[GIT PULL] sound fixes for 2.6.32-rc2
From: Takashi Iwai
Date: Sat Oct 03 2009 - 12:39:55 EST
Linus,
please pull sound fixes for v2.6.32-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following fixes.
Thanks!
Takashi
===
Barry Song (1):
ASoC: fix kconfig order of Blackfin drivers
Chaithrika U S (1):
ASoC: DaVinci: Correct McASP FIFO initialization
Cliff Cai (1):
ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G
Daniel T Chen (3):
ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist
ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P
ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP
Giuliano Pochini (1):
ALSA: echoaudio - Re-enable the line-out control for the Mia card
Jean Delvare (1):
sound: Make keywest_driver static
Lukasz Marcinowski (1):
ALSA: hda - CD-audio sound for hda-intel conexant benq laptop
Manoj Iyer (1):
ALSA: hda - Added quirk to enable sound on Toshiba NB200
Miguel de Barros (1):
ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
Sven Eckelmann (1):
ALSA: ctxfi: Swapped SURROUND-SIDE mute
Takashi Iwai (7):
ALSA: hda - Resurrect input-source mixer of ALC268 model=acer
ALSA: Don't assume i2c device probing always succeeds
ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2
ALSA: hda - Fix digital/analog mic auto-switching with IDT codecs
ALSA: hda - Fix / improve ALC66x parser
ALSA: Fix invalid __exit in sound/mips/*.c
ALSA: usb - Use strlcat() correctly
Troy Kisky (2):
ASoC: DaVinci: Fix divide by zero error during 1st execution
ASoC: Davinci: Fix race with cpu_dai->dma_data
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/aoa/codecs/tas.c | 9 +
sound/mips/hal2.c | 2 +-
sound/mips/sgio2audio.c | 2 +-
sound/pci/ctxfi/ctatc.c | 4 +-
sound/pci/echoaudio/echoaudio.c | 30 +++-
sound/pci/echoaudio/mia.c | 1 +
sound/pci/hda/hda_intel.c | 1 +
sound/pci/hda/patch_analog.c | 139 +++++++++++++++
sound/pci/hda/patch_conexant.c | 12 ++-
sound/pci/hda/patch_realtek.c | 244 +++++++++++++++++---------
sound/pci/hda/patch_sigmatel.c | 20 ++-
sound/pci/intel8x0.c | 12 ++
sound/ppc/keywest.c | 14 ++-
sound/soc/blackfin/Kconfig | 98 +++++-----
sound/soc/blackfin/bf5xx-i2s.c | 8 +-
sound/soc/blackfin/bf5xx-tdm.c | 8 +-
sound/soc/davinci/davinci-i2s.c | 37 ++---
sound/soc/davinci/davinci-mcasp.c | 80 +++------
sound/soc/davinci/davinci-mcasp.h | 7 +-
sound/soc/davinci/davinci-pcm.c | 13 +-
sound/soc/davinci/davinci-pcm.h | 1 -
sound/soc/pxa/Kconfig | 2 +-
sound/usb/usbmixer.c | 23 ++-
24 files changed, 512 insertions(+), 256 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index f1708b7..75fddb4 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -209,6 +209,7 @@ AD1884A / AD1883 / AD1984A / AD1984B
laptop laptop with HP jack sensing
mobile mobile devices with HP jack sensing
thinkpad Lenovo Thinkpad X300
+ touchsmart HP Touchsmart
AD1884
======
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index f0ebc97..1dd66dd 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -897,6 +897,15 @@ static int tas_create(struct i2c_adapter *adapter,
client = i2c_new_device(adapter, &info);
if (!client)
return -ENODEV;
+ /*
+ * We know the driver is already loaded, so the device should be
+ * already bound. If not it means binding failed, and then there
+ * is no point in keeping the device instantiated.
+ */
+ if (!client->driver) {
+ i2c_unregister_device(client);
+ return -ENODEV;
+ }
/*
* Let i2c-core delete that device on driver removal.
