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[alsa-devel] Driver code with mpc5200 pointer problem.
by Jon Smirl 05 May '09

05 May '09

Here's the code computing the mpc5200 dma pointer. Could you please take a look at it and let me know what it is doing wrong. /* * Freescale MPC5200 Audio DMA * ALSA SoC Platform driver * * Copyright (C) 2008 Secret Lab Technologies Ltd. */ #define DEBUG #include <linux/init.h> #include <linux/module.h> #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/of_device.h> #include <linux/of_platform.h> #include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-of-simple.h> #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_dma.h" MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); MODULE_LICENSE("GPL"); /* * Interrupt handlers */ static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) { struct psc_dma *psc_dma = _psc_dma; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 isr; isr = in_be16(&regs->mpc52xx_psc_isr); /* Playback underrun error */ if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) psc_dma->stats.underrun_count++; /* Capture overrun error */ if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) psc_dma->stats.overrun_count++; out_8(&regs->command, 4 << 4); /* reset the error status */ return IRQ_HANDLED; } /** * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer * @s: pointer to stream private data structure * * Enqueues another audio period buffer into the bestcomm queue. * * Note: The routine must only be called when there is space available in * the queue. Otherwise the enqueue will fail and the audio ring buffer * will get out of sync */ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) { struct bcom_bd *bd; /* Prepare and enqueue the next buffer descriptor */ bd = bcom_prepare_next_buffer(s->bcom_task); bd->status = s->period_bytes; bd->data[0] = s->period_next_pt; bcom_submit_next_buffer(s->bcom_task, NULL); /* Update for next period */ s->period_next_pt += s->period_bytes; if (s->period_next_pt >= s->period_end) s->period_next_pt = s->period_start; } /* Bestcomm DMA irq handler */ static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; /* For each finished period, dequeue the completed period buffer * and enqueue a new one in it's place. */ while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); s->period_current_pt += s->period_bytes; if (s->period_current_pt >= s->period_end) s->period_current_pt = s->period_start; psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); } /* If the stream is active, then also inform the PCM middle layer * of the period finished event. */ if (s->active) { s->jiffies = jiffies; snd_pcm_period_elapsed(s->stream); } return IRQ_HANDLED; } /** * psc_dma_startup: create a new substream * * This is the first function called when a stream is opened. * * If this is the first stream open, then grab the IRQ and program most of * the PSC registers. */ int mpc5200_dma_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; int rc; dev_dbg(psc_dma->dev, "psc_dma_startup(substream=%p)\n", substream); if (!psc_dma->playback.active && !psc_dma->capture.active) { /* Setup the IRQs */ rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, "psc-dma-status", psc_dma); rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-capture", &psc_dma->capture); rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { free_irq(psc_dma->irq, psc_dma); free_irq(psc_dma->capture.irq, &psc_dma->capture); free_irq(psc_dma->playback.irq, &psc_dma->playback); return -ENODEV; } } return 0; } EXPORT_SYMBOL_GPL(mpc5200_dma_startup); int mpc5200_dma_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; } EXPORT_SYMBOL_GPL(mpc5200_dma_hw_free); /** * psc_dma_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ int mpc5200_dma_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct psc_dma_stream *s; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; u8 psc_cmd; unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)" " stream_id=%i\n", substream, cmd, substream->pstr->stream); switch (cmd) { case SNDRV_PCM_TRIGGER_START: s->period_bytes = frames_to_bytes(runtime, runtime->period_size); s->period_start = virt_to_phys(runtime->dma_area); s->period_end = s->period_start + (s->period_bytes * runtime->periods); s->period_next_pt = s->period_start; s->period_current_pt = s->period_start; s->active = 1; /* First; reset everything */ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { out_8(&regs->command, MPC52xx_PSC_RST_RX); out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); } else { out_8(&regs->command, MPC52xx_PSC_RST_TX); out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); } /* Next, fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. */ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) bcom_gen_bd_rx_reset(s->bcom_task); else bcom_gen_bd_tx_reset(s->bcom_task); while (!bcom_queue_full(s->bcom_task)) psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); /* Due to errata in the dma mode; need to line up enabling * the transmitter with a transition on the frame sync * line */ spin_lock_irqsave(&psc_dma->lock, flags); /* first make sure it is low */ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) ; /* then wait for the transition to high */ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) ; /* Finally, enable the PSC. * Receiver must always be enabled; even when we only want * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ psc_cmd = MPC52xx_PSC_RX_ENABLE; if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) psc_cmd |= MPC52xx_PSC_TX_ENABLE; out_8(&regs->command, psc_cmd); spin_unlock_irqrestore(&psc_dma->lock, flags); break; case SNDRV_PCM_TRIGGER_STOP: /* Turn off the PSC */ s->active = 0; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { if (!psc_dma->playback.active) { out_8(&regs->command, 2 << 4); /* reset rx */ out_8(&regs->command, 3 << 4); /* reset tx */ out_8(&regs->command, 4 << 4); /* reset err */ } } else { out_8(&regs->command, 3 << 4); /* reset tx */ out_8(&regs->command, 4 << 4); /* reset err */ if (!psc_dma->capture.active) out_8(&regs->command, 2 << 4); /* reset rx */ } bcom_disable(s->bcom_task); while (!bcom_queue_empty(s->bcom_task)) bcom_retrieve_buffer(s->bcom_task, NULL, NULL); break; default: dev_dbg(psc_dma->dev, "invalid command\n"); return -EINVAL; } /* Update interrupt enable settings */ imr = 0; if (psc_dma->playback.active) imr |= MPC52xx_PSC_IMR_TXEMP; if (psc_dma->capture.active) imr |= MPC52xx_PSC_IMR_ORERR; out_be16(&regs->isr_imr.imr, imr); return 0; } EXPORT_SYMBOL_GPL(mpc5200_dma_trigger); /** * psc_dma_shutdown: shutdown the data transfer on a stream * * Shutdown the PSC if there are no other substreams open. */ void mpc5200_dma_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; dev_dbg(psc_dma->dev, "psc_dma_shutdown(substream=%p)\n", substream); /* * If this is the last active substream, disable the PSC and release * the IRQ. */ if (!psc_dma->playback.active && !psc_dma->capture.active) { /* Disable all interrupts and reset the PSC */ out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset tx */ out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */ out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ /* Release irqs */ free_irq(psc_dma->irq, psc_dma); free_irq(psc_dma->capture.irq, &psc_dma->capture); free_irq(psc_dma->playback.irq, &psc_dma->playback); } } EXPORT_SYMBOL_GPL(mpc5200_dma_shutdown); /* --------------------------------------------------------------------- * The PSC DMA 'ASoC platform' driver * * Can be referenced by an 'ASoC machine' driver * This driver only deals with the audio bus; it doesn't have any * interaction with the attached codec */ static const struct snd_pcm_hardware psc_dma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .period_bytes_max = 1024 * 1024, .period_bytes_min = 32, .periods_min = 2, .periods_max = 256, .buffer_bytes_max = 2 * 1024 * 1024, .fifo_size = 0, }; static int psc_dma_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; dev_dbg(psc_dma->dev, "psc_dma_pcm_open(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; snd_soc_set_runtime_hwparams(substream, &psc_dma_pcm_hardware); s->stream = substream; return 0; } static int psc_dma_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; dev_dbg(psc_dma->dev, "psc_dma_pcm_close(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; s->stream = NULL; return 0; } static snd_pcm_uframes_t psc_dma_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; dma_addr_t count; snd_pcm_uframes_t frames; int delta; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; count = s->period_current_pt - s->period_start; delta = jiffies - s->jiffies; delta = delta * runtime->rate / HZ; frames = bytes_to_frames(substream->runtime, count); return frames + delta; } static struct snd_pcm_ops psc_dma_pcm_ops = { .open = psc_dma_pcm_open, .close = psc_dma_pcm_close, .ioctl = snd_pcm_lib_ioctl, .pointer = psc_dma_pcm_pointer, }; static u64 psc_dma_pcm_dmamask = 0xffffffff; static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; size_t size = psc_dma_pcm_hardware.buffer_bytes_max; int rc = 0; dev_dbg(rtd->socdev->dev, "psc_dma_pcm_new(card=%p, dai=%p, pcm=%p)\n", card, dai, pcm); if (!card->dev->dma_mask) card->dev->dma_mask = &psc_dma_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (pcm->streams[0].