From 0cfae7c9378cf77434f6be89b5fb65d8f9a5031f Mon Sep 17 00:00:00 2001
From: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Date: Tue, 28 Jun 2011 16:59:14 +0200
Subject: [PATCH 1/6] ALSA: atmel - update author email for ABDAC, AC97C and
AT73C213
This patch updates the email address of the sound drivers supported by me to an
email account I will use on a more regular basis in the future.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/atmel/abdac.c | 2 +-
sound/atmel/ac97c.c | 2 +-
sound/spi/at73c213.c | 2 +-
3 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 6e2409181895..bfee60c4d4c0 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -599,4 +599,4 @@ module_exit(atmel_abdac_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index b310702c646e..ac35222ad0dd 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1199,4 +1199,4 @@ module_exit(atmel_ac97c_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Driver for Atmel AC97 controller");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 337a00241a1f..4dd051bdf4fd 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -1124,6 +1124,6 @@ static void __exit at73c213_exit(void)
}
module_exit(at73c213_exit);
-MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC");
MODULE_LICENSE("GPL");
From f5b2d0ef631bb0647ae8ed1752d2127b8fb6da70 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Wed, 29 Jun 2011 14:26:07 +0800
Subject: [PATCH 2/6] ALSA: HDMI - fix ELD monitor name length
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_eld.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b05f7be9dc1b..e3e853153d14 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_eld *e,
snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
goto out_fail;
} else
- strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl);
+ strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1);
for (i = 0; i < e->sad_count; i++) {
if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
From 71276410e17653cfacfa238a363475cde9e18fb3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2011 12:31:23 +0200
Subject: [PATCH 3/6] ALSA: cs5535 - Fix invalid big-endian conversions
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/cs5535audio/cs5535audio_pcm.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index f16bc8aad6ed..e083122ca55a 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -149,7 +149,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
&((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
desc->addr = cpu_to_le32(addr);
desc->size = cpu_to_le32(period_bytes);
- desc->ctlreserved = cpu_to_le32(PRD_EOP);
+ desc->ctlreserved = cpu_to_le16(PRD_EOP);
desc_addr += sizeof(struct cs5535audio_dma_desc);
addr += period_bytes;
}
@@ -157,7 +157,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods];
lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr);
lastdesc->size = 0;
- lastdesc->ctlreserved = cpu_to_le32(PRD_JMP);
+ lastdesc->ctlreserved = cpu_to_le16(PRD_JMP);
jmpprd_addr = cpu_to_le32(lastdesc->addr +
(sizeof(struct cs5535audio_dma_desc)*periods));
From 286bed0f0c447b6660e72093d7e778784fdd9ee6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2011 12:45:36 +0200
Subject: [PATCH 4/6] ALSA: hdspm - Fix compile warnings with PPC
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/rme9652/hdspm.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 3f08afc0f0d3..c8e402fc3782 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -896,11 +896,11 @@ struct hdspm {
unsigned char max_channels_in;
unsigned char max_channels_out;
- char *channel_map_in;
- char *channel_map_out;
+ signed char *channel_map_in;
+ signed char *channel_map_out;
- char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
- char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
+ signed char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
+ signed char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
char **port_names_in;
char **port_names_out;
From 4f3c7a18d9e8a287d31f828a259d713fe4859471 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jun 2011 15:08:04 +0200
Subject: [PATCH 5/6] ALSA: sb16 - Fix build errors on MIPS and others with
13bit ioctl size
One of ioctl definition in sound/sb16_csp.h contains the data size
over 8kB, and this causes build errors on architectures like MIPS,
which define _IOC_SIZEBITS=13.
For avoiding this build errors but keeping the compatibility, manually
expand with _IOC() instead of using _IOW() for the problematic ioctl.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/sb16_csp.h | 9 ++++++++-
1 file changed, 8 insertions(+), 1 deletion(-)
diff --git a/include/sound/sb16_csp.h b/include/sound/sb16_csp.h
index 736eac71d053..af1b49e982df 100644
--- a/include/sound/sb16_csp.h
+++ b/include/sound/sb16_csp.h
@@ -99,7 +99,14 @@ struct snd_sb_csp_info {
/* get CSP information */
#define SNDRV_SB_CSP_IOCTL_INFO _IOR('H', 0x10, struct snd_sb_csp_info)
/* load microcode to CSP */
-#define SNDRV_SB_CSP_IOCTL_LOAD_CODE _IOW('H', 0x11, struct snd_sb_csp_microcode)
+/* NOTE: struct snd_sb_csp_microcode overflows the max size (13 bits)
+ * defined for some architectures like MIPS, and it leads to build errors.
+ * (x86 and co have 14-bit size, thus it's valid, though.)
+ * As a workaround for skipping the size-limit check, here we don't use the
+ * normal _IOW() macro but _IOC() with the manual argument.
+ */
+#define SNDRV_SB_CSP_IOCTL_LOAD_CODE \
+ _IOC(_IOC_WRITE, 'H', 0x11, sizeof(struct snd_sb_csp_microcode))
/* unload microcode from CSP */
#define SNDRV_SB_CSP_IOCTL_UNLOAD_CODE _IO('H', 0x12)
/* start CSP */
From 9c7a083d94656ad6d6f2e03ba90194f2cc5bced5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 7 Jul 2011 09:25:54 +0200
Subject: [PATCH 6/6] ALSA: hda - Change all ADCs for dual-adc switching mode
for Realtek
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++----------
1 file changed, 23 insertions(+), 10 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d21191dcfe88..7d492713c1c1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
- getput_call_t func)
+ getput_call_t func, bool check_adc_switch)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- int err;
+ int i, err;
mutex_lock(&codec->control_mutex);
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
- 3, 0, HDA_INPUT);
- err = func(kcontrol, ucontrol);
+ if (check_adc_switch && spec->dual_adc_switch) {
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
+ err = func(kcontrol, ucontrol);
+ if (err < 0)
+ goto error;
+ }
+ } else {
+ i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
+ err = func(kcontrol, ucontrol);
+ }
+ error:
mutex_unlock(&codec->control_mutex);
return err;
}
@@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_volume_get);
+ snd_hda_mixer_amp_volume_get, false);
}
static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_volume_put);
+ snd_hda_mixer_amp_volume_put, true);
}
/* capture mixer elements */
@@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_switch_get);
+ snd_hda_mixer_amp_switch_get, false);
}
static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_switch_put);
+ snd_hda_mixer_amp_switch_put, true);
}
#define _DEFINE_CAPMIX(num) \