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c52691c..9a88cdf 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -915,7 +915,7 @@ static int __devinit hal2_probe(struct platform_device *pdev)
return 0;
}
-static int __exit hal2_remove(struct platform_device *pdev)
+static int __devexit hal2_remove(struct platform_device *pdev)
{
struct snd_card *card = platform_get_drvdata(pdev);
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index e497525..8691f4c 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -973,7 +973,7 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
return 0;
}
-static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
{
struct snd_card *card = platform_get_drvdata(pdev);
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b1b3a64..7545464 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1037,7 +1037,7 @@ static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state)
static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state)
{
- return atc_daio_unmute(atc, state, LINEO4);
+ return atc_daio_unmute(atc, state, LINEO2);
}
static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
@@ -1047,7 +1047,7 @@ static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state)
{
- return atc_daio_unmute(atc, state, LINEO2);
+ return atc_daio_unmute(atc, state, LINEO4);
}
static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index da2065c..1305f7c 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -950,7 +950,7 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
Control interface
******************************************************************************/
-#ifndef ECHOCARD_HAS_VMIXER
+#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN)
/******************* PCM output volume *******************/
static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
@@ -1003,6 +1003,19 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
return changed;
}
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+/* On the Mia this one controls the line-out volume */
+static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
+ .name = "Line Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_output_gain_info,
+ .get = snd_echo_output_gain_get,
+ .put = snd_echo_output_gain_put,
+ .tlv = {.p = db_scale_output_gain},
+};
+#else
static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
.name = "PCM Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1012,9 +1025,10 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
.put = snd_echo_output_gain_put,
.tlv = {.p = db_scale_output_gain},
};
-
#endif
+#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
+
#ifdef ECHOCARD_HAS_INPUT_GAIN
@@ -2030,10 +2044,18 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
goto ctl_error;
-#else
- if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&snd_echo_line_output_gain, chip));
+ if (err < 0)
goto ctl_error;
#endif
+#else /* ECHOCARD_HAS_VMIXER */
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&snd_echo_pcm_output_gain, chip));
+ if (err < 0)
+ goto ctl_error;
+#endif /* ECHOCARD_HAS_VMIXER */
#ifdef ECHOCARD_HAS_INPUT_GAIN
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
index f3b9b45..f05c8c0 100644
--- a/sound/pci/echoaudio/mia.c
+++ b/sound/pci/echoaudio/mia.c
@@ -29,6 +29,7 @@
#define ECHOCARD_HAS_ADAT FALSE
#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
#define ECHOCARD_HAS_MIDI
+#define ECHOCARD_HAS_LINE_OUT_GAIN
/* Pipe indexes */
#define PX_ANALOG_OUT 0 /* 8 */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 20a66f8..c9ad182 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2303,6 +2303,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
* white-list for enable_msi
*/
static struct snd_pci_quirk msi_white_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1),
SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
{}
};
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 215e72a..2d603f6 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
}
/*
+ * HP Touchsmart
+ * port-A (0x11) - front hp-out
+ * port-B (0x14) - unused
+ * port-C (0x15) - unused
+ * port-D (0x12) - rear line out
+ * port-E (0x1c) - front mic-in
+ * port-F (0x16) - Internal speakers
+ * digital-mic (0x17) - Internal mic
+ */
+
+static struct hda_verb ad1984a_touchsmart_verbs[] = {
+ /* DACs; unmute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ /* Port-A (HP) mixer - route only from analog mixer */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-A pin */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Port-A (HP) pin - always unmuted */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-E (int speaker) mixer - route only from analog mixer */
