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Re: [alsa-devel] Does LKML accept non-ASOC audio driver?
by Mark Brown 18 Sep '08

18 Sep '08

On Thu, Sep 18, 2008 at 11:01:07AM +0000, Cliff Cai wrote: > > outside of ASoC with just a platform data hook for doing the codec > > initialisation would probably be the best bet. Could you point me at > > the existing drivers? > One thing I should let you know ,the initialisation code will highly depend > on the low-level SPORT operation code in blackfin arch. That's absolutely fine - the machine driver is entirely specific to the hardware it's connecting so things like this are not a problem. It already explicitly references the codec and machine drivers anyway. > > Actually, looking at the datasheet for the AD73311 it is actually > > possible to reconfigure the codec at run time at the expense of one data > > bit. > You are right,but it's not a good way in real use,since we need to format every > PCM sample in that mode. Yeah, without trusting the application data to leave the bit clear it'd need some glue in hardware to force the MSB low when not doing control operations.

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Re: [alsa-devel] Does LKML accept non-ASOC audio driver?
by Mark Brown 18 Sep '08

18 Sep '08

On Thu, Sep 18, 2008 at 09:48:20AM +0000, Cliff Cai wrote: > Yes,the hardware doesn't have seperate codec control, > we use the same bus as for DMA transmission to configure the codec( e.g.AD73311) That shouldn't be an issue, AC97 does the same thing but... > and after configuring it,the codec will enter data mode,and is not configurable any more. ...this might be more of an issue. That said, is the SPORT on the DSP likely to be used with smarter codecs (it looks like it's already used by your existing ASoC drivers but perhaps differently?)? If it is then it's probably worth fitting it in to ASoC in order to avoid having two distinct drivers for it. It should be possible for the machine driver to do the codec setup when it probes and then just have a dumb DAI for the codec with no operations (actually, it's worth adding that dumb codec driver anyway - it's not like dumb ADCs and DACs are uncommon). If there's not much code to reuse then a simpler driver for the SPORT outside of ASoC with just a platform data hook for doing the codec initialisation would probably be the best bet. Could you point me at the existing drivers? Actually, looking at the datasheet for the AD73311 it is actually possible to reconfigure the codec at run time at the expense of one data bit.

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Re: [alsa-devel] Does LKML accept non-ASOC audio driver?
by Mark Brown 18 Sep '08

18 Sep '08

On Thu, Sep 18, 2008 at 09:16:08AM +0000, Cliff Cai wrote: > We(blackfin arch) have several audio codec drivers which are written in alsa style,actually they are hard to transfer to asoc. What are the problems with this - is it a case of the hardware not fitting the model well or are there other difficulties with the ASoC framework? > I also saw that there are several arch-based directories in sound/, e.g.sound/arm, > does this mean non-asoc driver is also accepted currently? Yes, ASoC only needs to be used in situations where it makes sense to do so. If the hardware doesn't have separate codec and DMA controllers, for example, then it wouldn't make much sense to try to shoehorn it into ASoC.

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[alsa-devel] scheduling midi events with pyalsa and making queue to loop after few bars..
by ki.ber.kom.uni.st 18 Sep '08

18 Sep '08

hm.. i would like to make simple example of scheduling few midi notes to queue and alsaseq where the length of the loop will be one bar or any number of ticks... i'm kind of stuck :( i got two midi notes triggered (event & event2) but don't know how to play the queue again or in loop while notes which are already there should be played in every repetition... i'm trying to do that placing alsaseq.SEQ_EVENT_SETPOS_TICK at 0 (or 1) but it doesn't play the notes again... my logic is to schedule the playhead of queue at the beginning after some number of ticks and i assume that already queued notes will be there.. i'm quite sure i'm wrong ;) what would be good logic to do that? do i need to leave queue alone and implement my own logic on top of it? something like adding notes all along the timeline simulating the repetition that way... i would like to know or to check what's the tick of running queue.. do i do that scheduling alsaseq.SEQ_EVENT_ECHO some place on the timeline? hm... lot's of questions... hope someone would explain to me a bit of the alsaseq inner logic... i'm sorry i couldn't find good documentation but testing the library inside ipython.... here is how my session looks like (i connect "myPort" to Virtual Raw Midi with a little help of qjackctl and monitor it with amidi -p hw:2,0 -d): from pyalsa import alsaseq seqsen = alsaseq.Sequencer() seqsen.create_simple_port("myPort", alsaseq.SEQ_PORT_TYPE_APPLICATION, alsaseq.SEQ_PORT_CAP_SUBS_READ | alsaseq.SEQ_PORT_CAP_READ) queue = seqsen.create_queue() seqsen.start_queue(queue) queue = seqsen.create_queue() seqsen.start_queue(queue) tempo, ppq = seqsen.queue_tempo(queue) event = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_PGMCHANGE) event.queue = queue event.time = 96 event.set_data({'control.value' : 127}) seqsen.output_event(event) event2 = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_PGMCHANGE) event2.queue = queue event2.time = 96*4 event2.set_data({'control.value' : 100}) seqsen.output_event(event2) event3 = alsaseq.SeqEvent(alsaseq.SEQ_EVENT_SETPOS_TICK) event3.queue = queue event3.time = 96*5 event3.set_data({'queue.queue' : 1}) seqsen.output_event(event3) seqsen.drain_output() ## event and event2 being sent but event3 is unnoticeable to me...

