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-rw-r--r--sound/soc/codecs/wm5110.c27
-rw-r--r--sound/soc/codecs/wm8731.c4
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8962.c13
-rw-r--r--sound/soc/codecs/wm8990.c2
-rw-r--r--sound/soc/codecs/wm_adsp.c10
6 files changed, 26 insertions, 32 deletions
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a6181..0ab2dc296474 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
248ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), 248ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
249ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), 249ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
250 250
251SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
252 ARIZONA_OUT1_OSR_SHIFT, 1, 0),
253SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
254 ARIZONA_OUT2_OSR_SHIFT, 1, 0),
255SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
256 ARIZONA_OUT3_OSR_SHIFT, 1, 0),
257SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
258 ARIZONA_OUT4_OSR_SHIFT, 1, 0),
259SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
260 ARIZONA_OUT5_OSR_SHIFT, 1, 0),
261SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
262 ARIZONA_OUT6_OSR_SHIFT, 1, 0),
263
264SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, 251SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
265 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), 252 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
266SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, 253SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
293 ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 280 ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
294 0xbf, 0, digital_tlv), 281 0xbf, 0, digital_tlv),
295 282
296SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
297 ARIZONA_OUTPUT_PATH_CONFIG_1R,
298 ARIZONA_OUT1L_PGA_VOL_SHIFT,
299 0x34, 0x40, 0, ana_tlv),
300SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
301 ARIZONA_OUTPUT_PATH_CONFIG_2R,
302 ARIZONA_OUT2L_PGA_VOL_SHIFT,
303 0x34, 0x40, 0, ana_tlv),
304SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
305 ARIZONA_OUTPUT_PATH_CONFIG_3R,
306 ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
307
308SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, 283SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
309 ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), 284 ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
310SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, 285SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
@@ -1037,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
1037 { "AEC Loopback", "HPOUT3L", "OUT3L" }, 1012 { "AEC Loopback", "HPOUT3L", "OUT3L" },
1038 { "AEC Loopback", "HPOUT3R", "OUT3R" }, 1013 { "AEC Loopback", "HPOUT3R", "OUT3R" },
1039 { "HPOUT3L", NULL, "OUT3L" }, 1014 { "HPOUT3L", NULL, "OUT3L" },
1040 { "HPOUT3R", NULL, "OUT3L" }, 1015 { "HPOUT3R", NULL, "OUT3R" },
1041 1016
1042 { "AEC Loopback", "SPKOUTL", "OUT4L" }, 1017 { "AEC Loopback", "SPKOUTL", "OUT4L" },
1043 { "SPKOUTLN", NULL, "OUT4L" }, 1018 { "SPKOUTLN", NULL, "OUT4L" },
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759..bc7472c968e3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
447 iface |= 0x0001; 447 iface |= 0x0001;
448 break; 448 break;
449 case SND_SOC_DAIFMT_DSP_A: 449 case SND_SOC_DAIFMT_DSP_A:
450 iface |= 0x0003; 450 iface |= 0x0013;
451 break; 451 break;
452 case SND_SOC_DAIFMT_DSP_B: 452 case SND_SOC_DAIFMT_DSP_B:
453 iface |= 0x0013; 453 iface |= 0x0003;
454 break; 454 break;
455 default: 455 default:
456 return -EINVAL; 456 return -EINVAL;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3938fb1c203e..53bbfac6a83a 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
1444 1444
1445 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 1445 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
1446 case SND_SOC_DAIFMT_DSP_B: 1446 case SND_SOC_DAIFMT_DSP_B:
1447 aif1 |= WM8904_AIF_LRCLK_INV; 1447 aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
1448 case SND_SOC_DAIFMT_DSP_A: 1448 case SND_SOC_DAIFMT_DSP_A:
1449 aif1 |= 0x3; 1449 aif1 |= 0x3;
1450 break; 1450 break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b6..0f17ed3e29f4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
2439 snd_soc_update_bits(codec, WM8962_CLOCKING_4, 2439 snd_soc_update_bits(codec, WM8962_CLOCKING_4,
2440 WM8962_SYSCLK_RATE_MASK, clocking4); 2440 WM8962_SYSCLK_RATE_MASK, clocking4);
2441 2441
2442 /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK.
2443 * So we here provisionally enable it and then disable it afterward
2444 * if current bias_level hasn't reached SND_SOC_BIAS_ON.
2445 */
2446 if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
2447 snd_soc_update_bits(codec, WM8962_CLOCKING2,
2448 WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
2449
2442 dspclk = snd_soc_read(codec, WM8962_CLOCKING1); 2450 dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
2451
2452 if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
2453 snd_soc_update_bits(codec, WM8962_CLOCKING2,
2454 WM8962_SYSCLK_ENA_MASK, 0);
2455
2443 if (dspclk < 0) { 2456 if (dspclk < 0) {
2444 dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); 2457 dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
2445 return; 2458 return;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 253c88bb7a4c..4f05fb88bddf 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
1259 1259
1260 /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ 1260 /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
1261 snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); 1261 snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
1262
1263 codec->cache_sync = 1;
1262 break; 1264 break;
1263 } 1265 }
1264 1266
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4..4fbcab63e61f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
1474 return ret; 1474 return ret;
1475 1475
1476 /* Wait for the RAM to start, should be near instantaneous */ 1476 /* Wait for the RAM to start, should be near instantaneous */
1477 count = 0; 1477 for (count = 0; count < 10; ++count) {
1478 do {
1479 ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, 1478 ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
1480 &val); 1479 &val);
1481 if (ret != 0) 1480 if (ret != 0)
1482 return ret; 1481 return ret;
1483 } while (!(val & ADSP2_RAM_RDY) && ++count < 10); 1482
1483 if (val & ADSP2_RAM_RDY)
1484 break;
1485
1486 msleep(1);
1487 }
1484 1488
1485 if (!(val & ADSP2_RAM_RDY)) { 1489 if (!(val & ADSP2_RAM_RDY)) {
1486 adsp_err(dsp, "Failed to start DSP RAM\n"); 1490 adsp_err(dsp, "Failed to start DSP RAM\n");