* wm9713.c -- ALSA Soc WM9713 codec support
*
* Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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.
*
- * Revision history
- * 4th Feb 2006 Initial version.
- *
* Features:-
*
* o Support for AC97 Codec, Voice DAC and Aux DAC
SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
-SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1),
-SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1),
+SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
+SND_SOC_DAPM_ADC("Left HiFi ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Right HiFi ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Left Voice ADC", "Left Voice Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Right Voice ADC", "Right Voice Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
SND_SOC_DAPM_VMID("VMID"),
};
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* left HP mixer */
{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
{"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
/* left ADC */
{"Left ADC", NULL, "Left Capture Source"},
+ {"Left Voice ADC", NULL, "Left ADC"},
+ {"Left HiFi ADC", NULL, "Left ADC"},
/* right ADC */
{"Right ADC", NULL, "Right Capture Source"},
+ {"Right Voice ADC", NULL, "Right ADC"},
+ {"Right HiFi ADC", NULL, "Right ADC"},
/* mic */
{"Mic A Pre Amp", NULL, "Mic A Source"},
{"Capture Mono Mux", "Stereo", "Capture Mixer"},
{"Capture Mono Mux", "Left", "Left Capture Source"},
{"Capture Mono Mux", "Right", "Right Capture Source"},
-
- {NULL, NULL, NULL},
};
static int wm9713_add_widgets(struct snd_soc_codec *codec)
{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+ ARRAY_SIZE(wm9713_dapm_widgets));
- /* set up audio path audio_mapnects */
- for (i = 0; audio_map[i][0] != NULL; i++)
- snd_soc_dapm_connect_input(codec, audio_map[i][0],
- audio_map[i][1], audio_map[i][2]);
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
return 0;
}
-static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
* Tristate the PCM DAI lines, tristate can be disabled by calling
* wm9713_set_dai_fmt()
*/
-static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
int tristate)
{
struct snd_soc_codec *codec = codec_dai->codec;
* Configure WM9713 clock dividers.
* Voice DAC needs 256 FS
*/
-static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
return 0;
}
-static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
gpio |= 0x0018;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- reg |= 0x0200;
+ reg |= 0x2000;
gpio |= 0x001a;
break;
case SND_SOC_DAIFMT_CBS_CFM:
static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
- u16 status;
-
- /* Gracefully shut down the voice interface. */
- status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
- ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
- schedule_timeout_interruptible(msecs_to_jiffies(1));
- ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
- ac97_write(codec, AC97_EXTENDED_MID, status);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 status;
+
+ /* Gracefully shut down the voice interface. */
+ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
+ ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+ schedule_timeout_interruptible(msecs_to_jiffies(1));
+ ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+ ac97_write(codec, AC97_EXTENDED_MID, status);
}
static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
}
-#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
- SNDRV_PCM_RATE_48000)
+#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
#define WM9713_PCM_FORMATS \
(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
SNDRV_PCM_FORMAT_S24_LE)
-struct snd_soc_codec_dai wm9713_dai[] = {
+struct snd_soc_dai wm9713_dai[] = {
{
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97_BUS,
.stream_name = "Voice Playback",
.channels_min = 1,
.channels_max = 1,
- .rates = WM9713_RATES,
+ .rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.capture = {
.stream_name = "Voice Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = WM9713_RATES,
+ .rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.ops = {
.hw_params = wm9713_pcm_hw_params,
{
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
- if (!(ac97_read(codec, 0) & 0x8000))
+ if (ac97_read(codec, 0) == wm9713_reg[0])
return 1;
}
soc_ac97_ops.reset(codec->ac97);
- if (ac97_read(codec, 0) & 0x8000)
+ if (ac97_read(codec, 0) != wm9713_reg[0])
return -EIO;
return 0;
}
EXPORT_SYMBOL_GPL(wm9713_reset);
-static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9713_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
{
u16 reg;
- switch (event) {
- case SNDRV_CTL_POWER_D0: /* full On */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
/* enable thermal shutdown */
reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
ac97_write(codec, AC97_EXTENDED_MID, reg);
break;
- case SNDRV_CTL_POWER_D1: /* partial On */
- case SNDRV_CTL_POWER_D2: /* partial On */
+ case SND_SOC_BIAS_PREPARE:
break;
- case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ case SND_SOC_BIAS_STANDBY:
/* enable master bias and vmid */
reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
ac97_write(codec, AC97_EXTENDED_MID, reg);
ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
- case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ case SND_SOC_BIAS_OFF:
/* disable everything including AC link */
ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->dapm_state = event;
+ codec->bias_level = level;
return 0;
}
return ret;
}
- wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* do we need to re-start the PLL ? */
if (wm9713->pll_out)
}
}
- if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
- wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
return ret;
}
codec->num_dai = ARRAY_SIZE(wm9713_dai);
codec->write = ac97_write;
codec->read = ac97_read;
- codec->dapm_event = wm9713_dapm_event;
+ codec->set_bias_level = wm9713_set_bias_level;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
goto reset_err;
}
- wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* unmute the adc - move to kcontrol */
reg = ac97_read(codec, AC97_CD) & 0x7fff;
projects / GitHub / mt8127 / android_kernel_alcatel_ttab.git / blobdiff