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[alsa-devel] Playing multiple sounds at the same time
by Andreas Falkenhahn 04 May '10

04 May '10

Hi, I'm new to ALSA programming so please bear with me if the following question is all too obvious. I did some research on my question before but I didn't come across a clear answer. What I want to do is really simple: Playing multiple sounds in different formats at the same time. E.g. Sound 1: 12345 hz, 8-bit signed, mono Sound 2: 44100 hz, 16-bit unsigned, stereo Sound 3: 8000 hz, 8-bit unsigned, stereo Now I want to play sounds 1 to 3 at the same time. My question is now: Can ALSA do automatic mixing for me here or do I have to do the mixing on my own? In other words, can I simply feed all these three sound streams to ALSA or do I have to mix them into a fourth audio stream first and feeding this new fourth stream to ALSA then? Thanks, Andreas

3 4

[alsa-devel] safe support for rewind in ALSA
by pl bossart 04 May '10

04 May '10

All, I'd like to reopen a thread on concerns with the current implementation of rewind in ALSA. This was already discussed on the mailing list last year without any progress. (see http://thread.gmane.org/gmane.linux.alsa.devel/65759/focus=65773 and http://article.gmane.org/gmane.linux.alsa.devel/66902) The problem is that snd_pcm_rewind() allows in theory to go back to the next sample to be rendered. This however isn't possible on any existing hardware. DMAs do prefetch samples ahead of time, and output interfaces have FIFOs on top of this. This isn't too bad in the PC world with HDAudio, since the prefetching is limited. However in the embedded space it has become a headache. To reduce power consumption, most of the recent embedded interfaces do embed some local internal memory (5kB for TI's McBSP3, a lot more for others), and we should provide a means for driver developers to set a bound on how much can be rewound safely when power-saving features are enabled. Note that there already are a set of definitions for FIFOs, but they are either obsolete or no one uses them (actually HDAudio makes use of this value internally in azx_dev but doesn't export it). My suggestion would be to add a 'fifo' parameter to the runtime structure (so that it's visible to the core), let the driver developer fill the value in the .open routine, and use this parameter in rewind to prevent rewinding too much. This parameter would include both hardware FIFOs and DMA buffers. Comments welcome. - Pierre

7 41

[alsa-devel] asoc: fsl_dma_hw_free being called twice at end of playback
by Timur Tabi 04 May '10

04 May '10

In my 8610 ASoC driver, I'm seeing the DMA driver's .hw_free function being called twice whenever playback ends. I've added some printk's to demonstrate: snd_pcm_common_ioctl1:2543 snd_pcm_hw_free:543 soc_pcm_hw_free:761 c023983c fsl_dma_hw_free:720 substream=df029d00 dma_private=df440000 stream=0 snd_pcm_release_substream:2022 soc_pcm_hw_free:761 c023983c fsl_dma_hw_free:720 substream=df029d00 dma_private=df440000 stream=0 fsl_dma_close:759 substream=df029d00 dma_private=df440000 So the ioctl is being called only once, but the both snd_pcm_hw_free() and snd_pcm_release_substream() are calling the .hw_free function. Is this normal? I don't know how to handle this, other than to add some boolean to my private data structure that indicates whether the resources have been freed or not. -- Timur Tabi Linux kernel developer at Freescale

3 3

[alsa-devel] [PATCH 2/2] [v2] powerpc/alsa: update MPC8610 HPCD audio drivers to support multiple components
by Timur Tabi 04 May '10

