[alsa-devel] RFC: ASoC driver for S3C24XX Simtec boards
I'd like some comments on the following driver which is currently being developed to get ASoC working on the TLV320AIC23 based Simtec boards.
My question is:
1) I'm not seeing a control to enable/disable the 'Speaker Amp', do I need to change the controls or how this is exported?
2) There are two boards with an GPIO AMP which has two gain controls. This is a TPA2001D1 which provides a gain of 6,12,18 or 23.5dB depending on the GPIO settings.
[ http://focus.ti.com/docs/prod/folders/print/tpa2001d1.html ]
This driver has yet to be tested on all our boards.
/* sound/soc/s3c24xx/s3c24xx_simtec.c * * Copyright 2009 Simtec Electronics * * 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. */
#include <linux/module.h> #include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <linux/clk.h> #include <linux/i2c.h>
#include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h>
#include <plat/audio-simtec.h>
#include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h"
#include "../codecs/tlv320aic23.h"
/* supported machines: * * Machine Connections AMP * ------- ----------- --- * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) * VR1000 HPOUT, LIN None * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) */
static struct s3c24xx_audio_simtec_pdata *pdata; static struct clk *xtal_clk;
/** * simtec_spk_event - handle power management event for speaker * @w: The widget for this event * @k: The sound control the event is for. * @event; The event being processed. * * Handle power management events to deal with the state of the * GPIO AMP attached to some boards. */ static int simtec_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { int value = SND_SOC_DAPM_EVENT_ON(event) ? 0 : 1;
gpio_set_value(pdata->amp_gpio, value); return 0; }
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, simtec_spk_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), };
static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "LHPOUT"}, { "Headphone Jack", NULL, "RHPOUT"},
{ "Line Out", NULL, "LLOUT" }, { "Line Out", NULL, "RLOUT" },
{ "LLINEIN", NULL, "Line In"}, { "RLINEIN", NULL, "Line In"},
{ "MICIN", NULL, "Mic Jack"}, };
static const struct snd_soc_dapm_route amp_map[] = { { "Speaker Amp", NULL, "RHPOUT" }, { "Speaker Amp", NULL, "LHPOUT" }, };
static int spk_gain;
/** * speaker_gain_get - read the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be updated. * * Read the value for the AMP gain control. */ static int speaker_gain_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.integer.value[0] = spk_gain; return 0; }
/** * speaker_gain_put - set the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be set. * * Set the value of the speaker gain from the specified * @ucontrol setting. */ static int speaker_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int value = ucontrol->value.integer.value[0];
spk_gain = value;
gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1);
return 0; }
static const struct snd_kcontrol_new amp_controls[] = { SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, speaker_gain_get, speaker_gain_put), };
/** * simtec_tlv320aic23_init - initialise and add controls * @codec; The codec instance to attach to. * * Attach our controls and configure the necessary codec * mappings for our sound card instance. */ static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
if (pdata->amp_gpio > 0) { printk(KERN_DEBUG "%s: adding amp routes\n", __func__); snd_soc_dapm_add_routes(codec, amp_map, ARRAY_SIZE(amp_map)); snd_soc_dapm_enable_pin(codec, "Speaker Amp"); }
if (pdata->amp_gain[0] > 0) { printk(KERN_DEBUG "%s: adding amp controls\n", __func__); snd_soc_add_controls(codec, amp_controls, ARRAY_SIZE(amp_controls)); }
snd_soc_dapm_enable_pin(codec, "Headphone Jack"); snd_soc_dapm_enable_pin(codec, "Line In"); snd_soc_dapm_enable_pin(codec, "Line Out"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_sync(codec);
return 0; }
#define CODEC_CLOCK 12000000
/** * simtec_hw_params - update hardware parameters * @substream: The audio substream instance. * @params: The parameters requested. * * Update the codec data routing and configuration settings * from the supplied data. */ static int simtec_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->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret;
/* Set the CODEC as the bus clock master, I2S */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set cpu dai format\n", __func__); return ret; }
/* Set the CODEC as the bus clock master */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set codec dai format\n", __func__); return ret; }
ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (ret) { printk(KERN_ERR "%s: failed setting codec sysclk\n", __func__); return ret; }
if (pdata->use_mpllin) { ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, 0, SND_SOC_CLOCK_OUT);