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, size, &pcm->streams[0].substream->dma_buffer); if (rc) goto playback_alloc_err; } if (pcm->streams[1].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, size, &pcm->streams[1].substream->dma_buffer); if (rc) goto capture_alloc_err; } if (rtd->socdev->card->codec->ac97) rtd->socdev->card->codec->ac97->private_data = psc_dma; return 0; capture_alloc_err: if (pcm->streams[0].substream) snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); playback_alloc_err: dev_err(card->dev, "Cannot allocate buffer(s)\n"); return -ENOMEM; } static void psc_dma_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_pcm_substream *substream; int stream; dev_dbg(rtd->socdev->dev, "psc_dma_pcm_free(pcm=%p)\n", pcm); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; if (substream) { snd_dma_free_pages(&substream->dma_buffer); substream->dma_buffer.area = NULL; substream->dma_buffer.addr = 0; } } } struct snd_soc_platform mpc5200_dma_platform = { .name = "mpc5200-psc-audio", .pcm_ops = &psc_dma_pcm_ops, .pcm_new = &psc_dma_pcm_new, .pcm_free = &psc_dma_pcm_free, }; EXPORT_SYMBOL_GPL(mpc5200_dma_platform); static int __init mpc5200_soc_platform_init(void) { /* Tell the ASoC OF helpers about it */ of_snd_soc_register_platform(&mpc5200_dma_platform); return snd_soc_register_platform(&mpc5200_dma_platform); } module_init(mpc5200_soc_platform_init); static void __exit mpc5200_soc_platform_exit(void) { snd_soc_unregister_platform(&mpc5200_dma_platform); } module_exit(mpc5200_soc_platform_exit); ------------------------------------------------------------------------------- /* * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. * * Copyright 2008 Jon Smirl, Digispeaker * Author: Jon Smirl <jonsmirl(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 version 2 as * published by the Free Software Foundation. */ #define DEBUG #include <linux/init.h> #include <linux/module.h> #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/of_device.h> #include <linux/of_platform.h> #include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-of-simple.h> #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_dma.h" #include "mpc5200_psc_ac97.h" MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); MODULE_DESCRIPTION("mpc5200 AC97 module"); MODULE_LICENSE("GPL"); #define DRV_NAME "mpc5200-psc-ac97" static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct psc_dma *psc_dma = ac97->private_data; int timeout; unsigned int val; spin_lock(&psc_dma->lock); //printk("ac97 read: reg %04x\n", reg); /* Wait for it to be ready */ timeout = 1000; while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND) ) udelay(10); if (!timeout) { printk(KERN_ERR DRV_NAME ": timeout on ac97 bus (rdy)\n"); return 0xffff; } /* Do the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); /* Wait for the answer */ timeout = 1000; while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_DATA_VAL) ) udelay(10); if (!timeout) { printk(KERN_ERR DRV_NAME ": timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); return 0xffff; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if ( ((val>>24) & 0x7f) != reg ) { printk(KERN_ERR DRV_NAME ": reg echo error on ac97 read\n"); return 0xffff; } val = (val >> 8) & 0xffff; //printk("ac97 read ok: reg %04x val %04x\n", reg, val); spin_unlock(&psc_dma->lock); return (unsigned short) val; } static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct psc_dma *psc_dma = ac97->private_data; int timeout; //printk("ac97 write: reg %04x val %04x\n", reg, val); spin_lock(&psc_dma->lock); /* Wait for it to be ready */ timeout = 1000; while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND) ) udelay(10); if (!timeout) { printk(KERN_ERR DRV_NAME ": timeout on ac97 write\n"); return; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); spin_unlock(&psc_dma->lock); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { struct psc_dma *psc_dma = ac97->private_data; printk("psc_ac97_cold_reset %p\n", ac97); /* Do a cold reset */ out_8(&psc_dma->psc_regs->op1, MPC52xx_PSC_OP_RES); udelay(10); out_8(&psc_dma->psc_regs->op0, MPC52xx_PSC_OP_RES); udelay(50); /* PSC recover from cold reset (cfr user manual, not sure if useful) */ out_be32(&psc_dma->psc_regs->sicr, in_be32(&psc_dma->psc_regs->sicr)); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { printk("psc_ac97_warm_reset\n"); } struct snd_ac97_bus_ops soc_ac97_ops = { .read = psc_ac97_read, .write = psc_ac97_write, .reset = psc_ac97_cold_reset, .warm_reset = psc_ac97_warm_reset, }; EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int