04 May '10

Update the Freescale MPC8610 HPCD audio drivers to support the new multiple- component architecture in ASoC. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 299 ++++++----- sound/soc/fsl/fsl_dma.h | 19 +- sound/soc/fsl/fsl_ssi.c | 211 +++++---- sound/soc/fsl/fsl_ssi.h | 25 - sound/soc/fsl/mpc8610_hpcd.c | 1174 ++++++++++++++++++++---------------------- 5 files changed, 844 insertions(+), 884 deletions(-) rewrite sound/soc/fsl/mpc8610_hpcd.c (62%) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 857e9f7..ff0064c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -3,10 +3,11 @@ * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. * * This driver implements ASoC support for the Elo DMA controller, which is * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, @@ -15,10 +16,11 @@ #include <linux/module.h> #include <linux/init.h> -#include <linux/platform_device.h> #include <linux/dma-mapping.h> #include <linux/interrupt.h> #include <linux/delay.h> +#include <linux/of_platform.h> +#include <linux/list.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,26 +53,16 @@ #define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ SNDRV_PCM_RATE_CONTINUOUS) -/* DMA global data. This structure is used by fsl_dma_open() to determine - * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does - * not allow the machine driver to provide this information to the PCM - * driver in advance, and there's no way to differentiate between the two - * DMA controllers. So for now, this driver only supports one SSI device - * using two DMA channels. We cannot support multiple DMA devices. - * - * ssi_stx_phys: bus address of SSI STX register - * ssi_srx_phys: bus address of SSI SRX register - * dma_channel: pointer to the DMA channel's registers - * irq: IRQ for this DMA channel - * assigned: set to 1 if that DMA channel is assigned to a substream - */ -static struct { +struct dma_object { + struct list_head list; + struct snd_soc_platform_driver dai; dma_addr_t ssi_stx_phys; dma_addr_t ssi_srx_phys; - struct ccsr_dma_channel __iomem *dma_channel[2]; - unsigned int irq[2]; - unsigned int assigned[2]; -} dma_global_data; + struct ccsr_dma_channel __iomem *channel; + unsigned int irq; + bool assigned; + char path[1]; +}; /* * The number of DMA links to use. Two is the bare minimum, but if you @@ -87,8 +79,6 @@ static struct { * structure. * * @link[]: array of link descriptors - * @controller_id: which DMA controller (0, 1, ...) - * @channel_id: which DMA channel on the controller (0, 1, 2, ...) * @dma_channel: pointer to the DMA channel's registers * @irq: IRQ for this DMA channel * @substream: pointer to the substream object, needed by the ISR @@ -103,8 +93,6 @@ static struct { */ struct fsl_dma_private { struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; - unsigned int controller_id; - unsigned int channel_id; struct ccsr_dma_channel __iomem *dma_channel; unsigned int irq; struct snd_pcm_substream *substream; @@ -211,6 +199,9 @@ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) static irqreturn_t fsl_dma_isr(int irq, void *dev_id) { struct fsl_dma_private *dma_private = dev_id; + struct snd_pcm_substream *substream = dma_private->substream; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; irqreturn_t ret = IRQ_NONE; u32 sr, sr2 = 0; @@ -221,11 +212,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) sr = in_be32(&dma_channel->sr); if (sr & CCSR_DMA_SR_TE) { - dev_err(dma_private->substream->pcm->card->dev, - "DMA transmit error (controller=%u channel=%u irq=%u\n", - dma_private->controller_id, - dma_private->channel_id, irq); - fsl_dma_abort_stream(dma_private->substream); + dev_err(dev, "dma transmit error\n"); + fsl_dma_abort_stream(substream); sr2 |= CCSR_DMA_SR_TE; ret = IRQ_HANDLED; } @@ -234,11 +222,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) ret = IRQ_HANDLED; if (sr & CCSR_DMA_SR_PE) { - dev_err(dma_private->substream->pcm->card->dev, - "DMA%u programming error (channel=%u irq=%u)\n", - dma_private->controller_id, - dma_private->channel_id, irq); - fsl_dma_abort_stream(dma_private->substream); + dev_err(dev, "dma programming error\n"); + fsl_dma_abort_stream(substream); sr2 |= CCSR_DMA_SR_PE; ret = IRQ_HANDLED; } @@ -252,8 +237,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) ret = IRQ_HANDLED; if (sr & CCSR_DMA_SR_EOSI) { - struct snd_pcm_substream *substream = dma_private->substream; - /* Tell ALSA we completed a period. */ snd_pcm_period_elapsed(substream); @@ -304,10 +287,8 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { - dev_err(card->dev, - "Can't allocate playback DMA buffer (size=%u)\n", - fsl_dma_hardware.buffer_bytes_max); - return -ENOMEM; + dev_err(card->dev, "can't allocate playback dma buffer\n"); + return ret; } ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, @@ -315,10 +296,8 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, &pcm->streams[1].substream->dma_buffer); if (ret) { snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - dev_err(card->dev, - "Can't allocate capture DMA buffer (size=%u)\n", - fsl_dma_hardware.buffer_bytes_max); - return -ENOMEM; + dev_err(card->dev, "can't allocate capture dma buffer\n"); + return ret; } return 0; @@ -389,6 +368,10 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, static int fsl_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_object *dma = + container_of(rtd->platform->driver, struct dma_object, dai); struct fsl_dma_private *dma_private; struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; @@ -406,52 +389,44 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { - dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + dev_err(dev, "invalid buffer size\n"); return ret; } channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - if (dma_global_data.assigned[channel]) { - dev_err(substream->pcm->card->dev, - "DMA channel already assigned\n"); + if (dma->assigned) { + dev_err(dev, "dma channel already assigned\n"); return -EBUSY; } - dma_private = dma_alloc_coherent(substream->pcm->card->dev, - sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); + dma_private = dma_alloc_coherent(dev, sizeof(struct fsl_dma_private), + &ld_buf_phys, GFP_KERNEL); if (!dma_private) { - dev_err(substream->pcm->card->dev, - "can't allocate DMA private data\n"); + dev_err(dev, "can't allocate dma private data\n"); return -ENOMEM; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys; + dma_private->ssi_sxx_phys = dma->ssi_stx_phys; else - dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys; + dma_private->ssi_sxx_phys = dma->ssi_srx_phys; - dma_private->dma_channel = dma_global_data.dma_channel[channel]; - dma_private->irq = dma_global_data.irq[channel]; + dma_private->dma_channel = dma->channel; + dma_private->irq = dma->irq; dma_private->substream = substream; dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - /* We only support one DMA controller for now */ - dma_private->controller_id = 0; - dma_private->channel_id = channel; - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); if (ret) { - dev_err(substream->pcm->card->dev, - "can't register ISR for IRQ %u (ret=%i)\n", + dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); - dma_free_coherent(substream->pcm->card->dev, - sizeof(struct fsl_dma_private), + dma_free_coherent(dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); return ret; } - dma_global_data.assigned[channel] = 1; + dma->assigned = 1; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); @@ -545,6 +520,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; /* Number of bits per sample */ unsigned int sample_size = @@ -605,8 +582,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, break; default: /* We should never get here */ - dev_err(substream->pcm->card->dev, - "unsupported sample size %u\n", sample_size); + dev_err(dev, "unsupported sample size %u\n", sample_size); return -EINVAL; } @@ -688,6 +664,8 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; dma_addr_t position; snd_pcm_uframes_t frames; @@ -709,8 +687,7 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) if ((position < dma_private->dma_buf_phys) || (position > dma_private->dma_buf_end)) { - dev_err(substream->pcm->card->dev, - "dma pointer is out of range, halting stream\n"); + dev_err(dev, "dma pointer is out of range, halting stream\n"); return SNDRV_PCM_POS_XRUN; } @@ -771,26 +748,28 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_object *dma = + container_of(rtd->platform->driver, struct dma_object, dai); if (dma_private) { if (dma_private->irq) free_irq(dma_private->irq, dma_private); if (dma_private->ld_buf_phys) { - dma_unmap_single(substream->pcm->card->dev, - dma_private->ld_buf_phys, - sizeof(dma_private->link), DMA_TO_DEVICE); + dma_unmap_single(dev, dma_private->ld_buf_phys, + sizeof(dma_private->link), + DMA_TO_DEVICE); } /* Deallocate the fsl_dma_private structure */ - dma_free_coherent(substream->pcm->card->dev, - sizeof(struct fsl_dma_private), - dma_private, dma_private->ld_buf_phys); + dma_free_coherent(dev, sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); substream->runtime->private_data = NULL; } - dma_global_data.assigned[dir] = 0; + dma->assigned = 0; return 0; } @@ -813,6 +792,41 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) } } +/* List of DMA nodes that we've probed */ +static LIST_HEAD(dma_list); + +/** + * fsl_soc_dma_to_dai - returning the DAI struct for a given DMA node + * + * When a machine driver registers itself with ASoC, it must provide the + * address of the DAI structures that it wants to connect. Most drivers simply + * reference global data structures in the various drivers, but we PowerPC + * developers are better than that. This driver keeps a list of every DMA + * node that it probes, and when the machine driver wants to reference a + * DMA DAI, this function provides that service. + * + * Hopefully one day, ASoC will just want the name of the DMA device, not + * some obscure pointer, and this function will go away. + */ +struct snd_soc_platform_driver *fsl_soc_dma_to_dai(const char *path, + dma_addr_t ssi_stx_phys, dma_addr_t