+ {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
+ /* Port-E pin */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /* Port-F (int speaker) mixer - route only from analog mixer */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-F pin */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* capture sources */
+ /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ /* allow to touch GPIO1 (for mute control) */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
+ /* internal mic - dmic */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* set magic COEFs for dmic */
+ {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+ {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = ad1884a_mobile_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/* switch to external mic if plugged */
+static void ad1984a_touchsmart_automic(struct hda_codec *codec)
+{
+ if (snd_hda_codec_read(codec, 0x1c, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) {
+ snd_hda_codec_write(codec, 0x0c, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x4);
+ } else {
+ snd_hda_codec_write(codec, 0x0c, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x5);
+ }
+}
+
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case AD1884A_HP_EVENT:
+ ad1884a_hp_automute(codec);
+ break;
+ case AD1884A_MIC_EVENT:
+ ad1984a_touchsmart_automic(codec);
+ break;
+ }
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_touchsmart_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1884a_hp_automute(codec);
+ ad1984a_touchsmart_automic(codec);
+ return 0;
+}
+
+
+/*
*/
enum {
@@ -4039,6 +4160,7 @@ enum {
AD1884A_LAPTOP,
AD1884A_MOBILE,
AD1884A_THINKPAD,
+ AD1984A_TOUCHSMART,
AD1884A_MODELS
};
@@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
[AD1884A_LAPTOP] = "laptop",
[AD1884A_MOBILE] = "mobile",
[AD1884A_THINKPAD] = "thinkpad",
+ [AD1984A_TOUCHSMART] = "touchsmart",
};
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
@@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
+ SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
{}
};
@@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec)
codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
codec->patch_ops.init = ad1984a_thinkpad_init;
break;
+ case AD1984A_TOUCHSMART:
+ spec->mixers[0] = ad1984a_touchsmart_mixers;
+ spec->init_verbs[0] = ad1984a_touchsmart_verbs;
+ spec->multiout.dig_out_nid = 0;
+ codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
+ codec->patch_ops.init = ad1984a_touchsmart_init;
+ /* set the upper-limit for mixer amp to 0dB for avoiding the
+ * possible damage by overloading
+ */
+ snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+ break;
}
return 0;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9d899ed..3fbbc8c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = {
};
static struct hda_input_mux cxt5045_capture_source_benq = {
- .num_items = 3,
+ .num_items = 5,
.items = {
{ "IntMic", 0x1 },
{ "ExtMic", 0x2 },
{ "LineIn", 0x3 },
+ { "CD", 0x4 },
+ { "Mixer", 0x0 },
}
};
@@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
};
static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
+ HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
+
HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
+
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1296058..7810d3d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12660,7 +12660,7 @@ static struct alc_config_preset alc268_presets[] = {
.init_hook = alc268_toshiba_automute,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
+ .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
@@ -16852,6 +16852,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
ALC662_3ST_6ch_DIG),
@@ -17145,70 +17146,145 @@ static struct alc_config_preset alc662_presets[] = {
* BIOS auto configuration
*/
+/* convert from MIX nid to DAC */
+static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
+{
+ if (nid == 0x0f)
+ return 0x02;
+ else if (nid >= 0x0c && nid <= 0x0e)
+ return nid - 0x0c + 0x02;
+ else
+ return 0;
+}
+
+/* get MIX nid connected to the given pin targeted to DAC */
+static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
+ hda_nid_t dac)
+{
+ hda_nid_t mix[4];
+ int i, num;
+
+ num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
+ for (i = 0; i < num; i++) {
+ if (alc662_mix_to_dac(mix[i]) == dac)
+ return mix[i];
+ }
+ return 0;
+}
+
+/* look for an empty DAC slot */
+static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t srcs[5];
+ int i, j, num;
+
+ num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
+ if (num < 0)
+ return 0;