psc_ac97_suspend(struct snd_soc_dai *dai) { return 0; } static int psc_ac97_resume(struct snd_soc_dai *dai) { return 0; } #else #define psc_ac97_suspend NULL #define psc_ac97_resume NULL #endif static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" " rate=%i format=%i\n", __func__, substream, params_period_size(params), params_period_bytes(params), params_periods(params), params_buffer_size(params), params_buffer_bytes(params), params_channels(params), params_rate(params), params_format(params)); // FIXME, need a spinlock to protect access if (params_channels(params) == 1) out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); else out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { return 0; } /** * psc_ac97_set_fmt: set the serial format. * * This function is called by the machine driver to tell us what serial * format to use. * * This driver only supports AC97 mode. Return an error if the format is * not SND_SOC_DAIFMT_AC97. * * @format: one of SND_SOC_DAIFMT_xxx */ static int psc_ac97_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { struct psc_dma *psc_dma = cpu_dai->private_data; dev_dbg(psc_dma->dev, "psc_ac97_set_fmt(cpu_dai=%p, format=%i)\n", cpu_dai, format); return (format == SND_SOC_DAIFMT_AC97) ? 0 : -EINVAL; } /* --------------------------------------------------------------------- * ALSA SoC Bindings * * - Digital Audio Interface (DAI) template * - create/destroy dai hooks */ /** * psc_ac97_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai_ops psc_ac97_analog_ops = { .startup = mpc5200_dma_startup, .hw_params = psc_ac97_hw_analog_params, .hw_free = mpc5200_dma_hw_free, .shutdown = mpc5200_dma_shutdown, .trigger = mpc5200_dma_trigger, .set_fmt = psc_ac97_set_fmt, }; static struct snd_soc_dai_ops psc_ac97_digital_ops = { .startup = mpc5200_dma_startup, .hw_params = psc_ac97_hw_digital_params, .hw_free = mpc5200_dma_hw_free, .shutdown = mpc5200_dma_shutdown, .trigger = mpc5200_dma_trigger, .set_fmt = psc_ac97_set_fmt, }; static struct snd_soc_dai psc_ac97_dai_template[] = { { .name = "%s analog", .suspend = psc_ac97_suspend, .resume = psc_ac97_resume, .playback = { .channels_min = 1, .channels_max = 6, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .ops = &psc_ac97_analog_ops, }, { .name = "%s digital", .playback = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, }, .ops = &psc_ac97_digital_ops, }}; /* --------------------------------------------------------------------- * Sysfs attributes for debugging */ static ssize_t psc_ac97_status_show(struct device *dev, struct device_attribute *attr, char *buf) { struct psc_dma *psc_dma = dev_get_drvdata(dev); return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x " "tfnum=%i tfstat=0x%.4x\n", in_be16(&psc_dma->psc_regs->sr_csr.status), in_be32(&psc_dma->psc_regs->sicr), in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff, in_be16(&psc_dma->fifo_regs->rfstat), in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff, in_be16(&psc_dma->fifo_regs->tfstat)); } static int *psc_ac97_get_stat_attr(struct psc_dma *psc_dma, const char *name) { if (strcmp(name, "playback_underrun") == 0) return &psc_dma->stats.underrun_count; if (strcmp(name, "capture_overrun") == 0) return &psc_dma->stats.overrun_count; return NULL; } static ssize_t psc_ac97_stat_show(struct device *dev, struct device_attribute *attr, char *buf) { struct psc_dma *psc_dma = dev_get_drvdata(dev); int *attrib; attrib = psc_ac97_get_stat_attr(psc_dma, attr->attr.name); if (!attrib) return 0; return sprintf(buf, "%i\n", *attrib); } static ssize_t psc_ac97_stat_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct psc_dma *psc_dma = dev_get_drvdata(dev); int *attrib; attrib = psc_ac97_get_stat_attr(psc_dma, attr->attr.name); if (!attrib) return 0; *attrib = simple_strtoul(buf, NULL, 0); return count; } static DEVICE_ATTR(status, 0644, psc_ac97_status_show, NULL); static DEVICE_ATTR(playback_underrun, 0644, psc_ac97_stat_show, psc_ac97_stat_store); static DEVICE_ATTR(capture_overrun, 0644, psc_ac97_stat_show, psc_ac97_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: * - Probe/remove operations * - OF device match table */ static int __devinit psc_ac97_of_probe(struct of_device *op, const struct of_device_id *match) { phys_addr_t fifo; struct psc_dma *psc_dma; struct resource res; int i, nDAI, size, psc_id, irq, rc; const __be32 *prop; void __iomem *regs; /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); if (!prop || size < sizeof *prop) return -ENODEV; psc_id = be32_to_cpu(*prop); /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); if (of_address_to_resource(op->node, 