ssi_srx_phys) +{ + struct list_head *ptr; + struct dma_object *dma; + + list_for_each(ptr, &dma_list) { + dma = list_entry(ptr, struct dma_object, list); + if (strcmp(path, dma->path) == 0) { + dma->ssi_stx_phys = ssi_stx_phys; + dma->ssi_srx_phys = ssi_srx_phys; + return &dma->dai; + } + } + + return NULL; +} +EXPORT_SYMBOL(fsl_soc_dma_to_dai); + static struct snd_pcm_ops fsl_dma_ops = { .open = fsl_dma_open, .close = fsl_dma_close, @@ -822,80 +836,91 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -struct snd_soc_platform_driver fsl_soc_platform = { - .name = "fsl-dma", - .ops = &fsl_dma_ops, - .pcm_new = fsl_dma_new, - .pcm_free = fsl_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(fsl_soc_platform); - -/** - * fsl_dma_configure: store the DMA parameters from the fabric driver. - * - * This function is called by the ASoC fabric driver to give us the DMA and - * SSI channel information. - * - * Unfortunately, ASoC V1 does make it possible to determine the DMA/SSI - * data when a substream is created, so for now we need to store this data - * into a global variable. This means that we can only support one DMA - * controller, and hence only one SSI. - */ -int fsl_dma_configure(struct fsl_dma_info *dma_info) +static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, + const struct of_device_id *match) { - static int initialized; + struct dma_object *dma; + struct device_node *np = of_dev->node; + int ret; - /* We only support one DMA controller for now */ - if (initialized) - return 0; + dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + if (!dma) { + dev_err(&of_dev->dev, "could not allocate dma object\n"); + return -ENOMEM; + } - dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys; - dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys; - dma_global_data.dma_channel[0] = dma_info->dma_channel[0]; - dma_global_data.dma_channel[1] = dma_info->dma_channel[1]; - dma_global_data.irq[0] = dma_info->dma_irq[0]; - dma_global_data.irq[1] = dma_info->dma_irq[1]; - dma_global_data.assigned[0] = 0; - dma_global_data.assigned[1] = 0; - - initialized = 1; - return 1; -} -EXPORT_SYMBOL_GPL(fsl_dma_configure); + strcpy(dma->path, np->full_name); + dma->dai.name = dma->path; + dma->dai.ops = &fsl_dma_ops; + dma->dai.pcm_new = fsl_dma_new; + dma->dai.pcm_free = fsl_dma_free_dma_buffers; -static int __devinit fsl_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, -1, &fsl_soc_platform); + ret = snd_soc_register_platform(&of_dev->dev, 0, &dma->dai); + if (ret) { + dev_err(&of_dev->dev, "could not register platform\n"); + kfree(dma); + return ret; + } + + dma->channel = of_iomap(np, 0); + dma->irq = irq_of_parse_and_map(np, 0); + list_add(&dma->list, &dma_list); + + return 0; } -static int __devexit fsl_soc_platform_remove(struct platform_device *pdev) +static int __devexit fsl_soc_dma_remove(struct of_device *of_dev) { - snd_soc_unregister_platform(&pdev->dev, -1); + struct list_head *n, *ptr; + struct dma_object *dma; + + list_for_each_safe(ptr, n, &dma_list) { + dma = list_entry(ptr, struct dma_object, list); + list_del_init(ptr); + + snd_soc_unregister_platform(&of_dev->dev, 0); + iounmap(dma->channel); + irq_dispose_mapping(dma->irq); + kfree(dma); + } + return 0; } -static struct platform_driver fsl_pcm_driver = { - .driver = { - .name = "fsl-pcm-audio", - .owner = THIS_MODULE, - }, - .probe = fsl_soc_platform_probe, - .remove = __devexit_p(fsl_soc_platform_remove), +static const struct of_device_id fsl_soc_dma_ids[] = { + { .compatible = "fsl,ssi-dma-channel", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids); + +static struct of_platform_driver fsl_soc_dma_driver = { + .name = "fsl-soc-dma-dai", + .match_table = fsl_soc_dma_ids, + .probe = fsl_soc_dma_probe, + .remove = __devexit_p(fsl_soc_dma_remove), }; -static int __init snd_fsl_pcm_init(void) +static int __init fsl_soc_dma_init(void) { - return platform_driver_register(&fsl_pcm_driver); + pr_info("Freescale Elo DMA ASoC PCM Driver\n"); + + return of_register_platform_driver(&fsl_soc_dma_driver); } -module_init(snd_fsl_pcm_init); -static void __exit snd_fsl_pcm_exit(void) +static void __exit fsl_soc_dma_exit(void) { - platform_driver_unregister(&fsl_pcm_driver); + of_unregister_platform_driver(&fsl_soc_dma_driver); } -module_exit(snd_fsl_pcm_exit); + +/* We want the DMA driver to be initialized before the SSI driver, so that + * when the SSI driver calls fsl_soc_dma_dai_from_node(), the DMA driver + * will already have been probed. The easiest way to do that is to make the + * __init function called via arch_initcall(). + */ +arch_initcall(fsl_soc_dma_init); +module_exit(fsl_soc_dma_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module"); -MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 253bd3c..88c686d 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,24 +126,9 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_dai object - * - * ssi_stx_phys: bus address of SSI STX register to use - * ssi_srx_phys: bus address of SSI SRX register to use - * dma[0]: points to the DMA channel to use for playback - * dma[1]: points to the DMA channel to use for capture - * dma_irq[0]: IRQ of the DMA channel to use for playback - * dma_irq[1]: IRQ of the DMA channel to use for capture - */ -struct fsl_dma_info { - dma_addr_t ssi_stx_phys; - dma_addr_t ssi_srx_phys; - struct ccsr_dma_channel __iomem *dma_channel[2]; - unsigned int dma_irq[2]; -}; - extern struct snd_soc_platform_driver fsl_soc_platform; -int fsl_dma_configure(struct fsl_dma_info *dma_info); +struct snd_soc_platform_driver *fsl_soc_dma_to_dai(const char *path, + dma_addr_t ssi_stx_phys, dma_addr_t ssi_srx_phys); #endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 780e243..56e66c2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -3,10 +3,11 @@ * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. */ #include <linux/init.h> @@ -14,6 +15,7 @@ #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/of_platform.h> #include <sound/core.h> #include <sound/pcm.h> @@ -70,33 +72,31 @@ /** * fsl_ssi_private: per-SSI private data * - * @name: short name for this device ("SSI0", "SSI1", etc) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI * @first_stream: pointer to the stream that was opened first * @second_stream: pointer to second stream - * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics + * @name: name for this device */ struct fsl_ssi_private { - char name[8]; struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; struct snd_pcm_substream *first_stream; struct snd_pcm_substream *second_stream; - struct device *dev; unsigned int playback; unsigned int capture; int asynchronous; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; + struct platform_device *pdev; struct { unsigned int rfrc; @@ -121,6 +121,8 @@ struct fsl_ssi_private { unsigned int tfe1; unsigned int tfe0; } stats; + + char name[1]; }; /** @@ -289,8 +291,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct ccsr_ssi __iomem *ssi = ssi_private->ssi; int ret; + /* The 'name' should not have any slashes in it. */ ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); + strrchr(ssi_private->name, '/') + 1, + ssi_private); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not claim irq %u\n", ssi_private->irq); @@ -522,56 +526,15 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, } } -/** - * fsl_ssi_set_sysclk: set the clock frequency and direction - * - * This function is called by the machine driver to tell us what the clock - * frequency and direction are. - * - * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), - * and we don't care about the frequency. Return an error if the direction - * is not SND_SOC_CLOCK_IN. - * - * @clk_id: reserved, should be zero - * @freq: the frequency of the given clock ID, currently ignored - * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) - */ -static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - - return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; -} - -/** - * fsl_ssi_set_fmt: set the serial format. - * - * This function is called by the machine driver to tell us what serial - * format to use. - * - * Currently, we only support I2S mode. Return an error if the format is - * not SND_SOC_DAIFMT_I2S. - * - * @format: one of SND_SOC_DAIFMT_xxx - */ -static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) -{ - return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; -} - -/** - * fsl_ssi_dai_template: template CPU DAI for the SSI - */ static struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, }; -struct snd_soc_dai_driver fsl_ssi_dai = { +/* Template for the CPU dai driver structure */ +static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -587,7 +550,6 @@ struct snd_soc_dai_driver fsl_ssi_dai = { }, .ops = &fsl_ssi_dai_ops, }; -EXPORT_SYMBOL_GPL(fsl_ssi_dai); /* Show the statistics of a flag only if its interrupt is enabled. The * compiler will optimze this code to a no-op if the interrupt is not @@ -639,92 +601,163 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, return length; } -static int fsl_ssi_probe(struct platform_device *pdev) +/** + * Make every character in a string lower-case + */ +static void make_lowercase(char *s) +{ + char *p = s; + char c; + + while ((c = *p)) { + if ((c >= 'A') && (c <= 'Z')) + *p = c + ('a' - 'A'); + p++; + } +} + +static int __devinit fsl_ssi_probe(struct of_device *of_dev, + const struct of_device_id *match) { - struct fsl_ssi_info *ssi_info = dev_get_platdata(&pdev->dev); struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; + struct device_node *np = of_dev->node; + const char *sprop; + struct resource res; + char name[64]; + struct platform_device *pdev; + void *platform_data; + + /* We are only interested in SSIs with a codec phandle in them, so let's + * make sure this SSI has one. + */ + if (!of_get_property(np, "codec-handle", NULL)) + return -ENODEV; + + /* We only support the SSI in "I2S Slave" mode */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop || strcmp(sprop, "i2s-slave")) { + dev_notice(&of_dev->dev, "mode %s is unsupported\n", sprop); + return -ENODEV; + } - ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL); + ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + + strlen(np->full_name), GFP_KERNEL); if (!ssi_private) { - dev_err(ssi_info->dev, "could not allocate DAI object\n"); + dev_err(&of_dev->dev, "could not allocate DAI object\n"); return -ENOMEM; } - dev_attr = &ssi_private->dev_attr; + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, + sizeof(fsl_ssi_dai_template)); + ssi_private->cpu_dai_drv.name = ssi_private->name; + + /* The name we register with ASoC is the full node name */ + strlcpy(ssi_private->name, np->full_name, sizeof(ssi_private->name)); + + /* Get the addresses and IRQ */ + ret = of_address_to_resource(np, 0, &res); + if (ret) { + dev_err(&of_dev->dev, "could not determine device resources\n"); + kfree(ssi_private); + return ret; + } + ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start); + ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); - sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id); - ssi_private->ssi = ssi_info->ssi; - ssi_private->ssi_phys = ssi_info->ssi_phys; - ssi_private->irq = ssi_info->irq; - ssi_private->dev = ssi_info->dev; - ssi_private->asynchronous = ssi_info->asynchronous; + /* Are the RX and the TX clocks locked? */ + if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) + ssi_private->asynchronous = 1; + else + ssi_private->cpu_dai_drv.symmetric_rates = 1; /* Initialize the the device_attribute structure */ - dev_attr->attr.name = "ssi-stats"; + dev_attr = &ssi_private->dev_attr; + dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; - ret = device_create_file(ssi_private->dev, dev_attr); + ret = device_create_file(&of_dev->dev, dev_attr); if (ret) { - dev_err(ssi_info->dev, "could not create sysfs %s file\n", + dev_err(&of_dev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); kfree(ssi_private); return ret; } - dev_set_drvdata(ssi_private->dev, ssi_private); - fsl_ssi_dai.name = ssi_private->name; - fsl_ssi_dai.symmetric_rates = 1; + /* Register with ASoC */ + dev_set_drvdata(&of_dev->dev, ssi_private); - ret = snd_soc_register_dai(ssi_private->dev, ssi_info->id, &fsl_ssi_dai); + ret = snd_soc_register_dai(&of_dev->dev, 0, &ssi_private->cpu_dai_drv); if (ret != 0) { - dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret); + dev_err(&of_dev->dev, "failed to register DAI: %d\n", ret); kfree(ssi_private); return ret; } - return ret; + /* Trigger the machine driver's probe function. The platform driver + * name of the machine driver is taken from the /model node of the + * device tree. We also pass the address of the CPU DAI driver + * structure. + */ + sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); + snprintf(name, sizeof(name), "snd-soc-%s", sprop); + make_lowercase(name); + + platform_data = &ssi_private->cpu_dai_drv; + pdev = platform_device_register_data(&of_dev->dev, name, -1, + &platform_data, sizeof(void *)); + + ssi_private->pdev = pdev; + + return 0; } /** * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * - * This function undoes the operations of fsl_ssi_create_dai() + * This function undoes the operations of fsl_ssi_probe() */ -static int fsl_ssi_remove(struct platform_device *pdev) +static int fsl_ssi_remove(struct of_device *of_dev) { - struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - struct fsl_ssi_info *ssi_info = dev_get_platdata(&pdev->dev); - - device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + struct fsl_ssi_private *ssi_private = dev_get_drvdata(&of_dev->dev); - snd_soc_unregister_dai(&pdev->dev, ssi_info->id); + platform_device_unregister(ssi_private->pdev); + snd_soc_unregister_dai(&of_dev->dev, 0); + device_remove_file(&of_dev->dev, &ssi_private->dev_attr); kfree(ssi_private); + dev_set_drvdata(&of_dev->dev, NULL); + return 0; } -static struct platform_driver fsl_ssi_driver = { +static const struct of_device_id fsl_ssi_ids[] = { + { .compatible = "fsl,mpc8610-ssi", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_ssi_ids); + +static struct of_platform_driver fsl_ssi_driver = { + .name = "fsl-ssi-dai", + .match_table = fsl_ssi_ids, .probe = fsl_ssi_probe, - .remove = __devexit_p(fsl_ssi_remove), - .driver = { - .name = "fsl-ssi-dai", - .owner = THIS_MODULE, - }, + .remove = fsl_ssi_remove, }; static int __init fsl_ssi_init(void) { printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); - return platform_driver_register(&fsl_ssi_driver); + return of_register_platform_driver(&fsl_ssi_driver); } static void __exit fsl_ssi_exit(void) { - platform_driver_unregister(&fsl_ssi_driver); + of_unregister_platform_driver(&fsl_ssi_driver); } module_init(fsl_ssi_init); @@ -732,4 +765,4 @@ module_exit(fsl_ssi_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 965fd86..2173000 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -196,30 +196,5 @@ struct ccsr_ssi { #define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) #define CCSR_SSI_SOR_SYNRST 0x00000001 -/* Instantiation data for an SSI interface - * - * This structure contains all the information that the the SSI driver needs - * to instantiate an SSI interface with ALSA. The machine driver should - * create this structure, fill it in, call fsl_ssi_create_dai(), and then - * delete the structure. - * - * id: which SSI this is (0, 1, etc. ) - * ssi: pointer to the SSI's registers - * ssi_phys: physical address of the SSI registers - * irq: IRQ of this SSI - * dev: struct device, used to create the sysfs statistics file - * asynchronous: 0=synchronous mode, 1=asynchronous mode -*/ -struct fsl_ssi_info { - unsigned int id; - struct ccsr_ssi __iomem *ssi; - dma_addr_t ssi_phys; - unsigned int irq; - struct device *dev; - int asynchronous; -}; - -extern struct snd_soc_dai_driver fsl_ssi_dai; - #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c dissimilarity index 62% index ad8d763..3e422f7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,616 +1,558 @@ -/** - * Freescale MPC8610HPCD ALSA SoC Fabric driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. - */ - -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> -#include <sound/soc.h> -#include <asm/immap_86xx.h> - -#include "../codecs/cs4270.h" -#include "fsl_dma.h" -#include "fsl_ssi.h" - -/** - * mpc8610_hpcd_data: fabric-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * MPC8610 HPCD. Some of the data is taken from the device tree. - */ -struct mpc8610_hpcd_data { - struct snd_soc_dai_link dai; - struct snd_soc_card machine; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - struct ccsr_guts __iomem *guts; - struct ccsr_ssi __iomem *ssi; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int ssi_irq; - unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_irq[2]; - struct ccsr_dma_channel __iomem *dma[2]; - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ -}; - -/** - * mpc8610_hpcd_machine_probe: initalize the board - * - * This function is called when platform_device_add() is called. It is used - * to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) -{ - struct mpc8610_hpcd_data *machine_data = - sound_device->dev.platform_data; - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[0], 0); - guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; - } - - return 0; -} - -/** - * mpc8610_hpcd_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct mpc8610_hpcd_data *machine_data = - rtd->card->dev->platform_data; - int ret = 0; - - /* Tell the CPU driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the CPU driver what the clock frequency is, and whether it's a - * slave or master. - */ - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * mpc8610_hpcd_machine_remove: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) -{ - struct mpc8610_hpcd_data *machine_data = - sound_device->dev.platform_data; - - /* Restore the signal routing */ - - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[0], 0); - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); - break; - } - - return 0; -} - -/** - * mpc8610_hpcd_ops: ASoC fabric driver operations - */ -static struct snd_soc_ops mpc8610_hpcd_ops = { - .startup = mpc8610_hpcd_startup, -}; - -/** - * mpc8610_hpcd_probe: OF probe function for the fabric driver - * - * This function gets called when an SSI node is found in the device tree. - * - * Although this is a fabric driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - * - * FIXME: Currently, we only support one DMA controller, so if there are - * multiple SSI nodes with codecs, only the first will be supported. - * - * FIXME: Even if we did support multiple DMA controllers, we have no - * mechanism for assigning DMA controllers and channels to the individual - * SSI devices. We also probably aren't compatible with the generic Elo DMA - * device driver. - */ -static int mpc8610_hpcd_probe(struct of_device *ofdev, - const struct of_device_id *match) -{ - struct device_node *np = ofdev->node; - struct device_node *codec_np = NULL; - struct device_node *guts_np = NULL; - struct device_node *dma_np = NULL; - struct device_node *dma_channel_np = NULL; - const phandle *codec_ph; - const char *sprop; - const u32 *iprop; - struct resource res; - struct platform_device *sound_device = NULL; - struct mpc8610_hpcd_data *machine_data; - struct fsl_ssi_info ssi_info; - struct fsl_dma_info dma_info; - int ret = -ENODEV; - unsigned int playback_dma_channel; - unsigned int capture_dma_channel; - - machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) - return -ENOMEM; - - memset(&ssi_info, 0, sizeof(ssi_info)); - memset(&dma_info, 0, sizeof(dma_info)); - - ssi_info.dev = &ofdev->dev; - - /* - * We are only interested in SSIs with a codec phandle in them, so let's - * make sure this SSI has one. - */ - codec_ph = of_get_property(np, "codec-handle", NULL); - if (!codec_ph) - goto error; - - codec_np = of_find_node_by_phandle(*codec_ph); - if (!codec_np) - goto error; - - /* The MPC8610 HPCD only knows about the CS4270 codec, so reject - anything else. */ - if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) - goto error; - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&ofdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - machine_data->ssi_id = *iprop; - ssi_info.id = *iprop; - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&ofdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* - * In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&ofdev->dev, "codec bus-frequency property " - "is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - machine_data->clk_frequency = *iprop; - } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&ofdev->dev, - "unrecognized fsl,mode property \"%s\"\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!machine_data->clk_frequency) { - dev_err(&ofdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Read the SSI information from the device tree */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&ofdev->dev, "could not obtain SSI address\n"); - goto error; - } - if (!res.start) { - dev_err(&ofdev->dev, "invalid SSI address\n"); - goto error; - } - ssi_info.ssi_phys = res.start; - - machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi)); - if (!machine_data->ssi) { - dev_err(&ofdev->dev, "could not map SSI address %x\n", - ssi_info.ssi_phys); - ret = -EINVAL; - goto error; - } - ssi_info.ssi = machine_data->ssi; - - - /* Get the IRQ of the SSI */ - machine_data->ssi_irq = irq_of_parse_and_map(np, 0); - if (!machine_data->ssi_irq) { - dev_err(&ofdev->dev, "could not get SSI IRQ\n"); - ret = -EINVAL; - goto error; - } - ssi_info.irq = machine_data->ssi_irq; - - /* Do we want to use asynchronous mode? */ - ssi_info.asynchronous = - of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0; - if (ssi_info.asynchronous) - dev_info(&ofdev->dev, "using asynchronous mode\n"); - - /* Map the global utilities registers. */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); - if (!guts_np) { - dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); - ret = -EINVAL; - goto error; - } - machine_data->guts = of_iomap(guts_np, 0); - of_node_put(guts_np); - if (!machine_data->guts) { - dev_err(&ofdev->dev, "could not map GUTS\n"); - ret = -EINVAL; - goto error; - } - - /* Find the DMA channels to use. Both SSIs need to use the same DMA - * controller, so let's use DMA#1. - */ - for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { - iprop = of_get_property(dma_np, "cell-index", NULL); - if (iprop && (*iprop == 0)) { - of_node_put(dma_np); - break; - } - } - if (!dma_np) { - dev_err(&ofdev->dev, "could not find DMA node\n"); - ret = -EINVAL; - goto error; - } - machine_data->dma_id = *iprop; - - /* SSI1 needs to use DMA Channels 0 and 1, and SSI2 needs to use DMA - * channels 2 and 3. This is just how the MPC8610 is wired - * internally. - */ - playback_dma_channel = (machine_data->ssi_id == 0) ? 0 : 2; - capture_dma_channel = (machine_data->ssi_id == 0) ? 1 : 3; - - /* - * Find the DMA channels to use. - */ - while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (iprop && (*iprop == playback_dma_channel)) { - /* dma_channel[0] and dma_irq[0] are for playback */ - dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); - dma_info.dma_irq[0] = - irq_of_parse_and_map(dma_channel_np, 0); - machine_data->dma_channel_id[0] = *iprop; - continue; - } - if (iprop && (*iprop == capture_dma_channel)) { - /* dma_channel[1] and dma_irq[1] are for capture */ - dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); - dma_info.dma_irq[1] = - irq_of_parse_and_map(dma_channel_np, 0); - machine_data->dma_channel_id[1] = *iprop; - continue; - } - } - if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] || - !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) { - dev_err(&ofdev->dev, "could not find DMA channels\n"); - ret = -EINVAL; - goto error; - } - - dma_info.ssi_stx_phys = ssi_info.ssi_phys + - offsetof(struct ccsr_ssi, stx0); - dma_info.ssi_srx_phys = ssi_info.ssi_phys + - offsetof(struct ccsr_ssi, srx0); - - /* We have the DMA information, so tell the DMA driver what it is */ - if (!fsl_dma_configure(&dma_info)) { - dev_err(&ofdev->dev, "could not instantiate DMA device\n"); - ret = -EBUSY; - goto error; - } - - /* - * Initialize our DAI data structure. We should probably get this - * information from the device tree. - */ - machine_data->dai.name = "CS4270"; - machine_data->dai.stream_name = "CS4270"; - - iprop = of_get_property(codec_np, "reg", NULL); - if (!iprop) { - dev_err(&ofdev->dev, "codec node is missing 'reg' property\n"); - goto error; - } - machine_data->dai.codec_id = *iprop; - - machine_data->dai.cpu_dai_drv = &fsl_ssi_dai; - machine_data->dai.codec_dai_drv = &cs4270_dai; /* The codec_dai we want */ - machine_data->dai.codec_drv = &soc_codec_device_cs4270; - machine_data->dai.ops = &mpc8610_hpcd_ops; - machine_data->dai.platform_drv = &fsl_soc_platform; - - machine_data->machine.probe = mpc8610_hpcd_machine_probe; - machine_data->machine.remove = mpc8610_hpcd_machine_remove; - machine_data->machine.name = "MPC8610 HPCD"; - machine_data->machine.num_links = 1; - machine_data->machine.dai_link = &machine_data->dai; - - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&ofdev->dev, "platform device allocation failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Set the platform device and ASoC device to point to each other */ - platform_set_drvdata(sound_device, &machine_data->machine); - - /* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(), - if it exists. */ - ret = platform_device_add(sound_device); - - if (ret) { - dev_err(&ofdev->dev, "platform device add failed\n"); - goto error; - } - - dev_set_drvdata(&ofdev->dev, sound_device); - - return 0; - -error: - of_node_put(codec_np); - of_node_put(guts_np); - of_node_put(dma_np); - of_node_put(dma_channel_np); - - if (sound_device) - platform_device_unregister(sound_device); - - if (ssi_info.ssi) - iounmap(ssi_info.ssi); - - if (ssi_info.irq) - irq_dispose_mapping(ssi_info.irq); - - if (dma_info.dma_channel[0]) - iounmap(dma_info.dma_channel[0]); - - if (dma_info.dma_channel[1]) - iounmap(dma_info.dma_channel[1]); - - if (dma_info.dma_irq[0]) - irq_dispose_mapping(dma_info.dma_irq[0]); - - if (dma_info.dma_irq[1]) - irq_dispose_mapping(dma_info.dma_irq[1]); - - if (machine_data->guts) - iounmap(machine_data->guts); - - kfree(machine_data); - - return ret; -} - -/** - * mpc8610_hpcd_remove: remove the OF device - * - * This function is called when the OF device is removed. - */ -static int mpc8610_hpcd_remove(struct of_device *ofdev) -{ - struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev); - struct mpc8610_hpcd_data *machine_data = - sound_device->dev.platform_data; - - platform_device_unregister(sound_device); - - if (machine_data->ssi) - iounmap(machine_data->ssi); - - if (machine_data->dma[0]) - iounmap(machine_data->dma[0]); - - if (machine_data->dma[1]) - iounmap(machine_data->dma[1]); - - if (machine_data->dma_irq[0]) - irq_dispose_mapping(machine_data->dma_irq[0]); - - if (machine_data->dma_irq[1]) - irq_dispose_mapping(machine_data->dma_irq[1]); - - if (machine_data->guts) - iounmap(machine_data->guts); - - kfree(machine_data); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&ofdev->dev, NULL); - - return 0; -} - -static struct of_device_id mpc8610_hpcd_match[] = { - { - .compatible = "fsl,mpc8610-ssi", - }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); - -static struct of_platform_driver mpc8610_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc8610_hpcd", - .match_table = mpc8610_hpcd_match, - .probe = mpc8610_hpcd_probe, - .remove = mpc8610_hpcd_remove, -}; - -/** - * mpc8610_hpcd_init: fabric driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init mpc8610_hpcd_init(void) -{ - int ret; - - printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); - - ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); - - if (ret) - printk(KERN_ERR - "mpc8610-hpcd: failed to register platform driver\n"); - - return ret; -} - -/** - * mpc8610_hpcd_exit: fabric driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit mpc8610_hpcd_exit(void) -{ - of_unregister_platform_driver(&mpc8610_hpcd_of_driver); -} - -module_init(mpc8610_hpcd_init); -module_exit(mpc8610_hpcd_exit); - -MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); -MODULE_LICENSE("GPL"); +/** + * Freescale MPC8610HPCD ALSA SoC Machine driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <asm/immap_86xx.h> + +#include "../codecs/cs4270.h" +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/* There's only one global utilities register */ +static phys_addr_t guts_phys; + +/** + * mpc8610_hpcd_data: machine-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * MPC8610 HPCD. Some of the data is taken from the device tree. + */ +struct mpc8610_hpcd_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int dma_id[0]; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ + char dai_name[32]; +}; + +/** + * mpc8610_hpcd_machine_probe: initialize the board + * + * This function is used to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +{ + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct mpc8610_hpcd_data *machine_data = + container_of(card, struct mpc8610_hpcd_data, card); + struct ccsr_guts __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(guts, machine_data->dma_id[0], + machine_data->dma_channel_id[0], + CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(guts, machine_data->dma_id[1], + machine_data->dma_channel_id[1], + CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(guts, machine_data->dma_id[0], + machine_data->dma_channel_id[0], 0); + guts_set_pmuxcr_dma(guts, machine_data->dma_id[1], + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + + iounmap(guts); + + return 0; +} + +/** + * mpc8610_hpcd_startup: program the board with various hardware parameters + * + * This function takes board-specific information, like clock frequencies + * and serial data formats, and passes that information to the codec and + * transport drivers. + */ +static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mpc8610_hpcd_data *machine_data = + container_of(rtd->card, struct mpc8610_hpcd_data, card); + struct device *dev = rtd->card->dev; + int ret = 0; + + /* Tell the codec driver what the