+ for (i = 0; i < num; i++) {
+ hda_nid_t nid = alc662_mix_to_dac(srcs[i]);
+ if (!nid)
+ continue;
+ for (j = 0; j < spec->multiout.num_dacs; j++)
+ if (spec->multiout.dac_nids[j] == nid)
+ break;
+ if (j >= spec->multiout.num_dacs)
+ return nid;
+ }
+ return 0;
+}
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+ hda_nid_t dac;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ for (i = 0; i < cfg->line_outs; i++) {
+ dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]);
+ if (!dac)
+ continue;
+ spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+ }
+ return 0;
+}
+
+static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
+ hda_nid_t nid, unsigned int chs)
+{
+ char name[32];
+ sprintf(name, "%s Playback Volume", pfx);
+ return add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+}
+
+static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
+ hda_nid_t nid, unsigned int chs)
+{
+ char name[32];
+ sprintf(name, "%s Playback Switch", pfx);
+ return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT));
+}
+
+#define alc662_add_stereo_vol(spec, pfx, nid) \
+ alc662_add_vol_ctl(spec, pfx, nid, 3)
+#define alc662_add_stereo_sw(spec, pfx, nid) \
+ alc662_add_sw_ctl(spec, pfx, nid, 3)
+
/* add playback controls from the parsed DAC table */
-static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
+static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- char name[32];
+ struct alc_spec *spec = codec->spec;
static const char *chname[4] = {
"Front", "Surround", NULL /*CLFE*/, "Side"
};
- hda_nid_t nid;
+ hda_nid_t nid, mix;
int i, err;
for (i = 0; i < cfg->line_outs; i++) {
- if (!spec->multiout.dac_nids[i])
+ nid = spec->multiout.dac_nids[i];
+ if (!nid)
+ continue;
+ mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
+ if (!mix)
continue;
- nid = alc880_idx_to_dac(i);
if (i == 2) {
/* Center/LFE */
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 1, 0,
- HDA_OUTPUT));
+ err = alc662_add_vol_ctl(spec, "Center", nid, 1);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
+ err = alc662_add_vol_ctl(spec, "LFE", nid, 2);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
- HDA_INPUT));
+ err = alc662_add_sw_ctl(spec, "Center", mix, 1);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
- HDA_INPUT));
+ err = alc662_add_sw_ctl(spec, "LFE", mix, 2);
if (err < 0)
return err;
} else {
const char *pfx;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
- if (!cfg->hp_pins)
+ if (cfg->hp_outs)
pfx = "Speaker";
else
pfx = "PCM";
} else
pfx = chname[i];
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
+ err = alc662_add_vol_ctl(spec, pfx, nid, 3);
if (err < 0)
return err;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
- 3, 0, HDA_INPUT));
+ err = alc662_add_sw_ctl(spec, pfx, mix, 3);
if (err < 0)
return err;
}
@@ -17217,54 +17293,38 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
}
/* add playback controls for speaker and HP outputs */
-static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
+/* return DAC nid if any new DAC is assigned */
+static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
const char *pfx)
{
- hda_nid_t nid;
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid, mix;
int err;
- char name[32];
if (!pin)
return 0;
-
- if (pin == 0x17) {
- /* ALC663 has a mono output pin on 0x17 */
+ nid = alc662_look_for_dac(codec, pin);
+ if (!nid) {
+ char name[32];
+ /* the corresponding DAC is already occupied */
+ if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
+ return 0; /* no way */
+ /* create a switch only */
sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
- return err;
+ return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
}
- if (alc880_is_fixed_pin(pin)) {
- nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
- /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
- /* specify the DAC as the extra output */
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = nid;
- else
- spec->multiout.extra_out_nid[0] = nid;
- /* control HP volume/switch on the output mixer amp */
- nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
- if (err < 0)
- return err;
- } else if (alc880_is_multi_pin(pin)) {
- /* set manual connection */
- /* we have only a switch on HP-out PIN */
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- return 0;
+ mix = alc662_dac_to_mix(codec, pin, nid);
+ if (!mix)
+ return 0;
+ err = alc662_add_vol_ctl(spec, pfx, nid, 3);
+ if (err < 0)
+ return err;
+ err = alc662_add_sw_ctl(spec, pfx, mix, 3);
+ if (err < 0)
+ return err;
+ return nid;
}
/* create playback/capture controls for input pins */
@@ -17273,30 +17333,35 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
