April 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Control vs Device
by Jean-Yves Avenard 27 Apr '10

27 Apr '10

Hi. Currently maintaining the audio code on the MythTV project... Lately, I've had reports on issues by users when using digital audio. This for some reasons however fails, unless the device opened is something like : IEC958:{AES0: 0x06} When we are about to configure a digital audio stream, we do the equivalent of what iecset does. So I would have thought that our code should work under most circumstances... Then I started to suspect the code setting the IEC958_AES0_NONAUDIO bit. We always open the "default" controller like so: snd_ctl_open(&ctl, "default", 0) I thought that maybe the issue is that the "default" controller isn't the one to open. Is there a way, from the name of the device card ; to retrieve which controller is used? So before opening a specific ALSA audio device, I find the corresponding controller, set the flag non-audio on it and continue as usual.. I couldn't find any functions nor documentation that would help me achieve such results.. Any hints would be greatly appreciated... Thank you !! Jean-Yves

1 0

[alsa-devel] [PATCH 01/20] ASoC: S3C: I2Sv2: Add missing semicolon
by Jassi Brar 27 Apr '10

27 Apr '10

Add missing semicolon after s3c2412_i2s_delay Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index aa84f4c..8f08508 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -753,7 +753,7 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) /* Allow overriding by (for example) IISv4 */ if (!ops->delay) - ops->delay = s3c2412_i2s_delay, + ops->delay = s3c2412_i2s_delay; dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; -- 1.6.2.5

4 52

[alsa-devel] "Resource temporarily unavailable" while reading although poll returns POLLIN event
by Stefan Schoenleitner 27 Apr '10

27 Apr '10

Hi, I wrote a small application that opens both streams (i.e. capture and playback) of a PCM in non blocking (i.e. SND_PCM_NONBLOCK) mode. During device configuration I also set up the minimum number of frames to consider a PCM ready: --------------------------------------------------------------------------------- // when to consider the device to be ready for the next data transfer operation if ((ret=snd_pcm_sw_params_set_avail_min(handle, swparams, pcm_period_size))<0) { fprintf(stderr, "could not set avail_min: %s\n", snd_strerror(ret)); return NULL; } --------------------------------------------------------------------------------- As you can see it is set to the number of frames one period contains. In other words, each time there are at least that many frames available for reading on the capture device, we should get a POLLIN event when using poll(). On the other side we should get a POLLOUT event if that many frames can be written to the playback device. Then, later in my application, I use poll() on the fds of the PCM devices. As, according to the ALSA reference, poll events can be "mangeled", I use snd_pcm_poll_descriptors_revents() to "demangle" the events: --------------------------------------------------------------------------------- if ((ret=snd_pcm_poll_descriptors_revents(device_handle, &poll_fds[i], 1, &revents))<0) { fprintf(stderr, "could not snd_pcm_poll_descriptors_revents: %s\n", snd_strerror(ret)); exit(EXIT_FAILURE); } --------------------------------------------------------------------------------- After that, each time there was a POLLIN event, I am reading exactly one period from the capture PCM. On the other side, if there was a POLLOUT event, I am writing exactly one period to the playback PCM. While this works for a small number of periods (which is not always the same), I always end up with a "Resource temporarily unavailable" error when trying to read one period from the capture PCM: --------------------------------------------------------------------------------- demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable --------------------------------------------------------------------------------- Since a POLLIN event only occurs after at least a full period is available for reading (as set up by snd_pcm_sw_params_set_avail_min() above) and I only read after a POLLIN event occured on the capture device fd, I really do not understand why I get the above error. So why does the above happen ? What am I doing wrong ? If it helps, I pasted the source code here so that you can view it nicely: http://pastebin.com/fCicqctq cheers, stefan

3 18

[alsa-devel] [PATCH 2/2] powerpc/alsa: update MPC8610 HPCD audio drivers to support multiple components
by Timur Tabi 26 Apr '10