serial protocol is. */ + ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(dev, "could not set codec driver audio format\n"); + return ret; + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(dev, "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +/** + * mpc8610_hpcd_machine_remove: Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +{ + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct mpc8610_hpcd_data *machine_data = + container_of(card, struct mpc8610_hpcd_data, card); + struct ccsr_guts __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Restore the signal routing */ + + guts_set_dmacr(guts, machine_data->dma_id[0], + machine_data->dma_channel_id[0], 0); + guts_set_dmacr(guts, machine_data->dma_id[1], + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + case 1: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); + break; + } + + iounmap(guts); + + return 0; +} + +/** + * mpc8610_hpcd_ops: ASoC machine driver operations + */ +static struct snd_soc_ops mpc8610_hpcd_ops = { + .startup = mpc8610_hpcd_startup, +}; + +/** + * get_node_by_phandle_name - get a node by its phandle name + * + * This function takes a node, the name of a property in that node, and a + * compatible string. Assuming the property is a phandle to another node, + * it returns that node, (optionally) if that node is compatible. + * + * If the property is not a phandle, or the node it points to is not compatible + * with the specific string, then NULL is returned. + */ +static struct device_node *get_node_by_phandle_name(struct device_node *np, + const char *name, + const char *compatible) +{ + const phandle *ph; + int len; + + ph = of_get_property(np, name, &len); + if (!ph || (len != sizeof(phandle))) + return NULL; + + np = of_find_node_by_phandle(*ph); + if (!np) + return NULL; + + if (compatible && !of_device_is_compatible(np, compatible)) { + of_node_put(np); + return NULL; + } + + return np; +} + +/** + * get_parent_cell_index -- return the cell-index of the parent of a node + * + * Return the value of the cell-index property of the parent of the given + * node. This is used for DMA channel nodes that need to know the DMA ID + * of the controller they are on. + */ +static int get_parent_cell_index(struct device_node *np) +{ + struct device_node *parent = of_get_parent(np); + const u32 *iprop; + + if (!parent) + return -1; + + iprop = of_get_property(parent, "cell-index", NULL); + of_node_put(parent); + + if (!iprop) + return -1; + + return *iprop; +} + +static int get_dma_channel(struct device_node *ssi_np, + const char *compatible, + dma_addr_t ssi_stx_phys, + dma_addr_t ssi_srx_phys, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id) +{ + struct device_node *dma_channel_np; + const u32 *iprop; + + dma_channel_np = get_node_by_phandle_name(ssi_np, compatible, + "fsl,ssi-dma-channel"); + if (!dma_channel_np) + return -EINVAL; + + dai->platform_drv = fsl_soc_dma_to_dai(dma_channel_np->full_name, + ssi_stx_phys, ssi_srx_phys); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_channel_np); + return -EINVAL; + } + + *dma_channel_id = *iprop; + *dma_id = get_parent_cell_index(dma_channel_np); + of_node_put(dma_channel_np); + + return 0; +} +/** + * mpc8610_hpcd_probe: platform probe function for the machine driver + * + * Although this is a machine driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + */ +static int mpc8610_hpcd_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_driver **platform_data = pdev->dev.platform_data; + struct snd_soc_dai_driver *cpu_dai_drv = *platform_data; + struct device *dev = pdev->dev.parent; + struct of_device *of_dev = container_of(dev, struct of_device, dev); + struct device_node *np = of_dev->node; + struct device_node *codec_np = NULL; + struct device_node *dma_np = NULL; + struct device_node *dma_channel_np = NULL; + struct platform_device *sound_device = NULL; + struct mpc8610_hpcd_data *machine_data; + int id; + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + int ret = -ENODEV; + const char *sprop; + const u32 *iprop; + struct resource res; + + /* We are only interested in SSIs with a codec phandle in them, + * so let's make sure this SSI has one. The MPC8610 HPCD only + * knows about the CS4270 codec, so reject anything else. + */ + codec_np = get_node_by_phandle_name(np, "codec-handle", + "cirrus,cs4270"); + if (!codec_np) { + dev_err(dev, "invalid codec node\n"); + return -EINVAL; + } + + machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); + if (!machine_data) + return -ENOMEM; + + machine_data->dai[0].cpu_dai_id = 0; + machine_data->dai[0].codec_dai_id = 0; + machine_data->dai[0].platform_id = 0; + + machine_data->dai[0].name = machine_data->dai_name; + + machine_data->dai[0].cpu_dai_drv = cpu_dai_drv; + machine_data->dai[0].codec_dai_drv = &cs4270_dai; + machine_data->dai[0].codec_drv = &soc_codec_device_cs4270; + machine_data->dai[0].ops = &mpc8610_hpcd_ops; + + /* Store the codec name, it will be used as the codec DAI name */ + of_modalias_node(codec_np, machine_data->dai_name, + sizeof(machine_data->dai_name)); + + memcpy(&machine_data->dai[1], &machine_data->dai[0], + sizeof(struct snd_soc_dai_link)); + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&pdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + machine_data->ssi_id = *iprop; + id = *iprop; + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&pdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; + } + + if (strcasecmp(sprop, "i2s-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* In i2s-slave mode, the codec has its own clock source, so we + * need to get the frequency from the device tree and pass it to + * the codec driver. + */ + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&pdev->dev, "codec bus-frequency " + "property is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + machine_data->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&pdev->dev, + "unrecognized fsl,mode property '%s'\n", sprop); + ret = -EINVAL; + goto error; + } + + if (!machine_data->clk_frequency) { + dev_err(&pdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Read the SSI information from the device tree */ + ret = of_address_to_resource(np, 0, &res); + if (ret || !res.start) { + dev_err(&pdev->dev, "could not obtain SSI address\n"); + goto error; + } + ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); + ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + + /* Find the playback DMA channel to use. */ + ret = get_dma_channel(np, "fsl,playback-dma", ssi_stx_phys, + ssi_srx_phys, &machine_data->dai[0], + &machine_data->dma_channel_id[0], + &machine_data->dma_id[0]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); + goto error; + } + + /* Find the capture DMA channel to use. */ + ret = get_dma_channel(np, "fsl,capture-dma", ssi_stx_phys, + ssi_srx_phys, &machine_data->dai[1], + &machine_data->dma_channel_id[1], + &machine_data->dma_id[1]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); + goto error; + } + + /* Initialize our DAI data structure. */ + iprop = of_get_property(codec_np, "reg", NULL); + if (!iprop) { + dev_err(&pdev->dev, "codec node is missing 'reg' property\n"); + goto error; + } + machine_data->dai[0].codec_id = *iprop; + machine_data->dai[1].codec_id = *iprop; + + machine_data->dai[0].stream_name = "playback"; + machine_data->dai[1].stream_name = "capture"; + + machine_data->card.probe = mpc8610_hpcd_machine_probe; + machine_data->card.remove = mpc8610_hpcd_machine_remove; + machine_data->card.name = "MPC8610 HPCD"; + machine_data->card.num_links = 2; + machine_data->card.dai_link = machine_data->dai; + + /* Allocate a new audio platform device structure */ + sound_device = platform_device_alloc("soc-audio", -1); + if (!sound_device) { + dev_err(&pdev->dev, "platform device alloc failed\n"); + ret = -ENOMEM; + goto error; + } + + /* Associate the card data with the sound device */ + platform_set_drvdata(sound_device, &machine_data->card); + + /* Register with ASoC */ + ret = platform_device_add(sound_device); + if (ret) { + dev_err(&pdev->dev, "platform device add failed\n"); + goto error; + } + + of_node_put(codec_np); + + return 0; + +error: + of_node_put(codec_np); + of_node_put(dma_np); + of_node_put(dma_channel_np); + + if (sound_device) + platform_device_unregister(sound_device); + + kfree(machine_data); + + return ret; +} + +/** + * mpc8610_hpcd_remove: remove the platform device + * + * This function is called when the platform device is removed. + */ +static int __devexit mpc8610_hpcd_remove(struct platform_device *pdev) +{ + struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct mpc8610_hpcd_data *machine_data = + container_of(card, struct mpc8610_hpcd_data, card); + + platform_device_unregister(sound_device); + + kfree(machine_data); + sound_device->dev.platform_data = NULL; + + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static struct platform_driver mpc8610_hpcd_driver = { + .probe = mpc8610_hpcd_probe, + .remove = __devexit_p(mpc8610_hpcd_remove), + .driver = { + /* The name must match the 'model' property in the device tree, + * in lowercase letters. + */ + .name = "snd-soc-mpc8610hpcd", + .owner = THIS_MODULE, + }, +}; + +/** + * mpc8610_hpcd_init: machine driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610_hpcd_init(void) +{ + struct device_node *guts_np; + struct resource res; + + pr_info("Freescale MPC8610 HPCD ALSA SoC machine driver\n"); + + /* Get the physical address of the global utilities registers */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (of_address_to_resource(guts_np, 0, &res)) { + pr_err("mpc8610-hpcd: missing/invalid global utilities node\n"); + return -EINVAL; + } + guts_phys = res.start; + + return platform_driver_register(&mpc8610_hpcd_driver); +} + +/** + * mpc8610_hpcd_exit: machine driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit mpc8610_hpcd_exit(void) +{ + platform_driver_unregister(&mpc8610_hpcd_driver); +} + +module_init(mpc8610_hpcd_init); +module_exit(mpc8610_hpcd_exit); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC machine driver"); +MODULE_LICENSE("GPL v2"); -- 1.6.5