- int dac_idx)
+ hda_nid_t dac)
{
+ int i, num;
+ hda_nid_t srcs[4];
+
alc_set_pin_output(codec, nid, pin_type);
/* need the manual connection? */
- if (alc880_is_multi_pin(nid)) {
- struct alc_spec *spec = codec->spec;
- int idx = alc880_multi_pin_idx(nid);
- snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
- AC_VERB_SET_CONNECT_SEL,
- alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+ num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
+ if (num <= 1)
+ return;
+ for (i = 0; i < num; i++) {
+ if (alc662_mix_to_dac(srcs[i]) != dac)
+ continue;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
+ return;
}
}
static void alc662_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
int i;
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
alc662_auto_set_output_and_unmute(codec, nid, pin_type,
- i);
+ spec->multiout.dac_nids[i]);
}
}
@@ -17306,12 +17371,13 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front */
- /* use dac 0 */
- alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ if (pin)
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_HP,
+ spec->multiout.hp_nid);
pin = spec->autocfg.speaker_pins[0];
if (pin)
- alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+ spec->multiout.extra_out_nid[0]);
}
#define ALC662_PIN_CD_NID ALC880_PIN_CD_NID
@@ -17349,21 +17415,25 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
- err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+ err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc662_auto_create_extra_out(spec,
+ err = alc662_auto_create_extra_out(codec,
spec->autocfg.speaker_pins[0],
"Speaker");
if (err < 0)
return err;
- err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+ if (err)
+ spec->multiout.extra_out_nid[0] = err;
+ err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
"Headphone");
if (err < 0)
return err;
+ if (err)
+ spec->multiout.hp_nid = err;
err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 826137e..a9b2682 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -182,8 +182,8 @@ struct sigmatel_jack {
struct sigmatel_mic_route {
hda_nid_t pin;
- unsigned char mux_idx;
- unsigned char dmux_idx;
+ signed char mux_idx;
+ signed char dmux_idx;
};
struct sigmatel_spec {
@@ -3469,18 +3469,26 @@ static int set_mic_route(struct hda_codec *codec,
break;
if (i <= AUTO_PIN_FRONT_MIC) {
/* analog pin */
- mic->dmux_idx = 0;
i = get_connection_index(codec, spec->mux_nids[0], pin);
if (i < 0)
return -1;
mic->mux_idx = i;
+ mic->dmux_idx = -1;
+ if (spec->dmux_nids)
+ mic->dmux_idx = get_connection_index(codec,
+ spec->dmux_nids[0],
+ spec->mux_nids[0]);
} else if (spec->dmux_nids) {
/* digital pin */
- mic->mux_idx = 0;
i = get_connection_index(codec, spec->dmux_nids[0], pin);
if (i < 0)
return -1;
mic->dmux_idx = i;
+ mic->mux_idx = -1;
+ if (spec->mux_nids)
+ mic->mux_idx = get_connection_index(codec,
+ spec->mux_nids[0],
+ spec->dmux_nids[0]);
}
return 0;
}
@@ -4557,11 +4565,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec)
mic = &spec->ext_mic;
else
mic = &spec->int_mic;
- if (mic->dmux_idx)
+ if (mic->dmux_idx >= 0)
snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
AC_VERB_SET_CONNECT_SEL,
mic->dmux_idx);
- else
+ if (mic->mux_idx >= 0)
snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
AC_VERB_SET_CONNECT_SEL,
mic->mux_idx);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 171ada5..754867e 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1954,6 +1954,18 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
.name = "Sony S1XP",
.type = AC97_TUNE_INV_EAPD
},
+ {
+ .subvendor = 0x104d,
+ .subdevice = 0x81c0,
+ .name = "Sony VAIO VGN-T350P", /*AD1981B*/
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x104d,
+ .subdevice = 0x81c5,
+ .name = "Sony VAIO VGN-B1VP", /*AD1981B*/
+ .type = AC97_TUNE_INV_EAPD
+ },
{
.subvendor = 0x1043,
.subdevice = 0x80f3,
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 835fa19..d06f780 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -59,6 +59,18 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
strlcpy(info.type, "keywest", I2C_NAME_SIZE);
info.addr = keywest_ctx->addr;
keywest_ctx->client = i2c_new_device(adapter, &info);
+ if (!keywest_ctx->client)
+ return -ENODEV;
+ /*
+ * We know the driver is already loaded, so the device should be
+ * already bound. If not it means binding failed, and then there
+ * is no point in keeping the device instantiated.
+ */
+ if (!keywest_ctx->client->driver) {
+ i2c_unregister_device(keywest_ctx->client);
+ keywest_ctx->client = NULL;
+ return -ENODEV;
+ }
/*
* Let i2c-core delete that device on driver removal.
