ALSA: ASoC: cs4271: add optional soft reset workaround

The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.

One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.

This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

1 files changed, 34 insertions, 0 deletions

diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index ac8742a1f25a..2415a4118dbd 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -167,6 +167,8 @@ struct cs4271_private {
         int                             gpio_nreset;
         /* GPIO that disable serial bus, if any */
         int                             gpio_disable;
+        /* enable soft reset workaround */
+        bool                            enable_soft_reset;
 };
 
 /*
@@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
         int i, ret;
         unsigned int ratio, val;
 
+        if (cs4271->enable_soft_reset) {
+                /*
+                 * Put the codec in soft reset and back again in case it's not
+                 * currently streaming data. This way of bringing the codec in
+                 * sync to the current clocks is not explicitly documented in
+                 * the data sheet, but it seems to work fine, and in contrast
+                 * to a read hardware reset, we don't have to sync back all
+                 * registers every time.
+                 */
+
+                if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+                     !dai->capture_active) ||
+                    (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+                     !dai->playback_active)) {
+                        ret = snd_soc_update_bits(codec, CS4271_MODE2,
+                                                  CS4271_MODE2_PDN,
+                                                  CS4271_MODE2_PDN);
+                        if (ret < 0)
+                                return ret;
+
+                        ret = snd_soc_update_bits(codec, CS4271_MODE2,
+                                                  CS4271_MODE2_PDN, 0);
+                        if (ret < 0)
+                                return ret;
+                }
+        }
+
         cs4271->rate = params_rate(params);
 
         /* Configure DAC */
@@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec)
                 if (of_get_property(codec->dev->of_node,
                                      "cirrus,amutec-eq-bmutec", NULL))
                         amutec_eq_bmutec = true;
+
+                if (of_get_property(codec->dev->of_node,
+                                     "cirrus,enable-soft-reset", NULL))
+                        cs4271->enable_soft_reset = true;
         }
 #endif
 
@@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec)
                         gpio_nreset = cs4271plat->gpio_nreset;
 
                 amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec;
+                cs4271->enable_soft_reset = cs4271plat->enable_soft_reset;
         }
 
         if (gpio_nreset >= 0)