[alsa-devel] [PATCH] ASoC: wm8962: Provide a .hw_free function

[Fabio Estevam]

From: Fabio Estevam <fabio.estevam at freescale.com>

Currently after playing any audio file, we get the following error message:

$ aplay clarinet.wav 
Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
$ wm8962 0-001a: Unsupported sysclk ratio 544

This error message appears about 5 seconds after the audio playback has 
finished.

In order to avoid such error message, let's add a .hw_free function which will
zero out wm8962->lrclk and wm8962->bclk, which according to the code below:

	if (!wm8962->bclk || !wm8962->lrclk) {
		dev_dbg(codec->dev, "No audio clocks configured\n");
		return;
	}

,will cause the wm8962_configure_bclk() function not to search a valid sysclk 
ratio after playback has finished.

Suggested-by: Nicolin Chen <b42378 at freescale.com>
Signed-off-by: Fabio Estevam <fabio.estevam at freescale.com>
---
 sound/soc/codecs/wm8962.c | 13 +++++++++++++
 1 file changed, 13 insertions(+)

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 6a5066c..01033c2 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2914,6 +2914,18 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 				   WM8962_DAC_MUTE, val);
 }
 
+static int wm8962_hw_free(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+	wm8962->lrclk = 0;
+	wm8962->bclk = 0;
+
+	return 0;
+}
+
 #define WM8962_RATES SNDRV_PCM_RATE_8000_96000
 
 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
@@ -2921,6 +2933,7 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 
 static const struct snd_soc_dai_ops wm8962_dai_ops = {
 	.hw_params = wm8962_hw_params,
+	.hw_free = wm8962_hw_free,
 	.set_sysclk = wm8962_set_dai_sysclk,
 	.set_fmt = wm8962_set_dai_fmt,
 	.digital_mute = wm8962_mute,
-- 
1.8.1.2