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Diffstat (limited to 'sound/soc/codecs/cs42l42.c')
-rw-r--r--sound/soc/codecs/cs42l42.c44
1 files changed, 29 insertions, 15 deletions
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 99c022be94a6..9a463ab54bdd 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -586,6 +586,7 @@ struct cs42l42_pll_params {
* Table 4-5 from the Datasheet
*/
static const struct cs42l42_pll_params pll_ratio_table[] = {
+ { 1411200, 0, 1, 0x00, 0x80, 0x000000, 0x03, 0x10, 11289600, 128, 2},
{ 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2},
{ 2304000, 0, 1, 0x00, 0x55, 0xC00000, 0x02, 0x10, 12288000, 85, 2},
{ 2400000, 0, 1, 0x00, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2},
@@ -843,6 +844,13 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream,
if (channels == 1)
cs42l42->bclk *= 2;
+ /*
+ * Assume 24-bit samples are in 32-bit slots, to prevent SCLK being
+ * more than assumed (which would result in overclocking).
+ */
+ if (params_width(params) == 24)
+ cs42l42->bclk = (cs42l42->bclk / 3) * 4;
+
switch(substream->stream) {
case SNDRV_PCM_STREAM_CAPTURE:
if (channels == 2) {
@@ -890,10 +898,23 @@ static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
{
struct snd_soc_component *component = dai->component;
struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
+ int i;
- cs42l42->sclk = freq;
+ if (freq == 0) {
+ cs42l42->sclk = 0;
+ return 0;
+ }
- return 0;
+ for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) {
+ if (pll_ratio_table[i].sclk == freq) {
+ cs42l42->sclk = freq;
+ return 0;
+ }
+ }
+
+ dev_err(component->dev, "SCLK %u not supported\n", freq);
+
+ return -EINVAL;
}
static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
@@ -901,7 +922,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
struct snd_soc_component *component = dai->component;
struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
unsigned int regval;
- u8 fullScaleVol;
int ret;
if (mute) {
@@ -972,20 +992,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
cs42l42->stream_use |= 1 << stream;
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* Read the headphone load */
- regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT);
- if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) ==
- CS42L42_RLA_STAT_15_OHM) {
- fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK;
- } else {
- fullScaleVol = 0;
- }
-
- /* Un-mute the headphone, set the full scale volume flag */
+ /* Un-mute the headphone */
snd_soc_component_update_bits(component, CS42L42_HP_CTL,
CS42L42_HP_ANA_AMUTE_MASK |
- CS42L42_HP_ANA_BMUTE_MASK |
- CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol);
+ CS42L42_HP_ANA_BMUTE_MASK,
+ 0);
}
}
@@ -2106,4 +2117,7 @@ MODULE_DESCRIPTION("ASoC CS42L42 driver");
MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <[email protected]>");
MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <[email protected]>");
MODULE_AUTHOR("Michael White, Cirrus Logic Inc, <[email protected]>");
+MODULE_AUTHOR("Lucas Tanure <[email protected]>");
+MODULE_AUTHOR("Richard Fitzgerald <[email protected]>");
+MODULE_AUTHOR("Vitaly Rodionov <[email protected]>");
MODULE_LICENSE("GPL");