26 Apr '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 | 42 +-- sound/soc/fsl/fsl_dma.h | 1 + sound/soc/fsl/fsl_ssi.c | 88 ++-- sound/soc/fsl/fsl_ssi.h | 23 - sound/soc/fsl/mpc8610_hpcd.c | 1188 ++++++++++++++++++++---------------------- 5 files changed, 637 insertions(+), 705 deletions(-) rewrite sound/soc/fsl/mpc8610_hpcd.c (64%) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 857e9f7..326b852 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -15,7 +15,6 @@ #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> @@ -844,6 +843,7 @@ EXPORT_SYMBOL_GPL(fsl_soc_platform); int fsl_dma_configure(struct fsl_dma_info *dma_info) { static int initialized; + int ret; /* We only support one DMA controller for now */ if (initialized) @@ -859,42 +859,16 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info) dma_global_data.assigned[1] = 0; initialized = 1; - return 1; -} -EXPORT_SYMBOL_GPL(fsl_dma_configure); - -static int __devinit fsl_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, -1, &fsl_soc_platform); -} - -static int __devexit fsl_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev, -1); - 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 int __init snd_fsl_pcm_init(void) -{ - return platform_driver_register(&fsl_pcm_driver); -} -module_init(snd_fsl_pcm_init); + ret = snd_soc_register_platform(dma_info->dev, -1, &fsl_soc_platform); + if (ret) { + dev_err(dma_info->dev, "could not register platform"); + return 0; + } -static void __exit snd_fsl_pcm_exit(void) -{ - platform_driver_unregister(&fsl_pcm_driver); + return 1; } -module_exit(snd_fsl_pcm_exit); +EXPORT_SYMBOL(fsl_dma_configure); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 253bd3c..62bb617 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -140,6 +140,7 @@ struct fsl_dma_info { dma_addr_t ssi_srx_phys; struct ccsr_dma_channel __iomem *dma_channel[2]; unsigned int dma_irq[2]; + struct device *dev; }; extern struct snd_soc_platform_driver fsl_soc_platform; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 780e243..99cc662 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -14,6 +14,7 @@ #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> @@ -86,12 +87,12 @@ */ struct fsl_ssi_private { char name[8]; + int id; 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; @@ -639,92 +640,115 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, return length; } -static int fsl_ssi_probe(struct platform_device *pdev) +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 u32 *prop; + int id; + struct resource res; + + /* 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; ssi_private = kzalloc(sizeof(struct fsl_ssi_private), 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; - 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; + prop = of_get_property(np, "cell-index", NULL); + if (!prop) { + dev_err(&of_dev->dev, "node is missing cell-index property\n"); + return -ENODEV; + } + id = *prop; + + strcpy(ssi_private->name, "ssi"); + ssi_private->id = id; + ssi_private->ssi = of_iomap(np, 0); + of_address_to_resource(np, 0, &res); + ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); + + if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) + ssi_private->asynchronous = 1; + else + fsl_ssi_dai.symmetric_rates = 1; /* Initialize the the device_attribute structure */ dev_attr->attr.name = "ssi-stats"; 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); + dev_set_drvdata(&of_dev->dev, ssi_private); fsl_ssi_dai.name = ssi_private->name; - fsl_ssi_dai.symmetric_rates = 1; - ret = snd_soc_register_dai(ssi_private->dev, ssi_info->id, &fsl_ssi_dai); + ret = snd_soc_register_dai(&of_dev->dev, id, &fsl_ssi_dai); 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; + 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); + snd_soc_unregister_dai(&of_dev->dev, ssi_private->id); + device_remove_file(&of_dev->dev, &ssi_private->dev_attr); kfree(ssi_private); 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); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 965fd86..7cb229b 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -196,29 +196,6 @@ 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 64% index ad8d763..5276cbe 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,616 +1,572 @@ -/** - * 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 Fabric 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" + +/** + * 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 card; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + struct ccsr_guts __iomem *guts; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int dma_id; /* 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); + + /* 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 = + container_of(rtd->card, struct mpc8610_hpcd_data, card); + struct device *dev = rtd->card->dev; + 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(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(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(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(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); + + /* 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, +}; + +/** + * 