Merge tag 'asoc-v4.18' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v4.18

This is a very big update, mainly due to a huge set of new drivers some
of which are individually very large.  We also have a lot of fixes for
the topology stuff, several of the users have stepped up and fixed some
the serious issues there, and continued progress on the transition away
from CODEC specific drivers to generic component drivers.

 - Many fixes for the topology code, including fixes for the half done
   v4 ABI compatibility from Guenter Roeck and other ABI fixes from
   Kirill Marinushkin.
 - Lots of cleanup for Intel platforms based on Realtek CODECs from Hans
   de Goode.
 - More followups on removing legacy CODEC things and transitioning to
   components from Morimoto-san.
 - Conversion of OMAP DMA to the new, more standard SDMA-PCM driver.
 - A series of fixes and updates to the rather elderly Cirrus Logic SoC
   drivers from Alexander Sverdlin.
 - Qualcomm DSP support from Srinivas Kandagatla.
 - New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
   MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306 and
   RT5668 and TI TSCS454

1 files changed, 14 insertions, 7 deletions

diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 71e5f31fa306..9cfefe44a75f 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -80,9 +80,9 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
 	else
 		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
 	if (line1l)
-		snd_soc_dapm_enable_pin_unlocked(dapm, "LINE1L");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
 	else
-		snd_soc_dapm_disable_pin_unlocked(dapm, "LINE1L");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
 
 	if (n810_dmic_func)
 		snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
@@ -222,6 +222,7 @@ static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
 	SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
 	SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
 	SND_SOC_DAPM_MIC("DMic", NULL),
+	SND_SOC_DAPM_MIC("HS Mic", NULL),
 };
 
 static const struct snd_soc_dapm_route audio_map[] = {
@@ -231,8 +232,14 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Ext Spk", NULL, "LLOUT"},
 	{"Ext Spk", NULL, "RLOUT"},
 
-	{"DMic Rate 64", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "DMic"},
+	{"DMic Rate 64", NULL, "DMic"},
+	{"DMic", NULL, "Mic Bias"},
+
+	/*
+	 * Note that the mic bias is coming from Retu/Vilma and we don't have
+	 * control over it atm. The analog HS mic is not working. <- TODO
+	 */
+	{"LINE1L", NULL, "HS Mic"},
 };
 
 static const char *spk_function[] = {"Off", "On"};
@@ -257,9 +264,9 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
 static struct snd_soc_dai_link n810_dai = {
 	.name = "TLV320AIC33",
 	.stream_name = "AIC33",
-	.cpu_dai_name = "omap-mcbsp.2",
-	.platform_name = "omap-mcbsp.2",
-	.codec_name = "tlv320aic3x-codec.2-0018",
+	.cpu_dai_name = "48076000.mcbsp",
+	.platform_name = "48076000.mcbsp",
+	.codec_name = "tlv320aic3x-codec.1-0018",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 		   SND_SOC_DAIFMT_CBM_CFM,