powerel7
/
base
2
0
Fork 0
Code Issues Pull Requests Releases Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
1049 Commits
1 Branch
0 Tags
390 MiB
 
 
 
 
 
 
Tree: 6441ca8477
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '6441ca8477'
${ noResults }
base/SOURCES/alsa-lib-1.1.6-post.patch

2778 lines
91 KiB
Raw Blame History

From 93e03bdc2a3dcd5d12516f5de78e14d88a32ff2c Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 4 Apr 2018 09:58:12 +0200
Subject: [PATCH 01/26] alsa.conf: change the location for add-on configs to
/etc/alsa/conf.d
The add-on configuration files should be placed to a volatile place.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
src/conf/alsa.conf | 8 +-------
1 file changed, 1 insertion(+), 7 deletions(-)
diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf
index f22918fb..bb00ff40 100644
--- a/src/conf/alsa.conf
+++ b/src/conf/alsa.conf
@@ -8,13 +8,7 @@
{
func load
files [
- {
- @func concat
- strings [
- { @func datadir }
- "/alsa.conf.d/"
- ]
- }
+ "/etc/alsa/conf.d"
"/etc/asound.conf"
"~/.asoundrc"
]
--
2.13.6
From 78505dccd23546cc77e5221cb21c01325bc0138d Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 4 Apr 2018 10:02:49 +0200
Subject: [PATCH 02/26] conf: remove alsa.conf.d from the datadir
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
src/conf/Makefile.am | 2 +-
src/conf/alsa.conf.d/Makefile.am | 8 --------
src/conf/alsa.conf.d/README | 2 --
3 files changed, 1 insertion(+), 11 deletions(-)
delete mode 100644 src/conf/alsa.conf.d/Makefile.am
delete mode 100644 src/conf/alsa.conf.d/README
diff --git a/src/conf/Makefile.am b/src/conf/Makefile.am
index ef2ea9c1..2b46f95c 100644
--- a/src/conf/Makefile.am
+++ b/src/conf/Makefile.am
@@ -1,4 +1,4 @@
-SUBDIRS=cards pcm alsa.conf.d
+SUBDIRS=cards pcm
if BUILD_UCM
SUBDIRS += ucm
diff --git a/src/conf/alsa.conf.d/Makefile.am b/src/conf/alsa.conf.d/Makefile.am
deleted file mode 100644
index c91661e9..00000000
--- a/src/conf/alsa.conf.d/Makefile.am
+++ /dev/null
@@ -1,8 +0,0 @@
-alsaconfigdir = @ALSA_CONFIG_DIR@
-alsadir = $(alsaconfigdir)/alsa.conf.d
-cfg_files = README
-
-alsa_DATA = $(cfg_files)
-
-EXTRA_DIST = \
- $(cfg_files)
diff --git a/src/conf/alsa.conf.d/README b/src/conf/alsa.conf.d/README
deleted file mode 100644
index 99978848..00000000
--- a/src/conf/alsa.conf.d/README
+++ /dev/null
@@ -1,2 +0,0 @@
-You can place files named *.conf in this folder and they will be processed
-when initialising alsa-lib.
--
2.13.6
From 17bc74d3a25f0d4b1ca25d2d92fcad9c2a9d7f79 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Tue, 10 Apr 2018 08:57:07 +0200
Subject: [PATCH 03/26] configure: remove src/conf/alsa.conf.d/Makefile
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
configure.ac | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/configure.ac b/configure.ac
index cce195ae..5bc1994f 100644
--- a/configure.ac
+++ b/configure.ac
@@ -713,7 +713,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \
src/rawmidi/Makefile src/timer/Makefile \
src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \
src/alisp/Makefile src/topology/Makefile \
- src/conf/Makefile src/conf/alsa.conf.d/Makefile \
+ src/conf/Makefile \
src/conf/cards/Makefile \
src/conf/pcm/Makefile \
src/conf/ucm/Makefile \
--
2.13.6
From 3778a30bb0095c7d3275735718f33058e3c198d5 Mon Sep 17 00:00:00 2001
From: Kirill Marinushkin <k.marinushkin@gmail.com>
Date: Mon, 16 Apr 2018 20:26:38 +0200
Subject: [PATCH 04/26] ASoC: topology: Fix bclk and fsync inversion in
set_link_hw_format()
The values of bclk and fsync are inverted WRT the codec. But the existing
solution already works for Broadwell, see the alsa-lib config:
`alsa-lib/src/conf/topology/broadwell/broadwell.conf`
This commit provides the backwards-compatible solution to fix this misuse.
