the system) since I upgrade to kernel 2.6.26 (driver: ice1724) #rm asound.state helps but after stopping alsa #This is in /var/lib/alsa: localhost alsa # rm asound.state rm: remove regular file `asound.state'? y localhost alsa # /etc/init.d/alsasound restart * WARNING: you are stopping a boot service. * Stopping pulseaudio ... [ ok ] * Storing ALSA Mixer Levels ... [ ok ] * Killing processes using ALSA ... [ ok ] * Unloading ALSA modules ... [ ok ] * Loading ALSA modules ... * Could not detect custom ALSA settings. Loading all detected alsa drivers. * Loading: snd_ice1724 ... [ ok ] * Restoring Mixer Levels ... alsactl: set_control:1300: incompatible field type for control #1 alsactl: set_control:1300: incompatible field type for control #32 [ ok ] * Starting pulseaudio ... #No sound with amarok (I restart amarok) [ ok ] localhost alsa # /etc/init.d/alsasound stop * WARNING: you are stopping a boot service. * Stopping pulseaudio ... [ ok ] * Storing ALSA Mixer Levels ... [ ok ] * Killing processes using ALSA ... [ ok ] * Unloading ALSA modules ... [ ok ] localhost alsa # rm asound.state rm: remove regular file `asound.state'? y localhost alsa # /etc/init.d/alsasound start * Loading ALSA modules ... * Could not detect custom ALSA settings. Loading all detected alsa drivers. * Loading: snd_ice1724 ... [ ok ] * Restoring Mixer Levels ... * No mixer config in /var/lib/alsa/asound.state, you have to unmute your card! [ ok ] #So I raise up the level of PCM: localhost alsa # alsamixer localhost alsa # /etc/init.d/pulseaudio start * Starting pulseaudio ... [ ok ] localhost alsa # #Now I have sound....

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[alsa-devel] Altec Lansing FX5051 USB Speakers
by Jeremy Nickurak 19 Jul '08

19 Jul '08

This didn't appear to make it through to the list when I tried a few days ago to send it, so I'll try again. --- It's been about 7 months since I last brought this up, and didn't really get any success at the time, so I thought I'd check and see if anybody has any insights again. About a year ago I posted my original request here for any suggestions in getting a set of USB speakers from Altec Lansing, model# FX5051, working with ALSA. They're a set of 5.1 speakers with a built in USB sound card. Sound does appear to work, but the mixer is locked at 100%. There is no analog manual mixer, and the digital control dial exhibits the same problem as alsamixer, so I have to find a way to adjust the usb device's mixer volume in order to make the speakers usable. The speakers work fine in Windows XP, without installing any drivers, so it would seem that this device is operating based on some standard. In windows, adjusting the mixer via the digital control dial, or via the windows mixer control both result in the LED display moving in sync to indicate the new volume setting. In alsa, adjusting the digital control dial doesn't appear to have any effect, other then waking the LED display from its low-light mode. Attempting to adjust volume from alsamixer likewise wakes the display from low-light mode, but also fails to significantly change the mixer setting. When I say significantly, I mean that the mixer flips between about 98% and 100%, when using the Up/Down keys (or PgUp or PgDown). End appears to drop the volume to 0%, but Home doesn't restore it. Pressing the device's 'Power' button also behaves strangely. I can turn the device off, but it is turned back on shortly afterward. No usb connect/disconnect events appear here, instead, the power button seems to function more like a mute. amixer -c 2 contents yields: atrus@kedri:~$ amixer -c 2 contents numid=3,iface=MIXER,name='Tone Control - Bass' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=1 numid=4,iface=MIXER,name='Tone Control - Treble' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=3 numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=49,step=0 : values=48,48 | dBscale-min=-49.00dB,step=1.00dB,mute=0 I'm attaching the contents of 'lsusb -v'. Any suggestions would be appreciated, and any requests for further information about this device will be accomodated ASAP. Thanks in advance. -- Jeremy Nickurak

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[alsa-devel] Altec Lansing FX5051 USB Speakers
by Jeremy Nickurak 19 Jul '08

