On 04/28/2014 02:17 PM, Stefan Roese wrote:
From: Jarkko Nikula <jarkko.nik...@bitmer.com>
This codec driver template represents an I2C controlled multichannel audio
codec that has many typical ASoC codec driver features like volume controls,
mixer stages, mux selection, output power control, in-codec audio routings,
codec bias management and DAI link configuration.
Updates from Stefan Roese, 2014-04-28:
Port the HA DSP codec driver to Linux v3.15-rc. This includes
support for DT based probing. No platform-data code is needed
any more, DT nodes are sufficient.
Signed-off-by: Jarkko Nikula <jarkko.nik...@bitmer.com>
Signed-off-by: Stefan Roese <s...@denx.de>
Cc: Thorsten Eisbein <thorsten.eisb...@head-acoustics.de>
Looks very good. Couple of bits inline.
[...]
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
There seem to be a couple of includes here that are not really needed.
+
+#include "ha-dsp.h"
[...]
+static const char *ha_dsp_mode_texts[] = {"Mode 1", "Mode 2"};
const char *const
+static SOC_ENUM_SINGLE_DECL(ha_dsp_mode_enum, HA_DSP_CTRL, 0,
+ ha_dsp_mode_texts);
+
+/* Monitor output mux selection */
+static const char *ha_dsp_monitor_texts[] = {"Off", "ADC", "DAC"};
const char *const
+static SOC_ENUM_SINGLE_DECL(ha_dsp_monitor_enum, HA_DSP_CTRL, 1,
+ ha_dsp_monitor_texts);
+
[...]
+static const struct snd_soc_dapm_widget ha_dsp_widgets[] = {
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MIXER("OUT1 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out1_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out1_mixer_controls)),
There is the SOC_MIXER_ARRAY() helper macro that you can use here and below.
+ SND_SOC_DAPM_MIXER("OUT2 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out2_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT3 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out3_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT4 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out4_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out4_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT5 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out5_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out5_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT6 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out6_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out6_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT7 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out7_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out7_mixer_controls)),
+ SND_SOC_DAPM_MIXER("OUT8 Mixer", SND_SOC_NOPM, 0, 0,
+ &ha_dsp_out8_mixer_controls[0],
+ ARRAY_SIZE(ha_dsp_out8_mixer_controls)),
[...]
+static int ha_dsp_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;
A codec should never look at the pcm_runtime. The proper way to get a
pointer to the codec in dai callbacks is dai->codec. Or just use dai->dev below.
+
+ 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 ha_dsp_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ dev_dbg(codec->dev, "Changing bias from %d to %d\n",
+ codec->dapm.bias_level, level);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* Set PLL on */
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* Set power on, Set PLL off */
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Set power down */
+ break;
+ }
+ codec->dapm.bias_level = level;
If you don't do anything in set_bias_level, just don't implement the
function. The default implementation if no callback is specified is to set
the bias_level to the requested level.
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops ha_dsp_dai_ops = {
const
+ .hw_params = ha_dsp_hw_params,
+ .set_fmt = ha_dsp_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver ha_dsp_dai = {
+ .name = "ha-dsp-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ /* We use only 32 Bits for Audio */
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 16,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ /* We use only 32 Bits for Audio */
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &ha_dsp_dai_ops,
+};
+
+static int ha_dsp_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
Why do you want to use the regmap instance of the parent? That doesn't make
sense given that you initialized a remgap instance for the device itself.
+ ret = snd_soc_codec_set_cache_io(codec, codec->control_data);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int ha_dsp_remove(struct snd_soc_codec *codec)
+{
+ snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET);
To get the codec into a well know state it is best practice to also reset it
when probing the device.
+
+ return 0;
+}
+
[...]
+static int ha_dsp_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap *regmap;
+ int ret;
+
+ regmap = devm_regmap_init_i2c(client, &ha_dsp_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ha_dsp,
+ &ha_dsp_dai, 1);
Just return snd_soc_register_codec(...)
+
+ return ret;
+}
[...]
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