This commit goes in pair with the corresponding patch for linux.
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: linux-kernel@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/asoc.h | 16 ++++++++++++++--
include/topology.h | 4 ++--
src/conf/topology/broadwell/broadwell.conf | 4 ++--
src/topology/pcm.c | 30 ++++++++++++++++++++++++++----
4 files changed, 44 insertions(+), 10 deletions(-)
diff --git a/include/sound/asoc.h b/include/sound/asoc.h
index 0f5d9f9a..89b00703 100644
--- a/include/sound/asoc.h
+++ b/include/sound/asoc.h
@@ -156,6 +156,18 @@
#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2)
#define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3)
+/* DAI topology BCLK parameter
+ * For the backwards capability, by default codec is bclk master
+ */
+#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */
+#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */
+
+/* DAI topology FSYNC parameter
+ * For the backwards capability, by default codec is fsync master
+ */
+#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */
+#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */
+
/*
* Block Header.
* This header precedes all object and object arrays below.
@@ -311,8 +323,8 @@ struct snd_soc_tplg_hw_config {
__u8 clock_gated; /* 1 if clock can be gated to save power */
__u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */
__u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */
- __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */
- __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */
+ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */
+ __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */
__u8 mclk_direction; /* 0 for input, 1 for output */
__le16 reserved; /* for 32bit alignment */
__le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */
diff --git a/include/topology.h b/include/topology.h
index 8779da4d..5d7b46df 100644
--- a/include/topology.h
+++ b/include/topology.h
@@ -1000,8 +1000,8 @@ struct snd_tplg_hw_config_template {
unsigned char clock_gated; /* 1 if clock can be gated to save power */
unsigned char invert_bclk; /* 1 for inverted BCLK, 0 for normal */
unsigned char invert_fsync; /* 1 for inverted frame clock, 0 for normal */
- unsigned char bclk_master; /* 1 for master of BCLK, 0 for slave */
- unsigned char fsync_master; /* 1 for master of FSYNC, 0 for slave */
+ unsigned char bclk_master; /* SND_SOC_TPLG_BCLK_ value */
+ unsigned char fsync_master; /* SND_SOC_TPLG_FSYNC_ value */
unsigned char mclk_direction; /* 0 for input, 1 for output */
unsigned short reserved; /* for 32bit alignment */
unsigned int mclk_rate; /* MCLK or SYSCLK freqency in Hz */
diff --git a/src/conf/topology/broadwell/broadwell.conf b/src/conf/topology/broadwell/broadwell.conf
index b8405d93..09fc4daa 100644
--- a/src/conf/topology/broadwell/broadwell.conf
+++ b/src/conf/topology/broadwell/broadwell.conf
@@ -393,8 +393,8 @@ SectionGraph."dsp" {
SectionHWConfig."CodecHWConfig" {
id "1"
format "I2S" # physical audio format.
- bclk "master" # Platform is master of bit clock
- fsync "master" # platform is master of fsync
+ bclk "codec_slave" # platform is master of bit clock (codec is slave)
+ fsync "codec_slave" # platform is master of fsync (codec is slave)
}
SectionLink."Codec" {
diff --git a/src/topology/pcm.c b/src/topology/pcm.c
index bb47b9af..d0395182 100644
--- a/src/topology/pcm.c
+++ b/src/topology/pcm.c
@@ -1141,8 +1141,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
- if (!strcmp(val, "master"))
- hw_cfg->bclk_master = true;
+ if (!strcmp(val, "master")) {
+ /* For backwards capability,
+ * "master" == "codec is slave"
+ */
+ SNDERR("warning: deprecated bclk value '%s'\n",
+ val);
+
+ hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CS;
+ } else if (!strcmp(val, "codec_slave")) {
+ hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CS;
+ } else if (!strcmp(val, "codec_master")) {
+ hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CM;
+ }
continue;
}
@@ -1167,8 +1178,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
- if (!strcmp(val, "master"))
- hw_cfg->fsync_master = true;
+ if (!strcmp(val, "master")) {
+ /* For backwards capability,
+ * "master" == "codec is slave"
+ */
+ SNDERR("warning: deprecated fsync value '%s'\n",
+ val);
+
+ hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CS;
+ } else if (!strcmp(val, "codec_slave")) {
+ hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CS;
+ } else if (!strcmp(val, "codec_master")) {
+ hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CM;
+ }
continue;
}
--
2.13.6
From bdb709ab2a091743980c9154950c01f0c540476b Mon Sep 17 00:00:00 2001
From: Kirill Marinushkin <k.marinushkin@gmail.com>
Date: Mon, 16 Apr 2018 20:26:39 +0200
Subject: [PATCH 05/26] ASoC: topology: Add missing clock gating parameter when
parsing hw_configs
Clock gating parameter is a part of `dai_fmt`. It is supported by
`alsa-lib` when creating a topology binary file, but ignored by kernel
when loading this topology file.