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; +} + +/** + * mpc8610_hpcd_probe: platform probe function for the fabric driver + * + * 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. We have no + * mechanism for assigning DMA controllers and channels to the individual + * SSI devices. + */ +static int mpc8610_hpcd_probe(struct platform_device *pdev) +{ + struct device_node *np = NULL; + struct device_node *codec_np = NULL; + struct device_node *dma_np = NULL; + struct device_node *dma_channel_np = NULL; + struct device_node *guts_np; + struct platform_device *sound_device; + struct mpc8610_hpcd_data *machine_data; + int id; + struct fsl_dma_info dma_info; + int ret = -ENODEV; + + /* Map the global utilities registers. */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (!guts_np) { + dev_err(&pdev->dev, "missing global utilities node\n"); + return -EINVAL; + } + + while ((np = of_find_compatible_node(np, NULL, "fsl,mpc8610-ssi"))) { + const char *sprop; + const u32 *iprop; + struct resource res; + + sound_device = NULL; + + machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), + GFP_KERNEL); + if (!machine_data) + return -ENOMEM; + + memset(&dma_info, 0, sizeof(dma_info)); + + /* 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) + continue; + + /* 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; + } + dma_info.ssi_stx_phys = res.start + + offsetof(struct ccsr_ssi, stx0); + dma_info.ssi_srx_phys = res.start + + offsetof(struct ccsr_ssi, srx0); + + /* Find the playback DMA channelto use. */ + dma_channel_np = + get_node_by_phandle_name(np, "fsl,playback-dma", + "fsl,ssi-dma-channel"); + if (!dma_channel_np) { + dev_err(&pdev->dev, "missing or invalid phandle\n"); + ret = -EINVAL; + goto error; + } + dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + dev_err(&pdev->dev, "missing cell-index property\n"); + ret = -EINVAL; + goto error; + } + machine_data->dma_channel_id[0] = *iprop; + of_node_put(dma_channel_np); + + /* Find the capture DMA channelto use. */ + dma_channel_np = + get_node_by_phandle_name(np, "fsl,capture-dma", + "fsl,ssi-dma-channel"); + if (!dma_channel_np) { + dev_err(&pdev->dev, "missing or invalid phandle\n"); + ret = -EINVAL; + goto error; + } + dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + dev_err(&pdev->dev, "missing cell-index property\n"); + ret = -EINVAL; + goto error; + } + machine_data->dma_channel_id[1] = *iprop; + + /* Find the cell-index of the parent DMA node. We assume + * that both DMA channels are part of the same DMA + * controller, but that doesn't necessarily have to be true. + */ + dma_np = of_get_parent(dma_channel_np); + of_node_put(dma_channel_np); + dma_channel_np = NULL; + if (!dma_np) { + dev_err(&pdev->dev, "missing parent DMA node\n"); + ret = -EINVAL; + goto error; + } + iprop = of_get_property(dma_np, "cell-index", NULL); + if (!iprop) { + dev_err(&pdev->dev, "missing cell-index property\n"); + ret = -EINVAL; + goto error; + } + machine_data->dma_id = *iprop; + of_node_put(dma_np); + dma_np = NULL; + + /* Map the global utilities registers */ + machine_data->guts = of_iomap(guts_np, 0); + if (!machine_data->guts) { + dev_err(&pdev->dev, + "could not map global utilities node\n"); + ret = -EINVAL; + goto error; + } + + dma_info.dev = &pdev->dev; + + /* We have the DMA information, so tell the DMA driver */ + if (!fsl_dma_configure(&dma_info)) { + dev_err(&pdev->dev, + "could not instantiate DMA device\n"); + ret = -EBUSY; + 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.codec_id = *iprop; + machine_data->dai.cpu_dai_id = id; + machine_data->dai.codec_dai_id = 0; + machine_data->dai.platform_id = 0; + + /* Store the codec name as the DAI name */ + of_modalias_node(codec_np, machine_data->dai_name, + sizeof(machine_data->dai_name)); + of_node_put(codec_np); + codec_np = NULL; + machine_data->dai.name = machine_data->dai_name; + machine_data->dai.stream_name = machine_data->dai_name; + + machine_data->dai.cpu_dai_drv = &fsl_ssi_dai; + machine_data->dai.codec_dai_drv = &cs4270_dai; + 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->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 = 1; + 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(guts_np); + return 0; + +error: + of_node_put(codec_np); + of_node_put(dma_np); + of_node_put(guts_np); + of_node_put(dma_channel_np); + + if (sound_device) + platform_device_unregister(sound_device); + + 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 (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 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); + + if (machine_data->guts) + iounmap(machine_data->guts); + + 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 = { + .name = "snd-soc-mpc8610-hpcd", + .owner = THIS_MODULE, + }, +}; + +/** + * mpc8610_hpcd_init: fabric driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610_hpcd_init(void) +{ + printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + + return platform_driver_register(&mpc8610_hpcd_driver); +} + +/** + * mpc8610_hpcd_exit: fabric 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 fabric driver"); +MODULE_LICENSE("GPL"); -- 1.6.5