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[alsa-devel] How to play a sine wave in alsa?
by Timo Schneider 04 May '10

04 May '10

Hello! I am trying to use alsa to play a sine wave in async mode. Everything seems to work fine, except what I am hearing is _not_ a sine wave. I am checking for XRUNs, but none occur, so I guess the problem is that I am supplying the wrong sample data to alsa. I want the tone only on one channel, that seems to work fine the way i do it, so i guess my assumptions on how the interleaved mode works are not completely wrong. Can anyone give me a hint on what I am doing wrong in the example program below? I already checked the example in tests/pcm.c but it didn't really help me. Regards, Timo Example Program (compile with cc test.c -o test -lasound): #include <math.h> #include <stdlib.h> #include <inttypes.h> #include <alsa/asoundlib.h> struct sound_settings { // alsa parameters snd_pcm_t *pcm_handle; snd_pcm_hw_params_t *hw_params; snd_pcm_sw_params_t *sw_params; snd_async_handler_t *pcm_callback; snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t period_size; unsigned int sample_rate; char *sample_buffer; size_t sample_buffer_size; // user settings made in gui char *cardname; char *modename; int startfreq; int endfreq; int duration; int channels; // keep track of gnerated samples long int actsample; long int lastsample; }; struct sound_settings sdata; void generate_samples(struct sound_settings *sdata) { int i = 0; if (sdata->channels == 1) i=0; if (sdata->channels == 2) i=1; if (sdata->channels == 3) i=0; if (strcmp(sdata->modename, "Sine")==0) { while (i<(sdata->sample_buffer_size / 8)) { int samplerate = 44100; int freq = sdata->startfreq; double sample = sin(sdata->actsample * (double) 2*M_PI / ((double) samplerate / (double)freq)); int16_t rsamp = sample * 65535; sdata->actsample++; *(((int16_t*) (sdata->sample_buffer))+i) = rsamp; i += 2; } } } void MyCallback(snd_async_handler_t *pcm_callback) { snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); snd_pcm_sframes_t avail; struct sound_settings *sdata = (struct sound_settings*) snd_async_handler_get_callback_private(pcm_callback); avail = snd_pcm_avail_update(pcm_handle); if (avail == -EPIPE) { snd_pcm_prepare (pcm_handle); printf("Underrun occured\n"); } while (avail >= sdata->period_size) { generate_samples(sdata); snd_pcm_writei(pcm_handle, sdata->sample_buffer, sdata->period_size); avail = snd_pcm_avail_update(pcm_handle); } } void init_alsa(struct sound_settings *sdata) { int rc; // set some sound_settings data (i.e., sample_rate) to sensible defaults sdata->buffer_size = 1024; sdata->period_size = 64; sdata->sample_rate = 44100; // Open PCM device for playback rc = snd_pcm_open(&(sdata->pcm_handle), "default", SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); // Allocate a hardware parameters object snd_pcm_hw_params_malloc (&(sdata->hw_params)); snd_pcm_hw_params_any(sdata->pcm_handle, sdata->hw_params); snd_pcm_hw_params_set_access(sdata->pcm_handle, sdata->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(sdata->pcm_handle, sdata->hw_params, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(sdata->pcm_handle, sdata->hw_params, 2); snd_pcm_hw_params_set_rate_near(sdata->pcm_handle, sdata->hw_params, &(sdata->sample_rate), NULL); snd_pcm_hw_params_set_buffer_size_near(sdata->pcm_handle, sdata->hw_params, &(sdata->buffer_size)); snd_pcm_hw_params_set_period_size_near (sdata->pcm_handle, sdata->hw_params, &(sdata->period_size), NULL); rc = snd_pcm_hw_params(sdata->pcm_handle, sdata->hw_params); if (rc < 0) fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); // tell alsa we want to work asynchronously snd_pcm_sw_params_malloc(&(sdata->sw_params)); snd_pcm_sw_params_current(sdata->pcm_handle, sdata->sw_params); snd_pcm_sw_params_set_start_threshold(sdata->pcm_handle, sdata->sw_params, sdata->buffer_size - sdata->period_size); snd_pcm_sw_params_set_avail_min(sdata->pcm_handle, sdata->sw_params, sdata->period_size); rc = snd_pcm_sw_params(sdata->pcm_handle, sdata->sw_params); if (rc < 0) fprintf(stderr, "unable to set sw parameters: %s\n", snd_strerror(rc)); // Use a buffer large enough to hold two periods snd_pcm_hw_params_get_period_size(sdata->hw_params, &(sdata->period_size), NULL); sdata->sample_buffer_size = sdata->period_size * 8; // 2 bytes/sample, 2 channels, two periods sdata->sample_buffer = (char *) malloc(sdata->sample_buffer_size); generate_samples(sdata); // prepare dev so that we can write snd_pcm_prepare(sdata->pcm_handle); /* fill buffer with some data */ generate_samples(sdata); /* write first chunk of data */ rc = snd_pcm_writei(sdata->pcm_handle, sdata->sample_buffer, sdata->period_size); if (rc == -EPIPE) snd_pcm_prepare(sdata->pcm_handle); /* register callback */ rc = snd_async_add_pcm_handler(&(sdata->pcm_callback), sdata->pcm_handle, MyCallback, sdata); snd_pcm_start(sdata->pcm_handle); } int main(int argc, char **argv) { sdata.channels = 1; sdata.modename = "Sine"; sdata.startfreq = 220; sdata.actsample = 0; init_alsa(&sdata); static int i=0; while (i<1000000) { i++; sleep(1); } return 0; }