0, &res)) { dev_err(&op->dev, "Missing reg property\n"); return -ENODEV; } regs = ioremap(res.start, 1 + res.end - res.start); if (!regs) { dev_err(&op->dev, "Could not map registers\n"); return -ENODEV; } /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { iounmap(regs); return -ENOMEM; } printk("psc_dma %p\n", psc_dma); spin_lock_init(&psc_dma->lock); psc_dma->irq = irq; psc_dma->psc_regs = regs; psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; psc_dma->dev = &op->dev; psc_dma->playback.psc_dma = psc_dma; psc_dma->capture.psc_dma = psc_dma; snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u AC97", psc_id+1); /* Fill out the CPU DAI structure */ nDAI = ARRAY_SIZE(psc_ac97_dai_template); nDAI = min(nDAI, SOC_OF_SIMPLE_MAX_DAI); for (i = 0; i < nDAI; i++) { memcpy(&psc_dma->dai[i], &psc_ac97_dai_template[i], sizeof(struct snd_soc_dai)); psc_dma->dai[i].private_data = psc_dma; snprintf(psc_dma->stream_name[i], PSC_STREAM_NAME_LEN, psc_ac97_dai_template[i].name, psc_dma->name); psc_dma->dai[i].name = psc_dma->stream_name[i]; psc_dma->dai[i].id = psc_id; } /* Find the address of the fifo data registers and setup the * DMA tasks */ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); psc_dma->capture.bcom_task = bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); psc_dma->playback.bcom_task = bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); iounmap(regs); kfree(psc_dma); return -ENODEV; } /* Disable all interrupts and reset the PSC */ out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX); /* reset receiver */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX); /* reset transmitter */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT); /* reset error */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1); /* reset mode */ /* Do a cold reset of codec */ out_8(&psc_dma->psc_regs->op1, MPC52xx_PSC_OP_RES); udelay(10); out_8(&psc_dma->psc_regs->op0, MPC52xx_PSC_OP_RES); udelay(50); /* Configure the serial interface mode to AC97 */ psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr); /* No slots active */ out_be32(&psc_dma->psc_regs->ac97_slots, 0x00000000); /* Set up mode register; * First write: RxRdy (FIFO Alarm) generates rx FIFO irq * Second write: register Normal mode for non loopback */ out_8(&psc_dma->psc_regs->mode, 0); out_8(&psc_dma->psc_regs->mode, 0); /* Set the TX and RX fifo alarm thresholds */ out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); out_8(&psc_dma->fifo_regs->rfcntl, 0x4); out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); out_8(&psc_dma->fifo_regs->tfcntl, 0x7); /* Go */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_TX_ENABLE); out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RX_ENABLE); /* Lookup the IRQ numbers */ psc_dma->playback.irq = bcom_get_task_irq(psc_dma->playback.bcom_task); psc_dma->capture.irq = bcom_get_task_irq(psc_dma->capture.bcom_task); /* Save what we've done so it can be found again later */ dev_set_drvdata(&op->dev, psc_dma); /* Register the SYSFS files */ rc = device_create_file(psc_dma->dev, &dev_attr_status); rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun); rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_dma->dev, "error creating sysfs files\n"); rc = snd_soc_register_dais(psc_dma->dai, nDAI); if (rc != 0) { printk("Failed to register DAI\n"); return 0; } /* Tell the ASoC OF helpers about it */ of_snd_soc_register_cpu_dai(op->node, psc_dma->dai, nDAI); return 0; } static int __devexit psc_ac97_of_remove(struct of_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); dev_dbg(&op->dev, "psc_ac97_remove()\n"); bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); iounmap(psc_dma->psc_regs); iounmap(psc_dma->fifo_regs); kfree(psc_dma); dev_set_drvdata(&op->dev, NULL); return 0; } /* Match table for of_platform binding */ static struct of_device_id psc_ac97_match[] __devinitdata = { { .compatible = "fsl,mpc5200-psc-ac97", }, {} }; MODULE_DEVICE_TABLE(of, psc_ac97_match); static struct of_platform_driver psc_ac97_driver = { .match_table = psc_ac97_match, .probe = psc_ac97_of_probe, .remove = __devexit_p(psc_ac97_of_remove), .driver = { .name = "mpc5200-psc-ac97", .owner = THIS_MODULE, }, }; /* --------------------------------------------------------------------- * Module setup and teardown; simply register the of_platform driver * for the PSC in AC97 mode. */ static int __init psc_ac97_init(void) { return of_register_platform_driver(&psc_ac97_driver); } module_init(psc_ac97_init); static void __exit psc_ac97_exit(void) { of_unregister_platform_driver(&psc_ac97_driver); } module_exit(psc_ac97_exit); -- Jon Smirl jonsmirl(a)gmail.com