19 Jul '08

It's been about 7 months since I last brought this up, and didn't really get any success at the time, so I thought I'd check and see if anybody has any insights again. About a year ago I posted my original request here for any suggestions in getting a set of USB speakers from Altec Lansing, model# FX5051, working with ALSA. They're a set of 5.1 speakers with a built in USB sound card. Sound does appear to work, but the mixer is locked at 100%. There is no analog manual mixer, and the digital control dial exhibits the same problem as alsamixer, so I have to find a way to adjust the usb device's mixer volume in order to make the speakers usable. The speakers work fine in Windows XP, without installing any drivers, so it would seem that this device is operating based on some standard. In windows, adjusting the mixer via the digital control dial, or via the windows mixer control both result in the LED display moving in sync to indicate the new volume setting. In alsa, adjusting the digital control dial doesn't appear to have any effect, other then waking the LED display from its low-light mode. Attempting to adjust volume from alsamixer likewise wakes the display from low-light mode, but also fails to significantly change the mixer setting. When I say significantly, I mean that the mixer flips between about 98% and 100%, when using the Up/Down keys (or PgUp or PgDown). End appears to drop the volume to 0%, but Home doesn't restore it. Pressing the device's 'Power' button also behaves strangely. I can turn the device off, but it is turned back on shortly afterward. No usb connect/disconnect events appear here, instead, the power button seems to function more like a mute. amixer -c 2 contents yields: atrus@kedri:~$ amixer -c 2 contents numid=3,iface=MIXER,name='Tone Control - Bass' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=1 numid=4,iface=MIXER,name='Tone Control - Treble' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=3 numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=49,step=0 : values=48,48 | dBscale-min=-49.00dB,step=1.00dB,mute=0 I'm attaching the contents of 'lsusb -v'. Any suggestions would be appreciated, and any requests for further information about this device will be accomodated ASAP. Thanks in advance. -- Jeremy Nickurak

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 19 Jul '08

19 Jul '08

> From: steshenko_sergei(a)list.ru > To: b3zdomny(a)hotmail.com > CC: rene.herman(a)keyaccess.nl; tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; alsa(a)sc8-sf-spam2.sourceforge.net > Subject: Re[2]: [Alsa-user] is this card supported by ALSA? > Date: Fri, 18 Jul 2008 23:47:45 +0400 <snip> > > Were you in the correct subdirectory when running 'patch' ? > > --Sergei. Yes, I was. Thanks for confirming this. best regards, Landis _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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[alsa-devel] ASOC, master clock direction
by Jon Smirl 19 Jul '08

19 Jul '08

The MPC5200 has three clock configurations 1) Clock out, normal master mode 2) Clock in, normal slave mode 3) Something Freescale calls cellphone slave but it is more like clock distribution. In cellslave mode one i2s port is in normal slave mode. The clock from this port is used to drive up to three other i2s ports in master mode. I have two choices for set_sysclk: #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 When the clock frequency changes I need to notify the i2s drivers for the dependent channels. Do we need a third choice, SND_SOC_CLOCK_SLAVE? Alternatively I can use another clock id value. -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 19 Jul '08

19 Jul '08

> Date: Wed, 16 Jul 2008 16:52:39 +0200 > From: rene.herman(a)keyaccess.nl > To: b3zdomny(a)hotmail.com > CC: tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; ALSA(a)sc8-sf-spam2.sourceforge.net > Subject: Re: [Alsa-user] is this card supported by ALSA? > > On 16-07-08 16:44, Rene Herman wrote: > > > On 16-07-08 16:40, Rene Herman wrote: > > > >> # moodprobe -r ens1371-ac97_reset_hack.diff > >> # moodprobe -r ens1371-ac97.diff > > > > I'm getting really sick of the fact that I seem to need to make at least > > one typo or other small error per post. 'modprobe' ofcourse. > > Oh, I so love myself. Try this instead: > > # patch -p1 -r < ens1371-ac97_reset_hack.diff > # patch -p1 -r < ens1371-ac97.diff > > I'll make sure I get some sleep before I post again. > > Rene. OK, this is interesting: # patch -p1 -r < ens1371-ac97_reset_hack.diff= bash: ens1371-ac97_reset_hack.diff: No such file or directory # patch -p1 -r < ens1371-ac97.diff bash: ens1371-ac97.diff: No such file or directory Go figure. Attached are two high-res pix of the card. The smallest chips were just too darned small for me to read the printing; maybe you can magnify the pix and make something of them. Thanks again for all your help. I think I'm just going to be happy with OSS for awhile and see if I can't get it to support Skype. If you decide to keep working on this and you want to test something, you're welcome to contact me. best regards as always, Landis _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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