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[alsa-devel] ALSA:AT91SAM9G20: Add support(DAI) for wolfson8731
by Sedji Gaouaou 17 Sep '08

17 Sep '08

Hi, So here is a patch to add support for the wolfson 8731 on the at91sam9g20 evaluation board. Any comments will be welcomed!! Regards, Sedji sound/soc/at91/Kconfig | 19 +- sound/soc/at91/Makefile | 6 + sound/soc/at91/at91-ssc.h | 1 + sound/soc/at91/atmel_ssc_dai.c | 715 ++++++++++++++++++++++++++++++++++ sound/soc/at91/sam9g20_wm8731.c | 290 ++++++++++++++ sound/soc/codecs/Kconfig | 2 +- 6 files changed, 1031 insertions(+), 2 deletions(-) diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 9051865..86621b9 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -7,7 +7,15 @@ config SND_AT91_SOC to select the audio interfaces to support below. config SND_AT91_SOC_SSC - tristate + tristate "at91 soc ssc" + +config SND_ATMEL_SOC_SSC + tristate "ATMEL SoC ssc dai" + depends on ATMEL_SSC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface(interface between at91sam9g20 and + WM8731 for instance). config SND_AT91_SOC_ETI_B1_WM8731 tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" @@ -18,6 +26,15 @@ config SND_AT91_SOC_ETI_B1_WM8731 Say Y if you want to add support for SoC audio on WM8731-based Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_AT91_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + config SND_AT91_SOC_ETI_SLAVE bool "Run codec in slave Mode on Endrelia boards" depends on SND_AT91_SOC_ETI_B1_WM8731 diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile index f23da17..f12a798 100644 --- a/sound/soc/at91/Makefile +++ b/sound/soc/at91/Makefile @@ -1,11 +1,17 @@ # AT91 Platform Support snd-soc-at91-objs := at91-pcm.o snd-soc-at91-ssc-objs := at91-ssc.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + # AT91 Machine Support snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index 6b7bf38..b967691 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -22,6 +22,7 @@ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ extern struct snd_soc_dai at91_ssc_dai[]; +extern struct snd_soc_dai atmel_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/atmel_ssc_dai.c b/sound/soc/at91/atmel_ssc_dai.c new file mode 100644 index 0000000..a3888c0 --- /dev/null +++ b/sound/soc/at91/atmel_ssc_dai.c @@ -0,0 +1,715 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.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. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> +#include <mach/at91_pmc.h> + +#include "at91-pcm.h" +#include "at91-ssc.h" + +#if 0 +#define DBG(x...) printk(KERN_DEBUG "atmel_ssc_dai:" x) +#else +#define DBG(x...) +#endif + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct at91_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct at91_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct at91_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct at91_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + +static struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* 1=SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at91_pcm_dma_params *dma_params[2]; +} ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = 0, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = 0, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = 0, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct at91_pcm_dma_params *dma_params; + u32 ssc_sr; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if (dma_params != NULL && dma_params->dma_intr_handler != NULL && + (ssc_sr & + (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) + + dma_params->dma_intr_handler(ssc_sr, dma_params->substream); + } + + return IRQ_HANDLED; +} + +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + DBG("ssc_startup: SSC_SR=0x%08lx\n", ssc_readl(ssc_p->ssc->regs, SR)); + dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at91_pcm_dma_params *dma_params; + int dir, dir_mask; + u32 pid; + + pid = ssc_p->ssc->pdev->resource[1].start; + + dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc_base = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + DBG("Stopping pid %d clock\n", pid); + at91_sys_write(AT91_PMC_PCDR, 1<<pid); + + if (ssc_p->initialized) { + free_irq(pid, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); + ssc_free(ssc_p->ssc); +} + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT91SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT91SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT91SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct at91_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + u32 pid; + int start_event; + int ret; + + ssc_p->ssc = ssc_request(0); + pid = ssc_p->ssc->pdev->resource[1].start; + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc_base = ssc_p->ssc->regs; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, 4) + | SSC_BF(RCMR_CKI, 1) + | SSC_BF(RCMR_CKO, 0) + | SSC_BF(RCMR_CKS, 0); + + rfmr = SSC_BF(RFMR_FSEDGE, 0) + | SSC_BF(RFMR_FSOS, 1) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BF(RFMR_MSBF, 1) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_CKI, 0) + | SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_CKS, 0); + + tfmr = SSC_BF(TFMR_FSEDGE, 0) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, 1) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BF(TFMR_MSBF, 1) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is generated from the + * transmit clock. + * + * For single channel data, one sample is transferred on the falling + * edge of the LRC clock. For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = channels == 1 + ? 4 + : 7; + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, 1) + | SSC_BF(RCMR_CKO, 0) + | SSC_BF(RCMR_CKS, 1); + + rfmr = SSC_BF(RFMR_FSEDGE, 0) + | SSC_BF(RFMR_FSOS, 0) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_MSBF, 1) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, 0) + | SSC_BF(TCMR_CKO, 0) + | SSC_BF(TCMR_CKS, 2); + + tfmr = SSC_BF(TFMR_FSEDGE, 0) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, 0) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_MSBF, 1) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, 5) + | SSC_BF(RCMR_CKI, 1) + | SSC_BF(RCMR_CKO, 0) + | SSC_BF(RCMR_CKS, 0); + + rfmr = SSC_BF(RFMR_FSEDGE, 0) + | SSC_BF(RFMR_FSOS, 2) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BF(RFMR_MSBF, 1) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, 5) + | SSC_BF(TCMR_CKI, 1) + | SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_CKS, 0); + + tfmr = SSC_BF(TFMR_FSEDGE, 0) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, 2) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BF(TFMR_MSBF, 1) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x.\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + DBG("Starting pid %d clock\n", pid); + at91_sys_write(AT91_PMC_PCER, 1<<pid); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(pid, atmel_ssc_interrupt, + 0, ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING "atmel_ssc_dai: request_irq failure\n"); + + DBG("Stopping pid %d clock\n", pid); + at91_sys_write(AT91_PMC_PCER, 1<<pid); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + DBG("hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at91_pcm_dma_params *dma_params; + int dir; + + dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +#define atmel_ssc_suspend NULL +#define atmel_ssc_resume NULL +#else +#define atmel_ssc_suspend NULL +#define atmel_ssc_resume NULL +#endif + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0].ssc, + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1].ssc, + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2].ssc, + }, +#endif +}; + +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/sam9g20_wm8731.c b/sound/soc/at91/sam9g20_wm8731.c new file mode 100644 index 0000000..8d77830 --- /dev/null +++ b/sound/soc/at91/sam9g20_wm8731.c @@ -0,0 +1,290 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "at91-pcm.h" +#include "at91-ssc.h" + + +#if 0 +#define DBG(x...) printk(KERN_INFO "at91sam9g20ek_wm8731: " x) +#else +#define DBG(x...) +#endif + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + DBG("pck0 stopped\n"); +} + +static int at91sam9g20ek_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 *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + unsigned int rate; + int cmr_div, period; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + AT91SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + AT91SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + + DBG("at91sam9g20ek_wm8731 : at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &at91_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + int ret; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + DBG("platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + DBG("platform device add failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + platform_device_unregister(at91sam9g20ek_snd_device); +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..1595b04 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_WM8510 tristate config SND_SOC_WM8731 - tristate + tristate "WM8731" config SND_SOC_WM8750 tristate

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[alsa-devel] snd-usb-audio: Roland Fantom-G Support
by Luís Filipe Lobo 17 Sep '08

