[alsa-devel] [PATCH 2/2] ALSA: ASoc: putting together AT91SAM9260 and TLV320AIC3X
Ryan Mallon
ryan at bluewatersys.com
Mon Apr 4 22:07:39 CEST 2011
On 04/04/2011 08:57 PM, Prchal Jiří wrote:
> Hi,
> this is separeted PATCH for glue AT91SAM9260 and TLV320AIC3X on CDU board.
Please repost this (and the other patch) as a new patch with a proper
change log. You should also Cc Liam Girdwood and Mark Brown, who
maintain the SoC audio system and will pick up this patch.
Some more comments below.
~Ryan
<snip>
>
> diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/atmel/Kconfig
> linux-2.6.38-patch/sound/soc/atmel/Kconfig
> --- linux-2.6.38-vanilla/sound/soc/atmel/Kconfig 2011-03-15 02:20:32.000000000 +0100
> +++ linux-2.6.38-patch/sound/soc/atmel/Kconfig 2011-03-22 09:46:46.751566158 +0100
> @@ -50,3 +50,11 @@ config SND_AT91_SOC_AFEB9260
> select SND_SOC_TLV320AIC23
> help
> Say Y here to support sound on AFEB9260 board.
> +
> +config SND_AT91_SOC_CDU
> + tristate "SoC Audio support for CDU board"
> + depends on ARCH_AT91 && MACH_CDU && SND_ATMEL_SOC
> + select SND_ATMEL_SOC_SSC
> + select SND_SOC_TLV320AIC3X
> + help
> + Say Y here to support sound on CDU board.
> diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/atmel/Makefile
> linux-2.6.38-patch/sound/soc/atmel/Makefile
> --- linux-2.6.38-vanilla/sound/soc/atmel/Makefile 2011-03-15 02:20:32.000000000 +0100
> +++ linux-2.6.38-patch/sound/soc/atmel/Makefile 2011-03-16 09:26:17.000000000 +0100
> @@ -14,3 +14,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
> obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
> obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
> obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
> +obj-$(CONFIG_SND_AT91_SOC_CDU) += snd-soc-cdu.o
> diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/atmel/snd-soc-cdu.c
> linux-2.6.38-patch/sound/soc/atmel/snd-soc-cdu.c
> --- linux-2.6.38-vanilla/sound/soc/atmel/snd-soc-cdu.c 1970-01-01 01:00:00.000000000 +0100
> +++ linux-2.6.38-patch/sound/soc/atmel/snd-soc-cdu.c 2011-03-25 13:42:07.545051043 +0100
> @@ -0,0 +1,259 @@
> +/*
> + * snd-soc-cdu -- SoC audio for AT91SAM9260-based
> + * AKsignal CDU board.
> + *
> + * Copyright (C) 2005 SAN People
> + * Copyright (C) 2008 Atmel
> + * Copyright (C) 2011 AK signal Brno
> + *
> + * Authors: Sedji Gaouaou <sedji.gaouaou at atmel.com>
> + * Jiri Prchal <jiri.prchal at aksignal.cz>
> + *
> + * Based on ati_b1_wm8731.c by:
> + * Frank Mandarino <fmandarino at endrelia.com>
> + * Copyright 2006 Endrelia Technologies Inc.
> + * Based on corgi.c by:
> + * Copyright 2005 Wolfson Microelectronics PLC.
> + * Copyright 2005 Openedhand Ltd.
> + *
> + * This program is free software; you can redistribute it and/or modify
> + * it under the terms of the GNU General Public License as published by
> + * the Free Software Foundation; either version 2 of the License, or
> + * (at your option) any later version.
> + *
> + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
> + */
> +
> +#include <linux/module.h>
> +#include <linux/kernel.h>
> +#include <linux/clk.h>
> +#include <linux/platform_device.h>
> +
> +#include <linux/atmel-ssc.h>
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +
> +#include <asm/mach-types.h>
> +#include <mach/hardware.h>
> +
> +#include "../codecs/tlv320aic3x.h"
> +#include "atmel-pcm.h"
> +#include "atmel_ssc_dai.h"
> +
> +/*
> + * Table of supported rates
> + * with their clock divider (cmrdiv) and number clock in frame (period + 1 * 2),
> + * codecclk sould be integer multiple of sample rate even if it is not exact true
> + * to avoid data on SSC underrun / overrun -
> + * CPU can not generate clock as fine as CODEC.
