From: Philipp Zabel <philipp.zabel@gmail.com>
To: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: alsa-devel@alsa-project.org, Philipp Zabel <philipp.zabel@gmail.com>
Subject: [PATCH 4/4] ASoC: UDA1380: change decimator/interpolator register handling
Date: Tue, 3 Mar 2009 16:10:54 +0100 [thread overview]
Message-ID: <1236093054-1952-4-git-send-email-philipp.zabel@gmail.com> (raw)
In-Reply-To: <1236093054-1952-1-git-send-email-philipp.zabel@gmail.com>
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).
* Queue work in the alsa PCM_START .trigger to flush registers
as soon as the link is running. This replaces the .prepare
and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
---
sound/soc/codecs/uda1380.c | 102 +++++++++++++++++++++-----------------------
1 files changed, 48 insertions(+), 54 deletions(-)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a6987d7..1b10f48 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -25,6 +25,7 @@
#include <linux/ioctl.h>
#include <linux/delay.h>
#include <linux/i2c.h>
+#include <linux/workqueue.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/initval.h>
@@ -35,6 +36,9 @@
#include "uda1380.h"
+static struct work_struct uda1380_work;
+static struct snd_soc_codec *uda1380_codec;
+
/*
* uda1380 register cache
*/
@@ -50,6 +54,8 @@ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
0x0000, 0x8000, 0x0002, 0x0000,
};
+static unsigned long uda1380_cache_dirty;
+
/*
* read uda1380 register cache
*/
@@ -71,8 +77,11 @@ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
+
if (reg >= UDA1380_CACHEREGNUM)
return;
+ if ((reg >= 0x10) && (cache[reg] != value))
+ set_bit(reg - 0x10, &uda1380_cache_dirty);
cache[reg] = value;
}
@@ -111,6 +120,8 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
(data[0]<<8) | data[1]);
return -EIO;
}
+ if (reg >= 0x10)
+ clear_bit(reg - 0x10, &uda1380_cache_dirty);
return 0;
} else
return -EIO;
@@ -118,6 +129,20 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
+static void uda1380_flush_work(struct work_struct *work)
+{
+ int bit, reg;
+
+ for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+ reg = 0x10 + bit;
+ pr_debug("uda1380: flush reg %x val %x:\n", reg,
+ uda1380_read_reg_cache(uda1380_codec, reg));
+ uda1380_write(uda1380_codec, reg,
+ uda1380_read_reg_cache(uda1380_codec, reg));
+ clear_bit(bit, &uda1380_cache_dirty);
+ }
+}
+
/* declarations of ALSA reg_elem_REAL controls */
static const char *uda1380_deemp[] = {
"None",
@@ -252,7 +277,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
- SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
@@ -443,41 +467,28 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
return 0;
}
-/*
- * Flush reg cache
- * We can only write the interpolator and decimator registers
- * when the DAI is being clocked by the CPU DAI. It's up to the
- * machine and cpu DAI driver to do this before we are called.
- */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- int reg, reg_start, reg_end, clk;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- reg_start = UDA1380_MVOL;
- reg_end = UDA1380_MIXER;
- } else {
- reg_start = UDA1380_DEC;
- reg_end = UDA1380_AGC;
- }
-
- /* FIXME disable DAC_CLK */
- clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
- uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
-
- for (reg = reg_start; reg <= reg_end; reg++) {
- pr_debug("uda1380: flush reg %x val %x:", reg,
- uda1380_read_reg_cache(codec, reg));
- uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+ int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ uda1380_write_reg_cache(codec, UDA1380_MIXER,
+ mixer & ~R14_SILENCE);
+ schedule_work(&uda1380_work);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ uda1380_write_reg_cache(codec, UDA1380_MIXER,
+ mixer | R14_SILENCE);
+ schedule_work(&uda1380_work);
+ break;
}
-
- /* FIXME restore DAC_CLK */
- uda1380_write(codec, UDA1380_CLK, clk);
-
return 0;
}
@@ -543,24 +554,6 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
uda1380_write(codec, UDA1380_CLK, clk);
}
-static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
-{
- struct snd_soc_codec *codec = codec_dai->codec;
- u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
-
- /* FIXME: mute(codec,0) is called when the magician clock is already
- * set to WSPLL, but for some unknown reason writing to interpolator
- * registers works only when clocked by SYSCLK */
- u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
- uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
- if (mute)
- uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
- else
- uda1380_write(codec, UDA1380_DEEMP, mute_reg);
- uda1380_write(codec, UDA1380_CLK, clk);
- return 0;
-}
-
static int uda1380_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -602,10 +595,9 @@ struct snd_soc_dai uda1380_dai[] = {
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
+ .trigger = uda1380_trigger,
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
- .prepare = uda1380_pcm_prepare,
- .digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt_both,
},
},
@@ -619,10 +611,9 @@ struct snd_soc_dai uda1380_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
+ .trigger = uda1380_trigger,
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
- .prepare = uda1380_pcm_prepare,
- .digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt_playback,
},
},
@@ -636,9 +627,9 @@ struct snd_soc_dai uda1380_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
+ .trigger = uda1380_trigger,
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
- .prepare = uda1380_pcm_prepare,
.set_fmt = uda1380_set_dai_fmt_capture,
},
},
@@ -697,6 +688,9 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
codec->reg_cache_step = 1;
uda1380_reset(codec);
+ uda1380_codec = codec;
+ INIT_WORK(&uda1380_work, uda1380_flush_work);
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
--
1.6.1.3
next prev parent reply other threads:[~2009-03-03 15:11 UTC|newest]
Thread overview: 13+ messages / expand[flat|nested] mbox.gz Atom feed top
2009-03-03 15:10 [PATCH 1/4] ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite Philipp Zabel
2009-03-03 15:10 ` [PATCH 2/4] ASoC: UDA1380: DATAI is slave only Philipp Zabel
2009-03-03 15:50 ` Mark Brown
2009-03-03 15:56 ` pHilipp Zabel
2009-03-03 15:58 ` Mark Brown
2009-03-03 15:10 ` [PATCH 3/4] ASoC: Remove version display from the UDA1380 driver Philipp Zabel
2009-03-03 15:52 ` Mark Brown
2009-03-03 15:10 ` Philipp Zabel [this message]
2009-03-03 15:48 ` [PATCH 1/4] ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite Mark Brown
2009-03-04 1:34 ` Daniel Mack
2009-03-04 14:29 ` Mark Brown
2009-03-04 20:45 ` pHilipp Zabel
2009-03-05 16:06 ` pHilipp Zabel
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