After applying this commit, the clock gating parameter is not ignored any
more. This solution is backwards compatible. The existing behaviour is
not broken, because by default the parameter value is 0 and is ignored.
snd_soc_tplg_hw_config.clock_gated = 0 => no effect
snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED
snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT
For example, the following config, based on
alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported:
~~~~
SectionHWConfig."CodecHWConfig" {
id "1"
format "I2S" # physical audio format.
pm_gate_clocks "true" # clock can be gated
}
SectionLink."Codec" {
# used for binding to the physical link
id "0"
hw_configs [
"CodecHWConfig"
]
default_hw_conf_id "1"
}
~~~~
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Pan Xiuli <xiuli.pan@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: linux-kernel@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/asoc.h | 7 ++++++-
include/topology.h | 2 +-
src/topology/pcm.c | 6 +++++-
3 files changed, 12 insertions(+), 3 deletions(-)
diff --git a/include/sound/asoc.h b/include/sound/asoc.h
index 89b00703..297e837c 100644
--- a/include/sound/asoc.h
+++ b/include/sound/asoc.h
@@ -135,6 +135,11 @@
#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1)
#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2)
+/* DAI clock gating */
+#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0
+#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1
+#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2
+
/* DAI physical PCM data formats.
* Add new formats to the end of the list.
*/
@@ -320,7 +325,7 @@ struct snd_soc_tplg_hw_config {
__le32 size; /* in bytes of this structure */
__le32 id; /* unique ID - - used to match */
__le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */
- __u8 clock_gated; /* 1 if clock can be gated to save power */
+ __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */
__u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */
__u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */
__u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */
diff --git a/include/topology.h b/include/topology.h
index 5d7b46df..3793115c 100644
--- a/include/topology.h
+++ b/include/topology.h
@@ -997,7 +997,7 @@ struct snd_tplg_pcm_template {
struct snd_tplg_hw_config_template {
int id; /* unique ID - - used to match */
unsigned int fmt; /* SND_SOC_DAI_FORMAT_ format value */
- unsigned char clock_gated; /* 1 if clock can be gated to save power */
+ unsigned char clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */
unsigned char invert_bclk; /* 1 for inverted BCLK, 0 for normal */
unsigned char invert_fsync; /* 1 for inverted frame clock, 0 for normal */
unsigned char bclk_master; /* SND_SOC_TPLG_BCLK_ value */
diff --git a/src/topology/pcm.c b/src/topology/pcm.c
index d0395182..b53f6b03 100644
--- a/src/topology/pcm.c
+++ b/src/topology/pcm.c
@@ -1233,7 +1233,11 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
return -EINVAL;
if (!strcmp(val, "true"))
- hw_cfg->clock_gated = true;
+ hw_cfg->clock_gated =
+ SND_SOC_TPLG_DAI_CLK_GATE_GATED;
+ else
+ hw_cfg->clock_gated =
+ SND_SOC_TPLG_DAI_CLK_GATE_CONT;
continue;
}
--
2.13.6
From 7cf73b56e4505ad194f5789293494a6ebaa1feff Mon Sep 17 00:00:00 2001
From: Kirill Marinushkin <k.marinushkin@gmail.com>
Date: Mon, 16 Apr 2018 20:26:40 +0200
Subject: [PATCH 06/26] ASoC: topology: Add definitions for mclk_direction
values
Current comment makes not clear the direction of mclk. Previously, similar
description caused a misunderstanding for bclk_master and fsync_master.
This commit solves the potential confusion the same way it is solved for
bclk_master and fsync_master.
