[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
wendelin klimann
wklimann at gmail.com
Sat Mar 23 22:56:45 CET 2013
Hello Daniel
> *root at beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay
> > -Dhw:1,0 /home/root/fifi.wav*
> > [ 1128.677337] omap-dma-engine omap-dma-engine: allocating channel
> > for 17
> > Playing WAVE '/h[ 1128.689544] can't set codec DAI configuration -
> > pcm3168
>
> This is where your trouble starts, and the reason is that when you call
> snd_soc_dai_set_fmt() on your codec dai, the core will do this:
>
> if (dai->driver->ops->set_fmt == NULL)
> return -ENOTSUPP;
>
> And because you didn't implement that callback in your codec driver, the
> setup will fail. You need to implement that callback, and acknowledge
> that the codec is able to operate under the wanted conditions. See other
> codec drivers for a reference.
>
Thanks that helped to understand the problem and i was able to measure
first signals on my McBSP :-)
>
> Also, you seem to wildly mix machine and codec code, which is exactly
> what ASoC tries to prevent. Codec drivers are completely separated from
> the machine part, because they are supposed to be exchangeable. See what
> other machine code and codec drivers do. In general, as soon you as you
> import a machine or platform specific header file from your codec code,
> you're doing something wrong.
>
>
well i had several headers in my codec file which were not used but i think
there is no code which is mixed between the codec- and the machine-driver.
Next time i will clean up my code before sending it.
Thanks a lot for the fast answer
Wendelin
ps.:
The cleaned up files are added, if there are still mixes between the codec-
and machine-driver i would be glad if you could tell me.
******************************************************************************************************
*
kernel_3.7.4+/sound/soc/codecs/pcm3168.c
*
******************************************************************************************************
/*
* ALSA Soc PCM3168 codec support
*
* Author: Klimann Wendelin <wklimann at hotmail.com>
*
* based on:
* > pcm3008.c
* > Author: Hugo Villeneuve
* > Copyright (C) 2008 Lyrtech inc
* >
* > Based on AC97 Soc codec, original copyright follow:
* > Copyright 2005 Wolfson Microelectronics PLC.
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <linux/init.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <linux/gpio.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "pcm3168.h"
#define PCM3168_VERSION "0.1"
static int pcm3168_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
dev_dbg(codec->dev, "Sample format 0x%X\n", params_format(params));
dev_dbg(codec->dev, "Channels %d\n", params_channels(params));
dev_dbg(codec->dev, "Rate %d\n", params_rate(params));
return 0;
}
static int pcm3168_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
/* codec role */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
dev_dbg(codec->dev, "Codec is master\n");
break;
case SND_SOC_DAIFMT_CBS_CFS:
dev_dbg(codec->dev, "Codec is slave\n");
break;
default:
return -EINVAL;
}
/* DAI format */
dev_dbg(codec->dev, "DAI format 0x%X",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
/* Bit clock and frame sync polarities */
dev_dbg(codec->dev, "Clock polarities 0x%X\n",
fmt & SND_SOC_DAIFMT_INV_MASK);
return 0;
}
#define PCM3168_RATES SNDRV_PCM_RATE_8000_96000 /*(SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)*/
#define PCM3168_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops pcm3168_dai_hifi_ops = {
.hw_params = pcm3168_hw_params,
.set_fmt = pcm3168_set_dai_fmt,
};
static struct snd_soc_dai_driver pcm3168_dai = {
.name = "pcm3168-hifi",
.playback = {
.stream_name = "PCM3168 Playback",
.channels_min = 2,
.channels_max = 8,
.rates = PCM3168_RATES,
.formats = PCM3168_FORMATS, //SNDRV_PCM_FMTBIT_S16_LE,
//SNDRV_PCM_FMTBIT_S32_LE
.sig_bits = 24,
},
.capture = {
.stream_name = "PCM3168 Capture",
.channels_min = 2,
.channels_max = 6,
.rates = PCM3168_RATES,
.formats = PCM3168_FORMATS, //SNDRV_PCM_FMTBIT_S16_LE,
//SNDRV_PCM_FMTBIT_S32_LE
.sig_bits = 24,
},
.ops = &pcm3168_dai_hifi_ops,
};
static int pcm3168_soc_probe(struct snd_soc_codec *codec)
{
struct pcm3168_setup_data *setup = codec->dev->platform_data;
int ret = 0;
printk(KERN_INFO "PCM3168 SoC Audio Codec %s\n", PCM3168_VERSION);
return ret;
}
static int pcm3168_soc_remove(struct snd_soc_codec *codec)
{
struct pcm3168_setup_data *setup = codec->dev->platform_data;
return 0;
}
#define pcm3168_soc_suspend NULL
#define pcm3168_soc_resume NULL
static struct snd_soc_codec_driver soc_codec_dev_pcm3168 = {
.probe = pcm3168_soc_probe,
.remove = pcm3168_soc_remove,
.suspend = pcm3168_soc_suspend,
.resume = pcm3168_soc_resume,
};
static int __devinit pcm3168_codec_probe(struct platform_device *pdev)
{
int ret;
printk(KERN_ALERT "probe pcm3168 codec -> kli \n");
ret = snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_pcm3168, &pcm3168_dai, 1);
printk(KERN_ALERT "probe_after pcm3168 codec -> kli = %d \n", ret);
return ret;
}
static int __devexit pcm3168_codec_remove(struct platform_device *pdev)
{
printk(KERN_ALERT "remove pcm3168 codec -> kli \n");
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver pcm3168_codec_driver = {
.probe = pcm3168_codec_probe,
.remove = __devexit_p(pcm3168_codec_remove),
.driver = {
.name = "pcm3168-codec",
.owner = THIS_MODULE,
},
};
static int __init pcm3168_modinit(void)
{
printk(KERN_ALERT "in init of 3168 -> kli \n");
return platform_driver_register(&pcm3168_codec_driver);
}
module_init(pcm3168_modinit);
static void __exit pcm3168_exit(void)
{
printk(KERN_ALERT "in exit of 3168 -> kli \n");
platform_driver_unregister(&pcm3168_codec_driver);
}
module_exit(pcm3168_exit);
MODULE_DESCRIPTION("Soc PCM3168 driver");
MODULE_AUTHOR("Klimann Wendelin <wklimann at hotmail.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:pcm3168-codec");
******************************************************************************************************
*
kernel_3.7.4+/sound/soc/omap/omap-pcm3168.c
*
******************************************************************************************************
/*
* PCM3168 ASoC driver for BeagleBoard.