1 0

[alsa-devel] [patch] Fix headphone automute on Asus G50Vt laptops
by Solomon Peachy 26 Apr '10

26 Apr '10

On an Asus G50Vt laptop, plugging in the headphone jack mutes the internal speakers and the headphone jack too. Unplugging the jack unmutes both. This is a regression introduced between linux 2.6.27 and 2.6.28, and is still present in current alsa git sources. The root cause is that the g50v hooks in the hda-intel/patch_reaktek driver call an automute function for a different model (m51va). This patch has been sitting in the ALSA bug tracker for more than a year. Any chance int can finally be applied? https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4468 https://bugtrack.alsa-project.org/alsa-bug/file_download.php?file_id=2913&t… Even with this patch applied, there's still a secondary problem of the master 'mute' toggling both headphone and speaker outputs (I'm still poking around with hda_analyzer to figure out why) but even that's a vast improvement over no sound output at all when something's plugged in. Thanks! - Solomon -- Solomon Peachy pizza at shaftnet dot org Melbourne, FL ^^ (mail/jabber/gtalk) ^^ Quidquid latine dictum sit, altum viditur.

2 5

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 26 Apr '10

26 Apr '10

The following changes since commit 7add84aa779b4f7ca39484a82a11e01d7acd4dca: Mark Brown (1): ASoC: Allow unspecified source when stopping WM8994 FLLs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Barry Song (2): ASoC: ad193x: fix wrong register setting in ad193x_set_dai_fmt ASoC: ad193x: fix typo, delete redundant space Mark Brown (3): ASoC: Allow reporting of NULL jacks ASoC: Correct inversion of speaker mixer PCM switch ASoC: Warn on low WM8994 AIFCLK Vladimir Zapolskiy (1): ASoC: UDA134X: Add UDA1345 CODEC support include/sound/uda134x.h | 1 + sound/soc/codecs/ad193x.c | 56 ++++++++++++++++++++++--------------------- sound/soc/codecs/uda134x.c | 13 ++++++++++ sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8994.c | 5 ++++ sound/soc/soc-jack.c | 5 +-- 6 files changed, 51 insertions(+), 31 deletions(-)

2 1

[alsa-devel] [PATCHv2 0/5] *** SUBJECT HERE ***ASoC: tlv320dac33: FIFO caused delay reporting and fixes
by Peter Ujfalusi 26 Apr '10

26 Apr '10

Hello, Changelog: - Calculated delay instead of msleep(2) in patch 1 - ns changed to us in patch 5 (in the macro for calculation) The main feature of this series is the FIFO caused delay reporting on the DAC33 codec. The main patch is quite big, but I don't think it can be divided further. Short details of the actual implementation is in the commit message(s), and I will not copy it here. The values reported by the delay call has been verified by using one GPIO as a signal, and printout from the kernel at the same time with the delay, than based on the time scale on the scope several (>100) cases has been hand calculated and the results were compared to the kernel reported values. The difference were around (or less) the 10 samples, but during the bursts it is much harder to count, but the reported value was reasonable accurate there as well. --- Peter Ujfalusi (5): ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1 ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode ASoC: tlv320dac33: Change magic numbers used in Mode7 ASoC: tlv320dac33: Calculate the interface speed during bursts ASoC: tlv320dac33: FIFO caused delay reporting sound/soc/codecs/tlv320dac33.c | 250 ++++++++++++++++++++++++++++++++++++++-- 1 files changed, 238 insertions(+), 12 deletions(-)