@@ -86,7 +98,7 @@ static const struct i2c_device_id keywest_i2c_id[] = {
{ }
};
-struct i2c_driver keywest_driver = {
+static struct i2c_driver keywest_driver = {
.driver = {
.name = "PMac Keywest Audio",
},
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ac927ff..97f1a25 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,15 +7,6 @@ config SND_BF5XX_I2S
mode (supports single stereo In/Out).
You will also need to select the audio interfaces to support below.
-config SND_BF5XX_TDM
- tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
- depends on (BLACKFIN && SND_SOC)
- help
- Say Y or M if you want to add support for codecs attached to
- the Blackfin SPORT (synchronous serial ports) interface in TDM
- mode.
- You will also need to select the audio interfaces to support below.
-
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio support for BF52x ezkit"
depends on SND_BF5XX_I2S
@@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
Enter the GPIO used to control AD73311's SE pin. Acceptable
values are 0 to 7
+config SND_BF5XX_TDM
+ tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+ depends on (BLACKFIN && SND_SOC)
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Blackfin SPORT (synchronous serial ports) interface in TDM
+ mode.
+ You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_AD1836
+ tristate "SoC AD1836 Audio support for BF5xx"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1836
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SOC_AD1938
+ tristate "SoC AD1938 Audio support for Blackfin"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1938
+ help
+ Say Y if you want to add support for AD1938 codec on Blackfin.
+
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
depends on BLACKFIN
@@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
Say y if you want AC97 driver to support up to 5.1 channel audio.
this mode will consume much more memory for DMA.
+config SND_BF5XX_HAVE_COLD_RESET
+ bool "BOARD has COLD Reset GPIO"
+ depends on SND_BF5XX_AC97
+ default y if BFIN548_EZKIT
+ default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+ int "Set a GPIO for cold reset"
+ depends on SND_BF5XX_HAVE_COLD_RESET
+ range 0 159
+ default 19 if BFIN548_EZKIT
+ default 5 if BFIN537_STAMP
+ default 0
+ help
+ Set the correct GPIO for RESET the sound chip.
+
+config SND_BF5XX_SOC_AD1980
+ tristate "SoC AD1980/1 Audio support for BF5xx"
+ depends on SND_BF5XX_AC97
+ select SND_BF5XX_SOC_AC97
+ select SND_SOC_AD1980
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
config SND_BF5XX_SOC_SPORT
tristate
@@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
select SND_SOC_AC97_BUS
select SND_BF5XX_SOC_SPORT
-config SND_BF5XX_SOC_AD1836
- tristate "SoC AD1836 Audio support for BF5xx"
- depends on SND_BF5XX_TDM
- select SND_BF5XX_SOC_TDM
- select SND_SOC_AD1836
- help
- Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1980
- tristate "SoC AD1980/1 Audio support for BF5xx"
- depends on SND_BF5XX_AC97
- select SND_BF5XX_SOC_AC97
- select SND_SOC_AD1980
- help
- Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1938
- tristate "SoC AD1938 Audio support for Blackfin"
- depends on SND_BF5XX_TDM
- select SND_BF5XX_SOC_TDM
- select SND_SOC_AD1938
- help
- Say Y if you want to add support for AD1938 codec on Blackfin.
-
config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
@@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
default 0
help
Set the correct SPORT for sound chip.
-
-config SND_BF5XX_HAVE_COLD_RESET
- bool "BOARD has COLD Reset GPIO"
- depends on SND_BF5XX_AC97
- default y if BFIN548_EZKIT
- default n if !BFIN548_EZKIT
-
-config SND_BF5XX_RESET_GPIO_NUM
- int "Set a GPIO for cold reset"
- depends on SND_BF5XX_HAVE_COLD_RESET
- range 0 159
- default 19 if BFIN548_EZKIT
- default 5 if BFIN537_STAMP
- default 0
- help
- Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 1e9d161..084b688 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
* TFS. When Port G is selected and EMAC then there is a conflict between
* the PHY interrupt line and TFS. Current settings prevent the conflict
* by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
*/
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
#define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
#endif
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 3096bad..ff546e9 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
* TFS. When Port G is selected and EMAC then there is a conflict between
* the PHY interrupt line and TFS. Current settings prevent the conflict
* by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
*/
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
#define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
#endif
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 12a6c54..4ae7070 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,22 +97,19 @@ enum {
DAVINCI_MCBSP_WORD_32,
};
-static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
- .name = "I2S PCM Stereo out",
-};
-
-static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
- .name = "I2S PCM Stereo in",
-};
-
struct davinci_mcbsp_dev {
+ /*
+ * dma_params must be first because rtd->dai->cpu_dai->private_data
+ * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+ * davinci_pcm_open.