5 44

[alsa-devel] Help-Pause/Resume API
by Sudeept Prusti 05 May '09

05 May '09

Hi all, Can anyone educate me which alsa-lib API used for pause and resume. I have a minimal alsa code for pcm playback and i need the needed API for test pause and resume feature on pcm playback. Best 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.

2 2

[alsa-devel] module snd-atiixp issue with suspend/hibernate/resume
by Ryan Dunn 05 May '09

05 May '09

I have a Compaq V2000 laptop with the ATI IXP chipset and make use of the snd-atiixp module. The laptop needs to have the ac97_quirk option set to 7 to enable the mute LED. After setting this option in the /etc/modprobe.d/options.conf file, the mute LED works after a reboot. However, after a resume from suspend or hibernate, the mute LED does not work. The alsa-info.sh script reports that the quirk is still set with 7. At this point a reboot will NOT fix the problem. The only way to fix it is to remove and reinsert the module with modprobe. This is on a fresh Ubuntu 9.04 install. Any ideas? I wasn't sure if this should be reported here or on the kernel list. I saw a similar issue with the ac97_quirk on the kernel list and they were referred here. I'm a software developer, so I'd be willing to try ideas/possible solutions if you have them. Thanks, Ryan

3 9

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 05 May '09

05 May '09

The following changes since commit 8560b9321f9050968f393ce1ec67e47c1a0bd5cf: Takashi Iwai (1): Merge branch 'fix/asoc' into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Jinyoung Park (1): ASoC: Fix errors in WM8990 Jonathan Cameron (1): ASoC: IMote2 ASoC Support Lopez Cruz, Misael (2): ASoC: TWL4030: Add VDL analog bypass ASoC: TWL4030: Add voice digital loopback: sidetone Mark Brown (15): ASoC: Fix error message formatting in s3c64xx-i2s driver ASoC: Check for supported CPUs when building s3c-i2s-v2 ASoC: Make S3C64xx clock export function to return struct clk ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bit ASoC: Fix boot warnings from S3C IISv2 ASoC: Allow use of resource from the platform device for S3C IISv2 ASoC: Display the clock rate used as the basis for rate calculation ASoC: Display S3C IISv2 mode and MS errors by default ASoC: Staticise txctrl and rxctrl for S3C IISv2 ASoC: Use platform device resource for S3C64xx IISv2 ASoC: Use a shared define for AC97 CODEC data formats ASoC: Remove unused DAI format defines Merge commit 'takashi/topic/asoc' into for-2.6.31 Merge branch 'for-2.6.30' into for-2.6.31 ASoC: Remove redundant codec pointer from DAIs Peter Ujfalusi (1): ASoC: TWL4030: Add VIBRA output include/sound/soc-dai.h | 28 ++------- sound/soc/codecs/ac97.c | 4 +- sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/twl4030.c | 121 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8990.c | 40 +++++++------- sound/soc/codecs/wm9705.c | 4 +- sound/soc/codecs/wm9712.c | 6 +- sound/soc/codecs/wm9713.c | 6 +- sound/soc/pxa/Kconfig | 9 +++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/imote2.c | 114 ++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 87 ++++++++++++++++++++++++---- sound/soc/s3c24xx/s3c64xx-i2s.c | 15 ++--- sound/soc/s3c24xx/s3c64xx-i2s.h | 4 +- 14 files changed, 362 insertions(+), 82 deletions(-) create mode 100644 sound/soc/pxa/imote2.c

2 19

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 05 May '09

05 May '09

The following changes since commit 6574612fbb34c63117581e68f2231ddce027e41e: Takashi Iwai (1): ASoC: Remove BROKEN from mpc5200 kconfig are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jinyoung Park (1): ASoC: Fix errors in WM8990 sound/soc/codecs/wm8990.c | 40 ++++++++++++++++++++-------------------- 1 files changed, 20 insertions(+), 20 deletions(-)

2 1

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: VIBRA output support
by Peter Ujfalusi 05 May '09

