[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
Wendelin Klimann
wklimann at gmail.com
Thu Apr 4 15:15:05 CEST 2013
Hello
Well i gess slowly i get some success, as i have some kind of audio
output on the audio codec :-)
On 23.03.2013 22:56, wendelin klimann wrote:
> 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.
It looks much better already, but your codec functions are currently
just stubs. So you need to fill them with logic that actually sets up
the hardware of course, and you should use regmap for abstracting the
registers.
Do you mean that i should setup the audiocodec over i2c/spi or miss i
something else?
Also, the codec driver has to become a i2c/spi driver rather than a
platform device. There are tons of other drivers for reference.
I am trying to add support for i2c but can not find how to setup the
correct i2c -> that one which will be used for the communication (i2c-2
in my case)
It would be nice if you could give me an hint how to find that.
There are many smaller things to comment on, but you can submit the
codec driver first once you're finished, so it can be reviewed
independently from the rest. Note, however, that all new drivers have to
be written in a device-tree aware fashion.
I give my best to learn from other device drivers :-)
Acctually i am stuck at a point where i get just 4 channels instead of 8
channels audio output on my audio codec (PCM3168 from TI)
i get every second channel on the audio codec
(alsa channel 1 -> to audio codec channel 1;
alsa channel 3 -> to audio codec channel 2;
alsa channel 5 -> to audio codec channel 3;
alsa channel 7 -> to audio codec channel 4)
The audio codec is setup to use I2S-TDM (24Bit (32Bit frame), 48kHz)
mode and i test the system with:
soc/omap# speaker-test -Dhw:1,0 -c8 -twav
speaker-test 1.0.25
Playback device is hw:1,0
Stream parameters are 48000Hz, S16_LE, 8 channels
WAV file(s)
[ 657.989562] omap-dma-engine omap-dma-engine: allocating channel
for 17
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 16 to 8192
Period size range from 2 to 4096
Using max buffer size 8192
Periods = 4
was set period_size = 2048
was set buffer_size = 8192
0 - Front Left
4 - Center
1 - Front Right
7 - Side Right
3 - Rear Right
2 - Rear Left
6 - Side Left
5 - LFE
Time per period = 11.223022
When i meassure the digital signal with an osziloscope i see that alsa
send all 8 channels (16Bit) on just half of the LRCKL
________________________________
| | |
| | | LRCKL
- AD/DA
| | |
| |______________________________|
| all 8 channels | nothing | AudioCodec-IN / Alsa-OUT
I gess that is because it is using S16_LE format instead of a S32_LE
format but till now i was not able to find a way to change that neither
in the driver by changing the
#define PCM3168_FORMATS SNDRV_PCM_FMTBIT_S32_LE
nor by using
speaker-test -Dhw:1,0 -c8 -twav -FS32_LE
which has the same output as the one without -F32_LE
Please help and thanks in advance
Wendelin
******************************************************************************************************
* kernel_3.7.4+/sound/soc/codecs/pcm3168.c *
******************************************************************************************************
/*
* ALSA Soc PCM3168 codec support
*
* Author: Klimann Wendelin <wklimann at hotmail.com
<mailto: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 = 8,
.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 = 6,
.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
<mailto: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 <mailto:steve at sakoman.com>>
*
* adapted by Klimann Wendelin <wklimann at hotmail.com
<mailto: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_I2S |
SND_SOC_DAIFMT_CBS_CFS |
SND_SOC_DAIFMT_NB_NF;
#else
fmt = SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_CBM_CFM |
SND_SOC_DAIFMT_NB_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;
}
/* Set the codec system clock for DAC and ADC */
/*
ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set codec system clock\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
<mailto: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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