+ */
+ struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
#define MOD_DSP_A 0
#define MOD_DSP_B 1
int mode;
u32 pcr;
struct clk *clk;
- struct davinci_pcm_dma_params *dma_params[2];
};
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
toggle_clock(dev, playback);
}
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
-}
-
#define DEFAULT_BITPERSAMPLE 16
static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct davinci_pcm_dma_params *dma_params = dai->dma_data;
struct davinci_mcbsp_dev *dev = dai->private_data;
+ struct davinci_pcm_dma_params *dma_params =
+ &dev->dma_params[substream->stream];
struct snd_interval *i = NULL;
int mcbsp_word_length;
unsigned int rcr, xcr, srgr;
@@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
- .startup = davinci_i2s_startup,
.shutdown = davinci_i2s_shutdown,
.prepare = davinci_i2s_prepare,
.trigger = davinci_i2s_trigger,
@@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev->base = (void __iomem *)IO_ADDRESS(mem->start);
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
/* first TX, then RX */
@@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = -ENXIO;
goto err_free_mem;
}
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
@@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = -ENXIO;
goto err_free_mem;
}
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
davinci_i2s_dai.private_data = dev;
ret = snd_soc_register_dai(&davinci_i2s_dai);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7a06c0a..5d1f98a 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
printk(KERN_ERR "GBLCTL write error\n");
}
-static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct davinci_audio_dev *dev = cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
-}
-
static void mcasp_start_rx(struct davinci_audio_dev *dev)
{
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
@@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt) /* enable FIFO */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_start_tx(dev);
- else
+ } else {
+ if (dev->rxnumevt) /* enable FIFO */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_start_rx(dev);
-
- /* enable FIFO */
- if (dev->txnumevt)
- mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
- if (dev->rxnumevt)
- mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
}
static void mcasp_stop_rx(struct davinci_audio_dev *dev)
@@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt) /* disable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_stop_tx(dev);
- else
+ } else {
+ if (dev->rxnumevt) /* disable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_stop_rx(dev);
-
- /* disable FIFO */
- if (dev->txnumevt)
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
- if (dev->rxnumevt)
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
}
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -720,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
{
struct davinci_audio_dev *dev = cpu_dai->private_data;
struct davinci_pcm_dma_params *dma_params =
- dev->dma_params[substream->stream];
+ &dev->dma_params[substream->stream];
int word_length;
u8 numevt;
@@ -798,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
}
static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
- .startup = davinci_mcasp_startup,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
@@ -849,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
struct resource *mem, *ioarea, *res;
struct snd_platform_data *pdata;
struct davinci_audio_dev *dev;
- int count = 0;
int ret = 0;
dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
if (!dev)
return -ENOMEM;
- dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
- GFP_KERNEL);
- if (!dma_data) {
- ret = -ENOMEM;
- goto err_release_dev;
- }
-
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "no mem resource?\n");
@@ -897,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dev->txnumevt = pdata->txnumevt;
dev->rxnumevt = pdata->rxnumevt;
- dma_data[count].name = "I2S PCM Stereo out";
- dma_data[count].eventq_no = pdata->eventq_no;
- dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+ dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_data->eventq_no = pdata->eventq_no;
+ dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
io_v2p(dev->base));
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
/* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -910,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err_release_region;
}
- dma_data[count].channel = res->start;
- count++;
- dma_data[count].name = "I2S PCM Stereo in";
- dma_data[count].eventq_no = pdata->eventq_no;
- dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+ dma_data->channel = res->start;
+
+ dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data->eventq_no = pdata->eventq_no;
+ dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
io_v2p(dev->base));
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
@@ -924,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err_release_region;
}
- dma_data[count].channel = res->start;
+ dma_data->channel = res->start;
davinci_mcasp_dai[pdata->op_mode].private_data = dev;
davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
@@ -936,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
err_release_region:
release_mem_region(mem->start, (mem->end - mem->start) + 1);
err_release_data:
- kfree(dma_data);
-err_release_dev:
kfree(dev);
return ret;
@@ -946,7 +925,6 @@ err_release_dev:
static int davinci_mcasp_remove(struct platform_device *pdev)
{
struct snd_platform_data *pdata = pdev->dev.platform_data;
- struct davinci_pcm_dma_params *dma_data;
struct davinci_audio_dev *dev;
struct resource *mem;
@@ -959,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
release_mem_region(mem->start, (mem->end - mem->start) + 1);
- dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
- kfree(dma_data);
kfree(dev);
return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 554354c..9d179cc 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,10 +39,15 @@ enum {
};
struct davinci_audio_dev {
+ /*
+ * dma_params must be first because rtd->dai->cpu_dai->private_data
+ * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+ * davinci_pcm_open.