3 7

[alsa-devel] [PATCH] ASoC: UDA134X: Add UDA1345 CODEC support
by Vladimir Zapolskiy 26 Apr '10

26 Apr '10

This patch adds support for Philips UDA1345 CODEC. The CODEC has only volume control, de-emphasis, mute, DC filtering and power control features. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/uda134x.h | 1 + sound/soc/codecs/uda134x.c | 13 +++++++++++++ 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 475ef8b..509efb0 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -21,6 +21,7 @@ struct uda134x_platform_data { #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 +#define UDA134X_UDA1345 4 }; #endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 20deaca..2f4d728 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -431,6 +431,14 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +static const struct snd_kcontrol_new uda1345_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + static struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -486,6 +494,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) case UDA134X_UDA1340: case UDA134X_UDA1341: case UDA134X_UDA1344: + case UDA134X_UDA1345: break; default: printk(KERN_ERR "UDA134X SoC codec: " @@ -551,6 +560,10 @@ static int uda134x_soc_probe(struct platform_device *pdev) ret = snd_soc_add_controls(codec, uda1341_snd_controls, ARRAY_SIZE(uda1341_snd_controls)); break; + case UDA134X_UDA1345: + ret = snd_soc_add_controls(codec, uda1345_snd_controls, + ARRAY_SIZE(uda1345_snd_controls)); + break; default: printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); -- 1.7.0.4

3 2

[alsa-devel] ASoC - Support for multiple components
by Liam Girdwood 26 Apr '10

26 Apr '10

Currently ASoC is designed around a single CODEC and a single platform DMA engine in every sound card instance. This is fine for most embedded devices but current smart phone and STB designs are starting to outgrow this architecture. I'm currently working on adding ASoC support for multiple different CODECs and Platforms (DMA, Audio Engines) into a single sound card instance. This work will allow ASoC to support N CODECs and N platforms per sound card instance and also allow better integration of audio components from GSM MODEMs, BT, FM transceivers and PCM DSPs. I've now split each ASoC component (i.e. DMA, CODEC, DAI) into driver and device structures. This now means each ASoC component is a regular kernel device and can have private device data and platform_data. It should also provide better support for open firmware and flattened device tree. I've CC'ed folks on this mail who have either contributed or maintain ASoC architecture code. Please have a look at your architecture and test if you can. I only have access to OMAP and pxa3xx hardware and as such have only tested the changes on these two architectures only. I'd appreciated anyone else testing on the other architectures too. The changes are all purely mechanical to component registration and component private data only. However, some architectures (txx9, imx, s3c) required some extra effort to use the device model as they had (to varying degrees) coupled their DMA and DAI code more tightly. The fsl platform also required extra work around the open firmware interface. Grant/Timur do we still need soc-of-simple now that all components are regular devices ? The code is in my topic/multi-component branch here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git It's currently a patch for each component for easier review atm but will be rebased into a single commit when it's ready for upstream so it won't break bisect. If the testing can be done quickly then we can go for 2-6.35 otherwise we are looking at 2.6.36. Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

5 21

[alsa-devel] CA0110 bug report
by Guillaume Chevaleyre 26 Apr '10

26 Apr '10

Hello, I don't know if it's the good procedure, but anyway, maybe this could help the team. bash-4.1# lspci 02:00.0 PCI bridge: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG PCI to PCIe Bridge 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG bash-4.1# modprobe snd-ctxfi bash-4.1# cat /proc/asound/modules 0 snd_hda_intel Track: http://www.alsa-project.org/db/?f=5dd8c336ee72b09e158cb32f08749a8db64715d2 Hope it could help.

2 1