> + */
> +struct {
> + unsigned int rate;
> + unsigned int codecclk;
> + unsigned int cmrdiv;
> + unsigned int period;
> +} cdu_audio[] = {
Should be static.
> + {8000, 2096000, 25, 130,},
> + {16000, 2496000, 21, 77,},
> + {32000, 2496000, 21, 38,},
> + {48000, 2016000, 26, 20,},
> + {96000, 4032000, 13, 20,},
> +
> + {11025, 2381400, 22, 107,},
> + {22050, 2381400, 22, 53,},
> + {44100, 2381400, 22, 26,},
> + {88200, 4762800, 11, 26,},
> +
> + /* special rates for serial line transfer */
> + {11520, 2626560, 20, 113,},
> + {23040, 2626560, 20, 56,},
> + {46080, 2764800, 19, 29,},
> +};
> +
> +static int cdu_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;
> + struct device *dev = rtd->card->dev;
> + int ret;
> + int i, found = 0;
> +
> + /* set codec DAI configuration */
> + ret = snd_soc_dai_set_fmt(codec_dai,
> + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
> + SND_SOC_DAIFMT_CBS_CFS);
> + if (ret < 0)
> + return ret;
> +
> + /* set cpu DAI configuration */
> + ret = snd_soc_dai_set_fmt(cpu_dai,
> + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
> + SND_SOC_DAIFMT_CBS_CFS);
> + if (ret < 0)
> + return ret;
> +
> + /* Only support 1 or 2 channel S16_LE */
> + if ((params_channels(params) != 1 && params_channels(params) != 2) ||
> + params_format(params) != SNDRV_PCM_FORMAT_S16_LE)
> + return -EINVAL;
> +
> + /* Check rate support */
> + for (i = 0; i < ARRAY_SIZE(cdu_audio); i++)
> + if (params_rate(params) == cdu_audio[i].rate) {
> + found = 1;
> + break;
> + }
> + if (!found)
> + return -EINVAL;
> +
> + /* Set the codec system clock for DAC and ADC */
> + ret = snd_soc_dai_set_sysclk(codec_dai,
> + CLKIN_BCLK, cdu_audio[i].codecclk,
> + SND_SOC_CLOCK_IN);
> + if (ret < 0) {
> + dev_err(dev, "can't set codec system clock\n");
> + return ret;
> + }
> +
> + /* Set the cpu clock dividers to BCLK */
> + ret = snd_soc_dai_set_clkdiv(cpu_dai,
> + ATMEL_SSC_CMR_DIV, cdu_audio[i].cmrdiv);
> + ret |= snd_soc_dai_set_clkdiv(cpu_dai,
> + ATMEL_SSC_TCMR_PERIOD,
> + cdu_audio[i].period);
> + ret |= snd_soc_dai_set_clkdiv(cpu_dai,
> + ATMEL_SSC_RCMR_PERIOD,
> + cdu_audio[i].period);
> + if (ret < 0) {
If more than one of the functions above returns an error then you will
end up with an odd value for ret here. Either test each one individually
and return the correct error code in each case, or return something more
sensible like -EINVAL below.
