| lists.openwall.net | lists / announce owl-users owl-dev john-users john-dev passwdqc-users yescrypt popa3d-users / oss-security kernel-hardening musl sabotage tlsify passwords / crypt-dev xvendor / Bugtraq Full-Disclosure linux-kernel linux-netdev linux-ext4 linux-hardening linux-cve-announce PHC | |
|
Open Source and information security mailing list archives
| ||
|
Date: Mon, 27 Apr 2009 17:49:12 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound updates
Linus,
please pull another ALSA fixes for v2.6.30-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following trivial patches. The only large LOC is
found for HD-audio, but it's just an addition of a new init verb
table and pretty specific to that h/w model.
Thanks!
Takashi
===
Daniel Mack (1):
ALSA: snd-usb-caiaq: fix reported elapsed periods
Mark Brown (3):
ASoC: Fix offset of freqmode in WM8580 PLL configuration
ASoC: Fix WM8580 volume update handling for large register changes
ASoC: s3c-i2s-v2 needs to declare a license for modular builds
Subrata Modak (1):
ALSA: Fix Trivial Warnining in sound/pci/cmipci.c
Takashi Iwai (2):
ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991
ALSA: hda - Fix init verbs of AD1884A mobile model
---
sound/pci/cmipci.c | 2 +-
sound/pci/hda/patch_analog.c | 45 +++++++++++++++++++++++++++++++++++++++-
sound/soc/codecs/Makefile | 1 -
sound/soc/codecs/wm8580.c | 16 ++++++++-----
sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +-
sound/usb/caiaq/audio.c | 12 ++++++----
sound/usb/caiaq/device.c | 2 +-
7 files changed, 65 insertions(+), 16 deletions(-)
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index c7899c3..449fe02 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3014,7 +3014,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
.dev_free = snd_cmipci_dev_free,
};
unsigned int val;
- long iomidi;
+ long iomidi = 0;
int integrated_midi = 0;
char modelstr[16];
int pcm_index, pcm_spdif_index;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 9bcd8ab..84cc49c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3817,6 +3817,49 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
{ } /* end */
};
+static struct hda_verb ad1884a_mobile_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-B (mic jack) pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ /* Port-C (int mic) pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ /* 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},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ { } /* end */
+};
+
/*
* Thinkpad X300
* 0x11 - HP
@@ -3988,7 +4031,7 @@ static int patch_ad1884a(struct hda_codec *codec)
break;
case AD1884A_MOBILE:
spec->mixers[0] = ad1884a_mobile_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+ spec->init_verbs[0] = ad1884a_mobile_verbs;
spec->multiout.dig_out_nid = 0;
codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
codec->patch_ops.init = ad1884a_hp_init;
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d245..f265380 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f..9f6be3d 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int reg = kcontrol->private_value & 0xff;
- int reg2 = (kcontrol->private_value >> 24) & 0xff;
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
int ret;
u16 val;
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
return 0;
}
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+ xinvert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
- .private_value = (reg_left) | ((shift) << 8) | \
- ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .max = xmax, .invert = xinvert} }
static const struct snd_kcontrol_new wm8580_snd_controls[] = {
SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
reg = wm8580_read(codec, WM8580_PLLA4 + offset);
reg &= ~0x3f;
reg |= pll_div.prescale | pll_div.postscale << 1 |
- pll_div.freqmode << 4;
+ pll_div.freqmode << 3;
wm8580_write(codec, WM8580_PLLA4 + offset, reg);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 689ffcd..ab680aa 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
return snd_soc_register_dai(dai);
}
-
EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 3f45c0f..b13ce76 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -195,11 +195,14 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
debug("%s(%p)\n", __func__, substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
- else
+ } else {
+ dev->period_in_count[index] = BYTES_PER_SAMPLE;
dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
-
+ }
+
if (dev->streaming)
return 0;
@@ -300,8 +303,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev,
if (!sub)
continue;
- pb = frames_to_bytes(sub->runtime,
- sub->runtime->period_size);
+ pb = snd_pcm_lib_period_bytes(sub);
cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
&dev->period_out_count[stream] :
&dev->period_in_count[stream];
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 6d51770..515de1c 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@...aq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
--
To unsubscribe from this list: send the line "unsubscribe linux-kernel" in
the body of a message to majordomo@...r.kernel.org
More majordomo info at http://vger.kernel.org/majordomo-info.html
Please read the FAQ at http://www.tux.org/lkml/
Powered by blists - more mailing lists