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Pan Xiuli <xiuli.pan@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/asoc.h | 6 +++++-
include/topology.h | 2 +-
src/topology/pcm.c | 15 +++++++++++++--
3 files changed, 19 insertions(+), 4 deletions(-)
diff --git a/include/sound/asoc.h b/include/sound/asoc.h
index 297e837c..bb8aec78 100644
--- a/include/sound/asoc.h
+++ b/include/sound/asoc.h
@@ -140,6 +140,10 @@
#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1
#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2
+/* DAI mclk_direction */
+#define SND_SOC_TPLG_MCLK_CO 0 /* for codec, mclk is output */
+#define SND_SOC_TPLG_MCLK_CI 1 /* for codec, mclk is input */
+
/* DAI physical PCM data formats.
* Add new formats to the end of the list.
*/
@@ -330,7 +334,7 @@ struct snd_soc_tplg_hw_config {
__u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */
__u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */
__u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */
- __u8 mclk_direction; /* 0 for input, 1 for output */
+ __u8 mclk_direction; /* SND_SOC_TPLG_MCLK_ value */
__le16 reserved; /* for 32bit alignment */
__le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */
__le32 bclk_rate; /* BCLK freqency in Hz */
diff --git a/include/topology.h b/include/topology.h
index 3793115c..27da7308 100644
--- a/include/topology.h
+++ b/include/topology.h
@@ -1002,7 +1002,7 @@ struct snd_tplg_hw_config_template {
unsigned char invert_fsync; /* 1 for inverted frame clock, 0 for normal */
unsigned char bclk_master; /* SND_SOC_TPLG_BCLK_ value */
unsigned char fsync_master; /* SND_SOC_TPLG_FSYNC_ value */
- unsigned char mclk_direction; /* 0 for input, 1 for output */
+ unsigned char mclk_direction; /* SND_SOC_TPLG_MCLK_ value */
unsigned short reserved; /* for 32bit alignment */
unsigned int mclk_rate; /* MCLK or SYSCLK freqency in Hz */
unsigned int bclk_rate; /* BCLK freqency in Hz */
diff --git a/src/topology/pcm.c b/src/topology/pcm.c
index b53f6b03..2ce1651b 100644
--- a/src/topology/pcm.c
+++ b/src/topology/pcm.c
@@ -1223,8 +1223,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
- if (!strcmp(val, "master"))
- hw_cfg->mclk_direction = true;
+ if (!strcmp(val, "master")) {
+ /* For backwards capability,
+ * "master" == "for codec, mclk is input"
+ */
+ SNDERR("warning: deprecated mclk value '%s'\n",
+ val);
+
+ hw_cfg->mclk_direction = SND_SOC_TPLG_MCLK_CI;
+ } else if (!strcmp(val, "codec_mclk_in")) {
+ hw_cfg->mclk_direction = SND_SOC_TPLG_MCLK_CI;
+ } else if (!strcmp(val, "codec_mclk_out")) {
+ hw_cfg->mclk_direction = SND_SOC_TPLG_MCLK_CO;
+ }
continue;
}
--
2.13.6
From 4493f6a560a315970f5b068126120526a04ae6a2 Mon Sep 17 00:00:00 2001
From: Kirill Marinushkin <k.marinushkin@gmail.com>
Date: Mon, 16 Apr 2018 20:26:41 +0200
Subject: [PATCH 07/26] ASoC: topology: Add alias conf parameter names for
hw_configs
Currently, some parameter names in conf differ from field names in struct.
These look like typos.
This commit suggests to add aliases for such parameters, so that the names
in conf are similar to names in struct. This solution is backwards
compatible.
If the difference between conf names and struct names is done on purpose -
this commit can be dropped.