*
* based on:
* > omap3beagle.c -- SoC audio for OMAP3 Beagle
* > Author: Steve Sakoman <steve at sakoman.com>
*
* adapted by Klimann Wendelin <wklimann at hotmail.com>
*
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/pcm_params.h>
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
//#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
/*
* Uncomment to test codec in slave mode or without actual codec. This makes
* possible to test this driver by letting the OMAP to be DAI link master
*/
#define PCM3168_CODEC_SLAVE 1
static int pcm3168_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int fmt, div;
int ret;
#ifdef PCM3168_CODEC_SLAVE
fmt = SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_CBS_CFS |
SND_SOC_DAIFMT_IB_NF;
#else
fmt = SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_CBM_CFM |
SND_SOC_DAIFMT_IB_NF;
#endif
/* Set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, fmt);
if (ret < 0) {
printk(KERN_ERR "can't set codec DAI configuration - pcm3168\n");
return ret;
}
/* Set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
if (ret < 0) {
printk(KERN_ERR "can't set cpu DAI configuration - pcm3168\n");
return ret;
}
#ifdef PCM3168_CODEC_SLAVE
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK,
48000000, SND_SOC_CLOCK_IN);
if (ret < 0) {
pr_err("can't set McBSP sysclk - pcm3168\n");
return ret;
}
/*
* Calculate McBSP SRG divisor in McBSP master mode
*/
div = 48000000 / params_rate(params) / params_channels(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
div /= 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
div /= 32;
break;
};
/*
* Round to maximum divisor if needed. This means that extra bit-clock
* cycles are transmitted when sample rate and number of bits in frame
* (channels * sample bits) are low.
*/
if (div >= 256)
div = 256;
ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, div);
if (ret < 0) {
pr_err("can't set SRG clock divider - pcm3168\n");
return ret;
}
#endif
return 0;
}
static struct snd_soc_ops pcm3168_ops = {
.hw_params = pcm3168_hw_params,
};
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link pcm3168_dai = {
.name = "PCM3168",
.stream_name = "PCM3168",
.cpu_dai_name = "omap-mcbsp.3",
.platform_name = "omap-pcm-audio",
.codec_dai_name = "pcm3168-hifi",
.codec_name = "pcm3168-codec.0",
.ops = &pcm3168_ops,
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_pcm3168 = {
.name = "pcm3168",
.owner = THIS_MODULE,
.dai_link = &pcm3168_dai,
.num_links = 1,
};
struct platform_device pcm3168_codec = {
.name = "pcm3168-codec",
.id = 0,
};
struct platform_device pcm3168_soc_audio = {
.name = "pcm3168-soc-audio",
.id = 0,
};
static int __devinit pcm3168_soc_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &snd_soc_pcm3168;
int ret;
pr_info("OMAP3 Beagle - PCM3168 ASoC init\n");
card->dev = &pdev->dev;
ret = snd_soc_register_card(card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d -
pcm3168\n",
ret);
return ret;
}
return 0;
}
static int __devexit pcm3168_soc_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
snd_soc_unregister_card(card);
return 0;
}
static struct platform_driver pcm3168_driver = {
.driver = {
.name = "pcm3168-soc-audio",
.owner = THIS_MODULE,
},
.probe = pcm3168_soc_probe,
.remove = __devexit_p(pcm3168_soc_remove),
};
static int __init pcm3168_soc_init(void)
{
platform_device_register(&pcm3168_codec);
platform_device_register(&pcm3168_soc_audio);
return platform_driver_register(&pcm3168_driver);
}
module_init(pcm3168_soc_init);
static void __exit pcm3168_soc_exit(void)
{
platform_driver_unregister(&pcm3168_driver);
platform_device_unregister(&pcm3168_soc_audio);
platform_device_unregister(&pcm3168_codec);
}
module_exit(pcm3168_soc_exit);
MODULE_AUTHOR("Klimann Wendelin <wklimann at hotmail.com>");
MODULE_DESCRIPTION("ALSA SoC PCM3168 add on Soundcard");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:pcm3168-soc-audio");
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