05 May '09

Hello, The VIBRA output can be driven with audio data or with local vibrator driver, the following patch adds VIBRA output support (for driving the VIBRA output with audio data) for the TWL4030 codec. --- Peter Ujfalusi (1): ASoC: TWL4030: Add VIBRA output sound/soc/codecs/twl4030.c | 61 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 61 insertions(+), 0 deletions(-)

2 3

[alsa-devel] [ASoC PATCH 1/1] IMote 2 ASoC Support
by Jonathan Cameron 04 May '09

04 May '09

From: Jonathan Cameron <jic23(a)cam.ac.uk> IMote2 ASoC Support This patch adds the ASoC side of the board support for the Crossbow IMB400 daughter board. Thanks to Crossbow for considerable assistance. Signed-off-by: Jonathan Cameron <jic23(a)cam.ac.uk> --- Relatively straight forward configuration with the PXA acting as the clock master. sound/soc/pxa/Kconfig | 9 +++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/imote2.c | 131 +++++++++++++++++++++++++++++++++++++++++++++++ diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..eb75a1c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701 help Say Y if you want to add support for SoC audio on the MIO A701. + +config SND_PXA2XX_SOC_IMOTE2 + tristate "SoC Audio support for IMote 2" + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 + select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8940 + help + Say Y if you want to add support for SoC audio on the + IMote 2. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 4b90c3c..6e096b4 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o +snd-soc-imote2-objs := imote2.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -35,3 +36,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o +obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c new file mode 100644 index 0000000..8476151 --- /dev/null +++ b/sound/soc/pxa/imote2.c @@ -0,0 +1,131 @@ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/pxa27x.h> +#include <mach/hardware.h> + +#include <mach/audio.h> + +#include "../codecs/wm8940.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-pcm.h" + +static int imote2_asoc_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; + unsigned int clk = 0; + int ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + /* 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; + + /* CPU should be clock master */ + 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; + + /* Disable the PLL */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, 0); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set the I2S system clock as input (unused) */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk, + SND_SOC_CLOCK_OUT); + + return ret; +} + +static struct snd_soc_ops imote2_asoc_ops = { + .hw_params = imote2_asoc_hw_params, +}; + +static struct snd_soc_dai_link imote2_dai = { + .name = "WM8940", + .stream_name = "WM8940", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &wm8940_dai, + .ops = &imote2_asoc_ops, +}; + +static struct snd_soc_card snd_soc_imote2 = { + .name = "Imote2", + .platform = &pxa2xx_soc_platform, + .dai_link = &imote2_dai, + .num_links = 1, +}; + +static struct snd_soc_device imote2_snd_devdata = { + .card = &snd_soc_imote2, + .codec_dev = &soc_codec_dev_wm8940, +}; + +static struct platform_device *imote2_snd_device; + +static int __init imote2_asoc_init(void) +{ + int ret; + + if (!machine_is_intelmote2()) + return -ENODEV; + imote2_snd_device = platform_device_alloc("soc-audio", -1); + if (!imote2_snd_device) + return -ENOMEM; + + platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata); + imote2_snd_devdata.dev = &imote2_snd_device->dev; + ret = platform_device_add(imote2_snd_device); + if (ret) + platform_device_put(imote2_snd_device); + + return ret; +} +module_init(imote2_asoc_init); + +static void __exit imote2_asoc_exit(void) +{ + platform_device_unregister(imote2_snd_device); +} +module_exit(imote2_asoc_exit); + +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_DESCRIPTION("ALSA SoC Imote 2"); +MODULE_LICENSE("GPL");

2 4

[alsa-devel] [PATCH 1/2] ASoC: TWL4030: Add VDL analog bypass
by Lopez Cruz, Misael 04 May '09

04 May '09

This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++------ 1 files changed, 31 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80..efb371f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -468,6 +468,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); +/* Analog bypass for Voice */ +static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); + /* Digital bypass gain, 0 mutes the bypass */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -585,7 +589,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -597,14 +601,28 @@ static int bypass_event(struct snd_soc_dapm_widget *w, else twl4030->bypass_state &= ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { + /* Analog voice bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= (1 << 4); + else + twl4030->bypass_state &= ~(1 << 4); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0x1F) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -935,7 +953,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - TWL4030_REG_AVDAC_CTL, 4, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -962,6 +980,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassl2_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control, + bypass_event, SND_SOC_DAPM_POST_REG), /* Digital bypasses */ SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, @@ -979,6 +1000,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, + 4, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1067,13 +1090,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - - {"VDL_APGA", NULL, "DAC Voice"}, + {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -1169,11 +1192,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"}, /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, -- 1.5.4.3