> + dev_err(dev, "can't set cpu system clock\n");
> + return ret;
> + }
> +
> + return 0;
> +}
> +
> +static struct snd_soc_ops cdu_ops = {
> + .hw_params = cdu_hw_params,
> +};
> +
> +static const struct snd_soc_dapm_widget cdu_dapm_widgets[] = {
> + SND_SOC_DAPM_HP("Headphone Jack", NULL),
> + SND_SOC_DAPM_LINE("Line Out", NULL),
> + SND_SOC_DAPM_MIC("Mic Jack", NULL),
> + SND_SOC_DAPM_LINE("Line In", NULL),
> +};
> +
> +static const struct snd_soc_dapm_route intercon[] = {
> + /* Headphone connected to HPLOUT, HPROUT */
> + {"Headphone Jack", NULL, "HPLOUT"},
> + {"Headphone Jack", NULL, "HPROUT"},
> +
> + /* Line Out connected to LLOUT, RLOUT */
> + {"Line Out", NULL, "LLOUT"},
> + {"Line Out", NULL, "RLOUT"},
> +
> + /* Mic connected to (MIC3L | MIC3R) */
> + {"MIC3L", NULL, "Mic Bias 2V"},
> + {"MIC3R", NULL, "Mic Bias 2V"},
> + {"Mic Bias 2V", NULL, "Mic Jack"},
> +
> + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
> + {"LINE1L", NULL, "Line In"},
> + {"LINE2L", NULL, "Line In"},
> + {"LINE1R", NULL, "Line In"},
> + {"LINE2R", NULL, "Line In"},
> +};
> +
> +/*
> + * Logic for a aic3x as connected on a cdu board.
> + */
> +static int cdu_aic3x_init(struct snd_soc_pcm_runtime *rtd)
> +{
> + struct snd_soc_codec *codec = rtd->codec;
> + struct snd_soc_dai *codec_dai = rtd->codec_dai;
> + struct snd_soc_dapm_context *dapm = &codec->dapm;
> + int ret;
> +
> + /* Add specific widgets */
> + snd_soc_dapm_new_controls(dapm, cdu_dapm_widgets,
> + ARRAY_SIZE(cdu_dapm_widgets));
> + /* Set up specific audio path interconnects */
> + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
> +
> + return 0;
> +}
> +
> +static struct snd_soc_dai_link cdu_dai = {
> + .name = "TLV320AIC3106",
> + .stream_name = "PCM",
> + .cpu_dai_name = "atmel-ssc-dai.0",
> + .codec_dai_name = "tlv320aic3x-hifi",
> + .init = cdu_aic3x_init,
> + .platform_name = "atmel-pcm-audio",
> + .codec_name = "spi1.3",
> + .ops = &cdu_ops,
> +};
> +
> +static struct snd_soc_card snd_soc_cdu = {
> + .name = "TLV320AIC3106",
> + .dai_link = &cdu_dai,
> + .num_links = 1,
> +};
> +
> +static struct platform_device *cdu_snd_device;
> +
> +static int __init cdu_init(void)
> +{
> + int ret;
> +
> + ret = atmel_ssc_set_audio(0);
> + if (ret) {
> + pr_err("Failed to set SSC 0 for audio: %d\n", ret);
> + return ret;
goto err to be consistent.
> + }
> +
> + cdu_snd_device = platform_device_alloc("soc-audio", -1);
> + if (!cdu_snd_device) {
> + pr_err("ASoC: Platform device allocation failed: %d\n", ret);
> + ret = -ENOMEM;
goto err?
> + }
> +
> + platform_set_drvdata(cdu_snd_device, &snd_soc_cdu);
> +
> + ret = platform_device_add(cdu_snd_device);
> + if (ret) {
> + pr_err("ASoC: Platform device adding failed: %d\n", ret);
> + goto err_device_add;
> + }
> +
> + return ret;
> +
> +err_device_add:
> + platform_device_put(cdu_snd_device);
> +err:
> + return ret;
> +}
> +
> +static void __exit cdu_exit(void)
> +{
> + platform_device_unregister(cdu_snd_device);
> +}
> +
> +module_init(cdu_init);
> +module_exit(cdu_exit);
> +
> +/* Module information */
> +MODULE_AUTHOR("Jiri Prchal <jiri.prchal at aksignal.cz>");
> +MODULE_DESCRIPTION("ALSA SoC CDU_AIC3X");
> +MODULE_LICENSE("GPL");
--
Bluewater Systems Ltd - ARM Technology Solution Centre
Ryan Mallon 5 Amuri Park, 404 Barbadoes St
ryan at bluewatersys.com PO Box 13 889, Christchurch 8013
http://www.bluewatersys.com New Zealand
Phone: +64 3 3779127 Freecall: Australia 1800 148 751
Fax: +64 3 3779135 USA 1800 261 2934
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