17 Sep '08

Hi! The Roland Fantom-G is the most recent synth from Roland and is not supported in the current usb-audio version. I am contacting this list to know how can I contribute to get it working. I've added a new entry in usbquirks.h copied from Roland Edirol UA-100 and change the vendor/product id. Without further changes I got midi working perfectly. Now I am having difficulty establishing a relation between the output of 'lsusb -v ...' and the remaining parameters in my usbquirks entry (endpoint, ep_attr, ...). Any help/references would be much apreciated. I also attached the output of 'lsusb -v -d 0582:00de' hoping it will be of any use. Thanks in advance, Regards, -- Luís Filipe Lobo Bus 005 Device 005: ID 0582:00de 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 0x00de bcdDevice 0.01 iManufacturer 1 Roland iProduct 2 Fantom G 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 2 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 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 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 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 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 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 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 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 8 Mass Storage bDeviceSubClass 6 SCSI bDeviceProtocol 80 Bulk (Zip) bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0001 Self Powered

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Re: [alsa-devel] Unable to play aplay
by Nitin Mahajan 17 Sep '08

17 Sep '08

HI! ----- Original Message ---- > From: vasudha rao <vasudha.jrao(a)gmail.com> > To: aggarwal <anuj.aggarwal(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org > Sent: Tuesday, September 16, 2008 17:08:30 > Subject: Re: [alsa-devel] Unable to play aplay > > I inserted modules to my linux kernel (2.6.20)like.. > > $insmod snd.ko > $insmod snd-timer.ko > $insmod snd-page-alloc.ko > $insmod snd-pcm.ko > $insmod ac97_bus.ko > $insmod snd-ac97-codec.ko > $insmod snd-at91-ac97.ko > > After this, I am getting, > > $aplay > > No error messages.. > > $aplay laughter.wav > Playing WAVE 'laughter.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono > Bad pte = 5fc3e05f, process = aplay, vm_flags = 800fb, vaddr = 4032c000 > Bad pte = 5fc3f05f, process = aplay, vm_flags = 800fb, vaddr = 4032d000 > Unable to handle kernel paging request at virtual address ff800001 > pgd = c346c000 > [ff800001] *pgd=00000000 > Internal error: Oops: 1 [#1] > Modules linked in: snd_at91_ac97 snd_ac97_codec ac97_bus snd_pcm > snd_page_alloc snd_timer snd > CPU: 0 > PC is at snd_pcm_mmap_data_nopage+0xc4/0xe8 [snd_pcm] > LR is at 0x4032e000 > pc : [] lr : [<4032e000>] Not tainted > sp : c3bb3e6c ip : ff7ffffd fp : c3bb3e80 > r10: 4032e000 r9 : c001f380 r8 : c3bb2000 > r7 : c3449764 r6 : 00000000 r5 : c3bb3ea0 r4 : c34a8c00 > r3 : 00000093 r2 : 00000013 r1 : 00006000 r0 : c0aa3800 > Flags: nzcv IRQs off FIQs on Mode SVC_32 Segment user > Control: 5317F > Table: 2346C000 DAC: 00000015 > Process aplay (pid: 916, stack limit = 0xc3bb2250) > Stack: (0xc3bb3e6c to 0xc3bb4000) > 3e60: 00000000 c3e192ec c3bb3ecc c3bb3e84 > c0067c58 > 3e80: bf019030 c346d008 c001f3c0 5fc3f05f 00000000 00000000 c346d008 > 00000000 > 3ea0: 00000002 ffffffeb c3d2b040 c3449764 c001f380 c001f3b4 c3bb3fb0 > 4032e000 > 3ec0: c3bb3f04 c3bb3ed0 c00293a4 c0067b2c 00022b70 00000000 00000017 > ffffffff > 3ee0: c027900c 00000017 c3bb3fb0 4032e000 40388000 400cb5fc c3bb3fac > c3bb3f08 > 3f00: c00295c0 c00292dc c3bb3f2c c0377f20 ffffffe7 c0844123 00000000 > c0023ec4 > 3f20: 400cb5fc c3bb3f40 c3bb3f34 bf018af4 bf018194 c3bb3f5c c3bb3f44 > c00823d0 > 3f40: bf018ad0 c0377f20 00022b70 00000004 c3bb3f84 c3bb3f60 c00826ac > c00823a4 > 3f60: 000014c2 ffffffff c3bb3fac c3bb3f78 c01f7ee0 c00275a8 c3d2b080 > c3d2b154 > 3f80: 0082ca24 ffffffff 40388734 ffffffff 40389734 00000008 00000004 > 00000004 > 3fa0: 00000000 c3bb3fb0 c0023cc8 c0029594 00000000 4032e000 40354000 > 00000100 > 3fc0: 00006800 40389734 00000008 00000004 00000004 40388000 400cb5fc > 00006dcd > 3fe0: 00000100 bea539d0 00000100 4008ce20 20000010 ffffffff 00000000 > 00000029 > Backtrace: > [] (snd_pcm_mmap_data_nopage+0x0/0xe8 [snd_pcm]) from [] > (__handle_mm_fault+0x13c/0x760) > r5 = C3E192EC r4 = 00000000 > [] (__handle_mm_fault+0x0/0x760) from [] > (do_page_fault+0xd8/0x20c) > [] (do_page_fault+0x0/0x20c) from [] > (do_DataAbort+0x3c/0xa0) > [] (do_DataAbort+0x0/0xa0) from [] > (ret_from_exception+0x0/0x10) > r8 = 00000004 r7 = 00000004 r6 = 00000008 r5 = 40389734 > r4 = FFFFFFFF > Code: 1590c00c e10f2000 e3823080 e121f003 (e59c3004) > Segmentation fault > > > $aplay -l > card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > $arecord > ALSA lib pcm_dmix.c:812:(snd_pcm_dmix_open) The dmix plugin supports only > playback stream > arecord: main:446: audio open error: Invalid argument > > $arecord -l > As far as arecord is concerned,check if your codec chip has an ADC, what mixer controls are available for recording? Are u setting them ? > No message here.. > > $cat /proc/asound/pcm > 00-00: Atmel AC97 : Atmel AC97 : playback 1 > > Why I am not able to here or record sound..?? > Where I am doing wrong...?? > > Thank you.. > > > > > > > > > > > > > On Thu, Sep 11, 2008 at 10:46 AM, vasudha rao wrote: > > > Hi anuj , > > > > Sorry my mistake..that time I hav not loaded conf files in /usr/share/ .. I > > loaded now.. > > So now when I give command like, > > > > ------------------------------------------------- > > $aplay -l > > **** List of PLAYBACK Hardware Devices **** > > card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > --------------------------------------------------- > > > > but,aplay not proper,, > > ---------------------------- > > $aplay > > aplay: main:508: audio open error: Inappropriate ioctl for device > > ---------------------------- > > > > Where I am doing wrong..?? > > > > Thank you.. > > > > On Thu, Sep 11, 2008 at 10:07 AM, vasudha rao wrote: > > > >> > >> > >> > >> Hi Anuj, > >> > >> Nodes were not there before..I created now.. > >> > >> Now when I did $aplay -l , > >> I got, > >> > >> **** List of PLAYBACK Hardware Devices *** > >> *ALSA lib control.c:739:(snd_ctl_open_noupdate) Invalid CTL hw:0 > >> aplay: device_list:208: control open (0): No such file or directory > >> aplay: conf.c:3083: snd_config_update_free: Assertion `update->count > 0 > >> && update->finfo' failed.Aborted > >> > >> Where I am doing wrong..?? > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> Thank you.. > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Sep 10, 2008 at 11:54 PM, anuj aggarwal > wrote: > >> > >>> Vasudha, > >>> > >>> Could you please check the device nodes in /dev/snd folder? The > >>> desired output is: > >>> > >>> [root@ /home]# ls -l /dev/snd/ > >>> crw-rw---- 1 root root 116, 5 Jan 1 00:00 controlC0 > >>> crw-rw---- 1 root root 116, 4 Jan 1 00:00 pcmC0D0c > >>> crw-rw---- 1 root root 116, 3 Jan 1 00:00 pcmC0D0p > >>> > >>> If the nodes are not there, you have to create them first before trying > >>> aplay. > >>> > >>> Regards, > >>> Anuj > >>> > >>> > >>> On Wed, Sep 10, 2008 at 6:16 PM, vasudha rao > >>> wrote: > >>> > > >>> > Hi , > >>> > > >>> > I want to play and record audio file in my custom atmel 9263 board > >>> running > >>> > with linux(2.6.20). > >>> > when I tried to play aplay for a wave file, I got > >>> > > >>> > $aplay /home/alarm.wav > >>> > ALSA lib pcm.c:1707:(snd_pcm_open_noupdate) Unknown PCM default > >>> > aplay: main:446: audio open error: No such file or directory > >>> > > >>> > $aplay -l > >>> > aplay: device_list:187: no soundcards found... > >>> > > >>> > $cat /proc/asound/devices > >>> > 2: : timer > >>> > 3: : sequencer > >>> > 4: [ 0- 0]: digital audio playback > >>> > 5: [ 0] : control > >>> > > >>> > Can you tel me where I am doing wrong..?? > >>> > > >>> > Thanks in advance.. > >>> > _______________________________________________ > >>> > Alsa-devel mailing list > >>> > Alsa-devel(a)alsa-project.org > >>> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >>> > >>> > >>> > >>> -- > >>> Best Regards, > >>> Anuj Aggarwal > >>> > >> > >> > > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Get your preferred Email name! 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Re: [alsa-devel] [PATCH] ALSA: HDA: hda_local: Less magic numbers.
by Jonathan Woithe 17 Sep '08