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Re: [alsa-devel] Ooops when working with USB MIDI (2.6.33.1)
by Takashi Iwai 03 May '10

03 May '10

At Fri, 9 Apr 2010 07:51:35 +0100, Tvrtko Ursulin wrote: > > On Thursday 08 Apr 2010 13:22:36 Takashi Iwai wrote: > > > Takashi, do you remember what the original problem was? > > > > Well, I have only a vague memory -- it's a similar scenario that some app > > still accessing after disconnection. The URB can't be handled after > > the disconnection is finished. > > > > I think the patch below might fix in this case. You can try it > > instead of reverting the commit above. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c > > index 2c59afd..81c8d85 100644 > > --- a/sound/usb/usbmidi.c > > +++ b/sound/usb/usbmidi.c > > @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct > > snd_rawmidi_substream *substream) DEFINE_WAIT(wait); > > long timeout = msecs_to_jiffies(50); > > > > + if (ep->umidi->disconnected) > > + return; > > /* > > * The substream buffer is empty, but some data might still be in the > > * currently active URBs, so we have to wait for those to complete. > > @@ -1275,6 +1277,11 @@ void snd_usbmidi_disconnect(struct list_head* p) > > snd_usbmidi_in_endpoint_delete(ep->in); > > ep->in = NULL; > > } > > + ep->active_urbs = 0; > > + if (ep->drain_urbs) { > > + ep->drain_urbs = 0; > > + wake_up(&ep->drain_wait); > > + } > > } > > del_timer_sync(&umidi->error_timer); > > } > > For the second hunk, do you think ep->out->... and so on? That would be more > in-line with code present in 2.6.33. Ah, crap. Sorry, that's just messing up. The revised (compiled but untested) patch is below. thanks, Takashi --- diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2c59afd..9e28b20 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) DEFINE_WAIT(wait); long timeout = msecs_to_jiffies(50); + if (ep->umidi->disconnected) + return; /* * The substream buffer is empty, but some data might still be in the * currently active URBs, so we have to wait for those to complete. @@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, * Frees an output endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep) { unsigned int i; for (i = 0; i < OUTPUT_URBS; ++i) - if (ep->urbs[i].urb) + if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); + ep->urbs[i].urb = NULL; + } +} + +static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep) +{ + snd_usbmidi_out_endpoint_clear(ep); kfree(ep); } @@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); + ep->out->active_urbs = 0; + if (ep->out->drain_urbs) { + ep->out->drain_urbs = 0; + wake_up(&ep->out->drain_wait); + } } if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ - if (ep->out) { - snd_usbmidi_out_endpoint_delete(ep->out); - ep->out = NULL; - } + if (ep->out) + snd_usbmidi_out_endpoint_clear(ep->out); if (ep->in) { snd_usbmidi_in_endpoint_delete(ep->in); ep->in = NULL;

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[alsa-devel] [PATCHv2 0/4] ASoC: tlv320dac33: Support for turning off codec
by Peter Ujfalusi 03 May '10

03 May '10

Hello, Changes since v1: - patch 1: Removed extra register writes - patch 4: - Let core to bring the codec to BIAS_OFF (set idle_bias_off) - dac33_set_bias_level only takes care of power and minimum initialization - PRE_DAPM widget event is replacing the dac33_pcm_pointer call to take care of the initialization needed before _every_ playback. There are few scenarios, which has to be taken care:: 1. Analog bypass caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, but we does not need to execute the playback related configuration 2. Playback caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, and also we need to execute the playback related configuration. 3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is already on. 4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON) Nothing need to be done. 5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is still on. Since the power up, and the codec init is optimized, the added overhead in stream start is minimal. Withing this patch, the hard_power function is now only doing what it supposed to: only handle the powers, and GPIO reset line. The codec initialization and state restore has been moved out. --- Peter Ujfalusi (4): ASoC: tlv320dac33: Optimize power up, and restore ASoC: tlv320dac33: Revised module loading, and DAC33 ID read ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure ASoC: tlv320dac33: Support for turning off the codec sound/soc/codecs/tlv320dac33.c | 235 ++++++++++++++++++++++------------------ 1 files changed, 129 insertions(+), 106 deletions(-)

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[alsa-devel] [PATCHv3 0/2] ASoC: TWL4030: APLL/AIF ordering and DAPM updates
by Peter Ujfalusi 03 May '10

03 May '10

Hello, Changes since v2: apll_enable logic cleaned up. aif refcounting has been removed, aif handling simplified The new DAPM_OUTPUT and DAPM_INPUT widgets renamed Correct the APLL/AIF powering sequence. Also introducing DAPM outputs and input for keeping APLL/AIF enabled in case when no valid DAPM route selected during audio activity. The second patch removes the legacy OUTL/R. As a note: The APLL/AIF patch replaces the same patch in Liam's series. The OUTL/R removal can be applied after the OUTL/R reference has been removed from machine drivers (as in Lima's series). --- Peter Ujfalusi (2): ASoC: TWL4030: AIF/APLL fix in DAPM domain ASoC: TWL4030: Remove OUTL/R outputs sound/soc/codecs/twl4030.c | 86 ++++++++++++++++++++++++++++++------------- 1 files changed, 60 insertions(+), 26 deletions(-)

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[alsa-devel] Changing volume of a single PCM channel
by Andreas Falkenhahn 03 May '10

03 May '10

Hi, how do I change the volume of single PCM channel with ALSA? I'm playing multiple sounds at the same time by opening a new channel for every sound using snd_pcm_open() and let PulseAudio do the mixing. Now assuming that I have allocated four channels using snd_pcm_open()... now I'd like to change the volume of the second channel, the other three channels' volume should not be affected. How can I do this? AFAICS, volume control is provided by the ALSA mixer interface but I don't seem to figure out how to connect the mixer with my PCM channels opened using snd_pcm_open(). Thanks for any help, Andreas

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[alsa-devel] Reforming S3C I2S towards supporting I2Sv4
by Jassi Brar 02 May '10

02 May '10

Hello, I have prepared some patches that debug and modify the behaviour of extant S3C I2S support towards managing I2Sv4 controllers as well. Overall pic... For now the 'new' I2Sv4 CPU driver(s3c64xx-i2s-v4.c) is almost a copy of s3c64xx-i2s.c The driver will evolve as support for more features is added. Minor quirks in otherwise similar controllers are handled by defining a new field(feature) in 'struct s3c_i2sv2_info' that is meant to contain 1-bit flags for absence/presence of such quirks. The header with I2S register map and bit definitions has been copied to where the drivers are(sound/soc/s3c24xx/) since the header has nothing usable for platform code. Also, it will help avoid need for co-ordination between ASoC and S3C ARCH trees. For now, the header regs-s3c2412-iis.h is left intact but rendered useless by making ASoC drivers include the newly copied version of it (sound/soc/s3c24xx/regs-i2s-v2.h) Later the header could be dropped by patches to S3C PLAT tree. The patches apply against 'for-2.6.35' of http://www.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Regards. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com>

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