+ */
+ struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
int sample_rate;
struct clk *clk;
- struct davinci_pcm_dma_params *dma_params[2];
unsigned int codec_fmt;
/* McASP specific data */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 2f7da49..c73a915 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
struct edmacc_param p_ram;
int ret;
- if (!dma_data)
- return -ENODEV;
-
- prtd->params = dma_data;
-
/* Request master DMA channel */
ret = edma_alloc_channel(prtd->params->channel,
davinci_pcm_dma_irq, substream,
@@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct davinci_runtime_data *prtd;
int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
+ struct davinci_pcm_dma_params *params = &pa[substream->stream];
+ if (!params)
+ return -ENODEV;
snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
/* ensure that buffer size is a multiple of period size */
@@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
return -ENOMEM;
spin_lock_init(&prtd->lock);
+ prtd->params = params;
runtime->private_data = prtd;
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 63d9625..8746606 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -17,7 +17,6 @@
struct davinci_pcm_dma_params {
- char *name; /* stream identifier */
int channel; /* sync dma channel ID */
unsigned short acnt;
dma_addr_t dma_addr; /* device physical address for DMA */
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4e..dcb3181 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
config SND_PXA2XX_SOC_IMOTE2
tristate "SoC Audio support for IMote 2"
- depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+ depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8940
help
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index ab5a3ac..9efcfd0 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -898,6 +898,11 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = {
* build a feature control
*/
+static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
+{
+ return strlcat(kctl->id.name, str, sizeof(kctl->id.name));
+}
+
static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
unsigned int ctl_mask, int control,
struct usb_audio_term *iterm, int unitid)
@@ -978,13 +983,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
*/
if (! mapped_name && ! (state->oterm.type >> 16)) {
if ((state->oterm.type & 0xff00) == 0x0100) {
- len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name));
+ len = append_ctl_name(kctl, " Capture");
} else {
- len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name));
+ len = append_ctl_name(kctl, " Playback");
}
}
- strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume",
- sizeof(kctl->id.name));
+ append_ctl_name(kctl, control == USB_FEATURE_MUTE ?
+ " Switch" : " Volume");
if (control == USB_FEATURE_VOLUME) {
kctl->tlv.c = mixer_vol_tlv;
kctl->vd[0].access |=
@@ -1143,7 +1148,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc,
len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0);
if (! len)
len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
- strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name));
+ append_ctl_name(kctl, " Volume");
snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1400,8 +1405,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned
if (! len)
strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
}
- strlcat(kctl->id.name, " ", sizeof(kctl->id.name));
- strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name));
+ append_ctl_name(kctl, " ");
+ append_ctl_name(kctl, valinfo->suffix);
snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1610,9 +1615,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi
strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
if ((state->oterm.type & 0xff00) == 0x0100)
- strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name));
+ append_ctl_name(kctl, " Capture Source");
else
- strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name));
+ append_ctl_name(kctl, " Playback Source");
}
snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
--
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