Merge branch 'topic/asoc' into for-linus

1 files changed, 43 insertions, 50 deletions

diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 3686417f5ea5..e0d45fdaa750 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -43,9 +43,6 @@ struct  cs42l73_private {
 };
 
 static const struct reg_default cs42l73_reg_defaults[] = {
-        { 1, 0x42 },    /* r01  - Device ID A&B */
-        { 2, 0xA7 },    /* r02  - Device ID C&D */
-        { 3, 0x30 },    /* r03  - Device ID E */
         { 6, 0xF1 },    /* r06  - Power Ctl 1 */
         { 7, 0xDF },    /* r07  - Power Ctl 2 */
         { 8, 0x3F },    /* r08  - Power Ctl 3 */
@@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl =
 
 static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
         SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume",
-                        CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7,
-                        0xffffffC1, 0x0C, hpaloa_tlv),
+                        CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0,
+                        0x41, 0x4B, hpaloa_tlv),
 
         SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL,
-                        CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv),
+                        CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
 
         SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
-                        CS42L73_MICBPREPGABVOL, 5, 0xffffff35,
-                        0x34, micpga_tlv),
+                        CS42L73_MICBPREPGABVOL, 5, 0x34,
+                        0x24, micpga_tlv),
 
         SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,
                         CS42L73_MICBPREPGABVOL, 6, 1, 1),
 
         SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL,
-                        CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv),
+                        CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv),
 
         SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume",
-                        CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5,
-                        0xE4, hl_tlv),
+                        CS42L73_HLADVOL, CS42L73_HLBDVOL,
+                        0, 0x34, 0xE4, hl_tlv),
 
         SOC_SINGLE_TLV("ADC A Boost Volume",
                         CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv),
 
         SOC_SINGLE_TLV("ADC B Boost Volume",
-                        CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
+                       CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv),
 
-        SOC_SINGLE_TLV("Speakerphone Digital Playback Volume",
-                        CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv),
+        SOC_SINGLE_SX_TLV("Speakerphone Digital Volume",
+                            CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv),
 
-        SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume",
-                        CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv),
+        SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume",
+                            CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv),
 
         SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL,
                         CS42L73_HPBAVOL, 7, 1, 1),
@@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
         SND_SOC_DAPM_INPUT("MIC2"),
         SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0),
 
-        SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture",  0,
+        SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL,  0,
                         CS42L73_PWRCTL2, 1, 1),
-        SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture",  0,
+        SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL,  0,
                         CS42L73_PWRCTL2, 1, 1),
-        SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture",  0,
+        SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL,  0,
                         CS42L73_PWRCTL2, 3, 1),
-        SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture",  0,
+        SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL,  0,
                         CS42L73_PWRCTL2, 3, 1),
-        SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture",  0,
+        SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL,  0,
                         CS42L73_PWRCTL2, 4, 1),
-        SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture",  0,
+        SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL,  0,
                         CS42L73_PWRCTL2, 4, 1),
 
         SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
         SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
         SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 
-        SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
                                 CS42L73_PWRCTL2, 0, 1),
-        SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0,
                                 CS42L73_PWRCTL2, 0, 1),
-        SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0,
                                 CS42L73_PWRCTL2, 0, 1),
 
-        SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0,
                                 CS42L73_PWRCTL2, 2, 1),
-        SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0,
                                 CS42L73_PWRCTL2, 2, 1),
-        SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
                                 CS42L73_PWRCTL2, 2, 1),
 
-        SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
+        SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
                                 CS42L73_PWRCTL2, 4, 1),
 
         SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
         {"HL Left Mixer", NULL, "VSPIN"},
         {"HL Right Mixer", NULL, "VSPIN"},
 
+        {"ASPINL", NULL, "ASP Playback"},
+        {"ASPINM", NULL, "ASP Playback"},
+        {"ASPINR", NULL, "ASP Playback"},
+        {"XSPINL", NULL, "XSP Playback"},
+        {"XSPINM", NULL, "XSP Playback"},
+        {"XSPINR", NULL, "XSP Playback"},
+        {"VSPIN", NULL, "VSP Playback"},
+
         /* Capture Paths */
         {"MIC1", NULL, "MIC1 Bias"},
         {"PGA Left Mux", "Mic 1", "MIC1"},
@@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 
         {"VSPOUTL", NULL, "VSPL Output Mixer"},
         {"VSPOUTR", NULL, "VSPR Output Mixer"},
+
+        {"ASP Capture", NULL, "ASPOUTL"},
+        {"ASP Capture", NULL, "ASPOUTR"},
+        {"XSP Capture", NULL, "XSPOUTL"},
+        {"XSP Capture", NULL, "XSPOUTR"},
+        {"VSP Capture", NULL, "VSPOUTL"},
+        {"VSP Capture", NULL, "VSPOUTR"},
 };
 
 struct cs42l73_mclk_div {
@@ -1091,8 +1103,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
                                  struct snd_pcm_hw_params *params,
                                  struct snd_soc_dai *dai)
 {
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_codec *codec = dai->codec;
         struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
         int id = dai->id;
         int mclk_coeff;
@@ -1429,25 +1440,7 @@ static struct i2c_driver cs42l73_i2c_driver = {
 
 };
 
-static int __init cs42l73_modinit(void)
-{
-        int ret;
-        ret = i2c_add_driver(&cs42l73_i2c_driver);
-        if (ret != 0) {
-                pr_err("Failed to register CS42L73 I2C driver: %d\n", ret);
-                return ret;
-        }
-        return 0;
-}
-
-module_init(cs42l73_modinit);
-
-static void __exit cs42l73_exit(void)
-{
-        i2c_del_driver(&cs42l73_i2c_driver);
-}
-
-module_exit(cs42l73_exit);
+module_i2c_driver(cs42l73_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC CS42L73 driver");
 MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>");