[GIT PULL] sound fixes for v3.10-rc4
From: Takashi Iwai
Date: Thu May 30 2013 - 09:28:13 EST
Linus,
please pull sound fixes for v3.10-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-3.10
The topmost commit is 8a90bb5116889e98008fbc8178fc2a77bb51df4a
----------------------------------------------------------------
sound updates for v3.10-rc4
Again very calm updates at this time.
All small fixes for individual drivers, mostly ASoC codecs,
in addition to soc-compress fix for capture streams which is
safe to apply as there is no in-tree users yet.
----------------------------------------------------------------
Andrew Bresticker (1):
ASoC: max98090: request IRQF_ONESHOT interrupt
Charles Keepax (2):
ASoC: soc-compress: Send correct stream event for capture start
ASoC: wm5110: Correct DSP4R Mixer control name
Daniel Mack (1):
ASoC: davinci: fix sample rotation
Mark Brown (2):
ASoC: wm5110: Add missing speaker initialisation
ASoC: wm8994: Fix reporting of accessory removal on WM8958
Nicolas Schichan (4):
ASoC: cs42l52: microphone bias is controlled by IFACE_CTL2 register.
ASoC: cs42l52: fix bogus shifts in "Speaker Volume" and "PCM Mixer Volume" controls.
ASoC: cs42l52: fix master playback mute mask.
ASoC: cs42l52: fix default value for MASTERA_VOL.
Torsten Schenk (1):
ALSA: usb-6fire: Modify firmware version check
Vinod Koul (2):
ASoC: wm8994: use the correct pointer to get the control value
ASoC: wm8994: check for array index returned
---
sound/soc/codecs/cs42l52.c | 8 ++++----
sound/soc/codecs/cs42l52.h | 2 +-
sound/soc/codecs/max98090.c | 2 +-
sound/soc/codecs/wm5110.c | 4 +++-
sound/soc/codecs/wm8994.c | 12 +++++++++++-
sound/soc/davinci/davinci-mcasp.c | 7 ++++---
sound/soc/soc-compress.c | 8 ++++++--
sound/usb/6fire/firmware.c | 6 +++---
8 files changed, 33 insertions(+), 16 deletions(-)
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481..030f53c 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
{ CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */
{ CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */
{ CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */
- { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */
+ { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */
{ CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */
{ CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */
{ CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */
@@ -225,7 +225,7 @@ static const char * const mic_bias_level_text[] = {
};
static const struct soc_enum mic_bias_level_enum =
- SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+ SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -413,7 +413,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
- CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+ CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +441,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
- 6, 0x7f, 0x19, hl_tlv),
+ 0, 0x7f, 0x19, hl_tlv),
SOC_DOUBLE_R("PCM Mixer Switch",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c0..4277012 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
#define CS42L52_PB_CTL1_INV_PCMA (1 << 2)
#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1)
#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK 0x03
#define CS42L52_PB_CTL1_MUTE 3
#define CS42L52_PB_CTL1_UNMUTE 0
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d364..8d14a76 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
ret = request_threaded_irq(max98090->irq, NULL,
- max98090_interrupt, IRQF_TRIGGER_FALLING,
+ max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
"max98090_interrupt", codec);
if (ret < 0) {
dev_err(codec->dev, "request_irq failed: %d\n",
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e..ba38f06 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -976,6 +976,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
if (ret != 0)
return ret;
+ arizona_init_spk(codec);
+
snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1eb152c..dfd997a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int drc = wm8994_get_drc(kcontrol->id.name);
+ if (drc < 0)
+ return drc;
ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
return 0;
@@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
+ if (block < 0)
+ return block;
+
ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
return 0;
@@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
- struct wm8994 *control = codec->control_data;
+ struct wm8994 *control = wm8994->wm8994;
int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
int i;
int dac;
@@ -3833,6 +3838,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
dev_dbg(codec->dev, "Ignoring removed jack\n");
return IRQ_HANDLED;
}
+ } else if (!(reg & WM8958_MICD_STS)) {
+ snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+ SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+ wm8994->btn_mask);
+ goto out;
}
if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc7..81490fe 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
int word_length)
{
u32 fmt;
- u32 rotate = (word_length / 4) & 0x7;
+ u32 tx_rotate = (word_length / 4) & 0x7;
+ u32 rx_rotate = (32 - word_length) / 4;
u32 mask = (1ULL << word_length) - 1;
/*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXSSZ(fmt), TXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
- TXROT(rotate), TXROT(7));
+ TXROT(tx_rotate), TXROT(7));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
- RXROT(rotate), RXROT(7));
+ RXROT(rx_rotate), RXROT(7));
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
mask);
}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7e..06a8000 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
goto err;
}
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_START);
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+ SND_SOC_DAPM_STREAM_START);
/* cancel any delayed stream shutdown that is pending */
rtd->pop_wait = 0;
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a1d9b07..b9defcd 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = {
0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
};
-static const u8 known_fw_versions[][4] = {
- { 0x03, 0x01, 0x0b, 0x00 }
+static const u8 known_fw_versions[][2] = {
+ { 0x03, 0x01 }
};
struct ihex_record {
@@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version)
int i;
for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
- if (!memcmp(version, known_fw_versions + i, 4))
+ if (!memcmp(version, known_fw_versions + i, 2))
return 0;
snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
--
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