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Re: [alsa-devel] realtek ALC272 support
by Andres Salomon 04 May '09

04 May '09

On Tue, 28 Apr 2009 15:41:08 -0400 Andres Salomon <dilinger(a)queued.net> wrote: > On Mon, 27 Apr 2009 18:08:21 +0200 > Takashi Iwai <tiwai(a)suse.de> wrote: > [...] > > > > > > > > At Fri, 24 Apr 2009 15:24:15 -0400, > > > > > > > > Andres Salomon wrote: > > > > > > > > > > > > > > > > > > Hi Kailang, > > > > > > > > > > > > > > > > > > I noticed that your name was on the ALC272 support > > > > > > > > > patch for ALSA intel-hda. This patch basically sets > > > > > > > > > ALC272 to use (mostly) the same code as ALC662. I > > > > > > > > > have two machines that have ALC272, and both of them > > > > > > > > > need verb tables in order to function. I'm wondering > > > > > > > > > if ALC662 should really be used.. > > > > > > > > > > > > > > > > > > Here's one - > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > This isn't the final version of the patch (there are > > > > > > > > > further commits I made in order to support headphone > > > > > > > > > mic stuff), but it gives you an idea of the codec > > > > > > > > > values. The other is: > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > All of these leave me wonder if there's a specific > > > > > > > > > patch_alc272 function that could be written to rid > > > > > > > > > ourselves of these specific quirks. Are there > > > > > > > > > machines with ALC272 that are functional with the > > > > > > > > > current ALSA code? > > > > > > > > [...] > > > > Could you try sound-unstable tree a bit later again? > > I found a bug in my patch, and fixed and updated GIT tree now. > > At least, the headphone plugging should work now. > > > > The mic auto-detection still doesn't work with model=auto, though. > > So, I'm going to take your patch anyway later. But I just wanted > > to be sure that the current tree could work somehow better... > > > > Hi, > > I just updated and tried the current tree; still no sound/headphone > output. :( Ok, I believe I've made some progress on this. The problem appears to be related to the autoconfig handling of the line out nids. The current code ends up with something like the following: ALSA sound/pci/hda/hda_codec.c:3833: autoconfig: line_outs=1 (0x17/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3837: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3841: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3842: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3853: inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 However, NID 0x17 is actually a speaker_out. The code that checks for line_outs in snd_hda_parse_pin_def_config() unsets the speaker_out and uses that NID for a line_out. For whatever reason, this breaks things (no sound output, no headphone out). According to the ALC272 data sheet (the block diagram on page 5), there are actually 2 line_out NIDs; 0x14 and 0x15. The default pin configs for these two NIDs (0x411111f0 for both) have the Port Connectivity field set to AC_JACK_PORT_NONE, so the auto-config skips those NIDs. If I provide an override to a) set Port Connectivity to AC_JACK_PORT_COMPLEX, and b) set Default Device to AC_JACK_LINE_OUT (from AC_JACK_SPEAKER), I get usable audio (speakers work, headphone works, headphone plugging mutes the speaker properly, and even the mic works). The patch which I've used is below. Clearly, this isn't a very nice solution, so the question is: is there a way to provide an override/quirk for pincfgs (I see one for HWDEP, but that's configuration-specific) within the current hda_codec framework for this specific codec/laptop? Alternatively, I can try and figure out what's wrong with the code in the case where line_out_pins[0] is set to the speaker_out_pins[0] (ie, NID 0x17 in this case). I can check the other ALC272 machine tomorrow (the DELL_ZM1) to see if the line_out hack enables it to work without the verb table. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b91f6ed..5aa7a0c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3644,6 +3644,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("nid %x: pincfg def_conf=%x\n", nid, def_conf); + if (codec->vendor_id == 0x10ec0272 && + (nid == 0x14 || nid == 0x15)) { + def_conf &= ~(0x3 << AC_DEFCFG_PORT_CONN_SHIFT); + def_conf &= ~(0xf << AC_DEFCFG_DEVICE_SHIFT); + snd_printd("nid %x: overriding def_conf, now %x\n", nid, + def_conf); + } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf);

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[alsa-devel] Adding MMAP support to alsa driver
by Gabriel Johnson 04 May '09

04 May '09

What structures need to be filled out in the hardware driver to support MMAP calls?

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