if (ret) { printk(KERN_ERR "%s: failed to set MPLLin as clksrc\n", __func__); return ret; } }
if (pdata->output_cdclk) { int cdclk_scale;
cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; cdclk_scale--;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, cdclk_scale); }
return 0; }
static int simtec_hw_startup(struct snd_pcm_substream *substream) { /* call any board supplied startup code, this currently only * covers the bast/vr1000 which have a CPLD in the way of the * LRCLK */ if (pdata->startup) pdata->startup();
return 0; }
static struct snd_soc_ops simtec_ops = { .startup = simtec_hw_startup, .hw_params = simtec_hw_params, };
static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .cpu_dai = &s3c24xx_i2s_dai, .codec_dai = &tlv320aic23_dai, .init = simtec_tlv320aic23_init, .ops = &simtec_ops, };
/* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .platform = &s3c24xx_soc_platform, .dai_link = &simtec_dai_aic23, .num_links = 1, };
/* simtec audio subsystem */ static struct snd_soc_device simtec_snd_devdata_aic23 = { .card = &snd_soc_machine_simtec_aic23, .codec_dev = &soc_codec_dev_tlv320aic23, };
/** * attach_gpio_amp - get and configure the necessary gpios * @dev: The device we're probing. * @pd: The platform data supplied by the board. * * If there is a GPIO based amplifier attached to the board, claim * the necessary GPIO lines for it, and set default values. */ static int attach_gpio_amp(struct device *dev, struct s3c24xx_audio_simtec_pdata *pd) { int ret;
/* attach gpio amp gain (if any) */ if (pdata->amp_gain[0] > 0) { ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); return ret; }
ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); gpio_free(pdata->amp_gain[0]); return ret; }
gpio_direction_output(pd->amp_gain[0], 0); gpio_direction_output(pd->amp_gain[1], 0); }
/* note, curently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { dev_err(dev, "cannot get amp gpio %d (%d)\n", pd->amp_gpio, ret); goto err_amp; }
/* start with amp off */ gpio_direction_output(pd->amp_gpio, 1); }
return 0;
err_amp: if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); }
return ret; }
static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) { if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); }
if (pd->amp_gpio > 0) gpio_free(pd->amp_gpio); }
static int __devinit simtec_audio_probe(struct platform_device *pdev) { struct platform_device *snd_dev; int ret;
dev_info(&pdev->dev, "probe called\n");
pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "no platform data supplied\n"); return -EINVAL; }
xtal_clk = clk_get(&pdev->dev, "xtal"); if (IS_ERR(xtal_clk)) { dev_err(&pdev->dev, "could not get clkout0\n"); return -EINVAL; }
dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk));
ret = attach_gpio_amp(&pdev->dev, pdata); if (ret) goto err_clk;
snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); ret = -ENOMEM; goto err_gpio; }
platform_set_drvdata(snd_dev, &simtec_snd_devdata_aic23); simtec_snd_devdata_aic23.dev = &snd_dev->dev;
ret = platform_device_add(snd_dev); if (ret) { dev_err(&pdev->dev, "failed to add soc-audio dev\n"); platform_device_put(snd_dev); goto err_clk; }
platform_set_drvdata(pdev, snd_dev); return 0;
err_gpio: detach_gpio_amp(pdata);
err_clk: clk_put(xtal_clk); return ret; }
static int __devexit simtec_audio_remove(struct platform_device *pdev) { struct platform_device *snd_dev = platform_get_drvdata(pdev);
platform_device_unregister(snd_dev);
detach_gpio_amp(pdata); clk_put(xtal_clk); return 0; }
static struct platform_driver simtec_audio_platdrv = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-tlv320aic23", }, .probe = simtec_audio_probe, .remove = __devexit_p(simtec_audio_remove), };
MODULE_ALIAS("platform:s3c24xx-tlv320aic23");
static int __init simtec_init(void) { return platform_driver_register(&simtec_audio_platdrv); }
static void __exit simtec_exit(void) { platform_driver_unregister(&simtec_audio_platdrv); }
module_init(simtec_init); module_exit(simtec_exit);
MODULE_AUTHOR("Ben Dooks ben@simtec.co.uk"); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL");
On Wed, Jun 24, 2009 at 12:11:55PM +0100, Ben Dooks wrote:
I'd like some comments on the following driver which is currently being developed to get ASoC working on the TLV320AIC23 based Simtec boards.
Please do remember to CC maintainers on things.
- I'm not seeing a control to enable/disable the 'Speaker Amp', do I need to change the controls or how this is exported?
If you need to give manual control of this to user space use a DAPM_SOC_PIN_SWITCH widget. Normally there's no need to do this since it falls out of the internal routing within the CODEC and having an explicit control for the output itself just adds an extra knob for userspace to twiddle. Depending on the technology it may not even be desirable since having inputs connected with the amplifiers powered off can leave audible leakage paths.