17 Sep '08

Robin H. Johnson wrote: > Explain some of the magic numbers I saw while trying to fix the AD1989 SPDIF > issue. Maybe should just use the expanded form directly in the verbs? > > Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> > > diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h > index d688f50..7957fef 100644 > --- a/sound/pci/hda/hda_local.h > +++ b/sound/pci/hda/hda_local.h > @@ -368,12 +368,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, > #define AMP_OUT_UNMUTE 0xb000 > #define AMP_OUT_ZERO 0xb000 > /* pinctl values */ > -#define PIN_IN 0x20 > -#define PIN_VREF80 0x24 > -#define PIN_VREF50 0x21 > -#define PIN_OUT 0x40 > -#define PIN_HP 0xc0 > -#define PIN_HP_AMP 0x80 > +#define PIN_IN (AC_PINCTL_IN_EN) > +#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) > +#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50) > +#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) > +#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80) > +#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) > +#define PIN_OUT (AC_PINCTL_OUT_EN) > +#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) > +#define PIN_HP_AMP (AC_PINCTL_HP_EN) Shrug. In my opinion the above patch simply reworks the explanation, shifting the "magic" from the PIN_* definitions to the AC_PINCTL_* defines. One could take things one step further and replace the PIN_* defines with AC_PINCTL_* references throughout the code, but this would make the code far more verbose and harder to follow. At the end of the day I'm not fussed either way. Having said that, I don't really see the practical advantage of the change but if Takashi et al thing it's a good idea then I wouldn't oppose it. Regards jonathan

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[alsa-devel] [PATCH] alsactl crashes on 1.0.18rc3 alsa-utils
by Colin Guthrie 17 Sep '08

17 Sep '08

Hi, One of my Mandriva colleagues (Gustavo De Nardin) fixed this in the following patch: http://svn.mandriva.com/cgi-bin/viewvc.cgi/packages/cooker/alsa-utils/curre… Please apply if it's not already been fixed. Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] help, please nVidia Corporation MCP73
by lazaro souza 16 Sep '08

16 Sep '08

please help me, I have an equipment with a motherboard gft050vt-m5 with the descriptions below, the problem that springs to mind is with the sound front, because it uses a conexçao c, 97 and another hd audio but only works the front microphone and the sound does not come out, but connecting on the back the sound comes out perfect, how to resolve this? as tested in several router with kernel and alsa different, and the problem is the same, PS in ruindows works installing additional software. I look forward in a contact. lspci | grep Audio 00:09.0 Audio device: nVidia Corporation MCP73 High Definition Audio (rev a1) lspci -n | grep 00:09.0 00:09.0 0403: 10de:07fc (rev a1) cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfebf8000 irq 23 por favor me ajudem, tenho um equipamento com uma placa mae gft050vt-m5 com as descrições abaixo, o problema que me ocorre é com o som frontal, pois a mesma utiliza uma conexçao ac,97 e outra hd audio mas só funciona o microfone frontal e o som não sai, mas conectando na parte traseira o som sai perfeito, como resolver isto ? pois testei em varias distribuiçoes com kernel e alsa diferentes, e o problema é o mesmo, OBS: no ruindows funciona instalando um software adicional. fico no aguardo de um contato. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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