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, simtec_spk_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
This looks like a SND_SOC_DAPM_SPK(); using that will help with sequencing.
static const struct snd_soc_dapm_route audio_map[] = {
This wants a better name - base_map or something?
if (pdata->amp_gpio > 0) { printk(KERN_DEBUG "%s: adding amp routes\n", __func__);
pr_dbg() or dev_dbg()
static int simtec_hw_startup(struct snd_pcm_substream *substream) { /* call any board supplied startup code, this currently only * covers the bast/vr1000 which have a CPLD in the way of the * LRCLK */ if (pdata->startup) pdata->startup();
return 0; }
This looks odd. Does this really need to be done each time the audio device can be opened or should it be done once when the driver is being set up? There doesn't appear to be any similar code to reverse the startup() on teardown so I suspect that just doing it when the driver starts should be fine.
/* attach gpio amp gain (if any) */ if (pdata->amp_gain[0] > 0) { ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); return ret; }
ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n");
gain1.
static int __devinit simtec_audio_probe(struct platform_device *pdev) { struct platform_device *snd_dev; int ret;
dev_info(&pdev->dev, "probe called\n");
dev_dbg() at most.
pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "no platform data supplied\n"); return -EINVAL; }
Every individual bit of platform data looks optional; this suggests that either additional validation is desirable here or that platform data itself should be optional.
On Thu, Jun 25, 2009 at 12:26:59PM +0100, Mark Brown wrote:
On Wed, Jun 24, 2009 at 12:11:55PM +0100, Ben Dooks wrote:
I'd like some comments on the following driver which is currently being developed to get ASoC working on the TLV320AIC23 based Simtec boards.
Please do remember to CC maintainers on things.
- I'm not seeing a control to enable/disable the 'Speaker Amp', do I need to change the controls or how this is exported?
If you need to give manual control of this to user space use a DAPM_SOC_PIN_SWITCH widget. Normally there's no need to do this since it falls out of the internal routing within the CODEC and having an explicit control for the output itself just adds an extra knob for userspace to twiddle. Depending on the technology it may not even be desirable since having inputs connected with the amplifiers powered off can leave audible leakage paths.
We've not seen any problems with out boards, then the amps are coupled with an R/C network so leakage from HP->AMP should be negligible.
Our old ALSA drivers always gave the user the choice to have the speaker powered up or not.
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, simtec_spk_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
This looks like a SND_SOC_DAPM_SPK(); using that will help with sequencing.
thanks, changed. still no output in the alsactl control dump though.
static const struct snd_soc_dapm_route audio_map[] = {
This wants a better name - base_map or something?
thanks, changed.
if (pdata->amp_gpio > 0) { printk(KERN_DEBUG "%s: adding amp routes\n", __func__);
pr_dbg() or dev_dbg()
changed to pr_debug, we're not keeping the platform device around after the probe.
static int simtec_hw_startup(struct snd_pcm_substream *substream) { /* call any board supplied startup code, this currently only * covers the bast/vr1000 which have a CPLD in the way of the * LRCLK */ if (pdata->startup) pdata->startup();
return 0; }
This looks odd. Does this really need to be done each time the audio device can be opened or should it be done once when the driver is being set up? There doesn't appear to be any similar code to reverse the startup() on teardown so I suspect that just doing it when the driver starts should be fine.
I'll have to add it in the resume path as well.
/* attach gpio amp gain (if any) */ if (pdata->amp_gain[0] > 0) { ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); return ret; }
ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n");gain1.
thanks, changed.
static int __devinit simtec_audio_probe(struct platform_device *pdev) { struct platform_device *snd_dev; int ret;
dev_info(&pdev->dev, "probe called\n");
dev_dbg() at most.
pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "no platform data supplied\n"); return -EINVAL; }
Every individual bit of platform data looks optional; this suggests that either additional validation is desirable here or that platform data itself should be optional.
As such, pretty much all the systems we've got pass platform data for some from of information, so I decided that a lack of it was a good enough indication that there was something wrong.
On Fri, Jun 26, 2009 at 09:46:26AM +0100, Ben Dooks wrote:
On Thu, Jun 25, 2009 at 12:26:59PM +0100, Mark Brown wrote:
If you need to give manual control of this to user space use a DAPM_SOC_PIN_SWITCH widget. Normally there's no need to do this since
...
This looks like a SND_SOC_DAPM_SPK(); using that will help with sequencing.
thanks, changed. still no output in the alsactl control dump though.
Like I say, you need to use a pin switch if you want to add manual control.
I'll have to add it in the resume path as well.
So on probe and resume, then?
participants (3)
-
Ben Dooks -
Ben Dooks -
Mark Brown