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Pan Xiuli <xiuli.pan@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/topology/pcm.c | 30 ++++++++++++++++++++----------
1 file changed, 20 insertions(+), 10 deletions(-)
diff --git a/src/topology/pcm.c b/src/topology/pcm.c
index 2ce1651b..8ebfafd8 100644
--- a/src/topology/pcm.c
+++ b/src/topology/pcm.c
@@ -1126,7 +1126,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "format") == 0) {
+ if (strcmp(id, "format") == 0 ||
+ strcmp(id, "fmt") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1137,7 +1138,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "bclk") == 0) {
+ if (strcmp(id, "bclk") == 0 ||
+ strcmp(id, "bclk_master") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1157,7 +1159,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "bclk_freq") == 0) {
+ if (strcmp(id, "bclk_freq") == 0 ||
+ strcmp(id, "bclk_rate") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1165,7 +1168,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "bclk_invert") == 0) {
+ if (strcmp(id, "bclk_invert") == 0 ||
+ strcmp(id, "invert_bclk") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1174,7 +1178,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "fsync") == 0) {
+ if (strcmp(id, "fsync") == 0 ||
+ strcmp(id, "fsync_master") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1194,7 +1199,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "fsync_invert") == 0) {
+ if (strcmp(id, "fsync_invert") == 0 ||
+ strcmp(id, "invert_fsync") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1203,7 +1209,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "fsync_freq") == 0) {
+ if (strcmp(id, "fsync_freq") == 0 ||
+ strcmp(id, "fsync_rate") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1211,7 +1218,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "mclk_freq") == 0) {
+ if (strcmp(id, "mclk_freq") == 0 ||
+ strcmp(id, "mclk_rate") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1219,7 +1227,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "mclk") == 0) {
+ if (strcmp(id, "mclk") == 0 ||
+ strcmp(id, "mclk_direction") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
@@ -1239,7 +1248,8 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg,
continue;
}
- if (strcmp(id, "pm_gate_clocks") == 0) {
+ if (strcmp(id, "pm_gate_clocks") == 0 ||
+ strcmp(id, "clock_gated") == 0) {
if (snd_config_get_string(n, &val) < 0)
return -EINVAL;
--
2.13.6
From 23a20cda111232b5d21dde12d10e19e4ecb71cea Mon Sep 17 00:00:00 2001
From: Ricard Wanderlof <ricard.wanderlof@axis.com>
Date: Wed, 18 Apr 2018 17:03:09 +0200
Subject: [PATCH 08/26] pcm: softvol: Allow up to 90 dB of gain
The gain algorithm used in softvol can handle gain factors of up to
32767 which is slightly more than 90 dB, so allow a max_dB of 90 dB.
This doesn't affect existing asound.conf files, but does allow a
max_dB of up to 90 dB when needed.
Tested using Audacity that there is no undue distorsion or other
artefacts when 90 dB of gain is applied to a suitable signal (i.e.
a signal quiet enough not be clipped whan applying 90 dB of gain).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/pcm/pcm_softvol.c | 6 +++++-
1 file changed, 5 insertions(+), 1 deletion(-)
diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c
index 8bb4a397..0eaeacef 100644
--- a/src/pcm/pcm_softvol.c
+++ b/src/pcm/pcm_softvol.c
@@ -59,7 +59,11 @@ typedef struct {
#define PRESET_RESOLUTION 256
#define PRESET_MIN_DB -51.0
#define ZERO_DB 0.0
-#define MAX_DB_UPPER_LIMIT 50
+/*
+ * The gain algorithm as it stands supports gain factors up to 32767, which
+ * is a fraction more than 90 dB, so set 90 dB as the maximum possible gain.
+ */
+#define MAX_DB_UPPER_LIMIT 90
static const unsigned int preset_dB_value[PRESET_RESOLUTION] = {
0x00b8, 0x00bd, 0x00c1, 0x00c5, 0x00ca, 0x00cf, 0x00d4, 0x00d9,
--
2.13.6
From 181f8e251bc05832f9c9401544e680ea0572a2e3 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 2 May 2018 14:08:05 +0800
Subject: [PATCH 09/26] ucm: adding the folder of card_long_name when finding
verb conf file
The board configuration file and verb conf file are allowed to be
in the folder named of card_long_name, so when finding the verb conf
file, we need to check if it is in the folder of card_long_name or
card_name.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
src/ucm/parser.c | 10 ++++++++--
1 file changed, 8 insertions(+), 2 deletions(-)
diff --git a/src/ucm/parser.c b/src/ucm/parser.c
index 2d76152f..219edb96 100644
--- a/src/ucm/parser.c
+++ b/src/ucm/parser.c
@@ -1056,6 +1056,7 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr,
char filename[MAX_FILE];
char *env = getenv(ALSA_CONFIG_UCM_VAR);
int err;
+ char *folder_name;
/* allocate verb */
verb = calloc(1, sizeof(struct use_case_verb));
@@ -1082,12 +1083,17 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr,
}
/* open Verb file for reading */
+ if (!strncmp(uc_mgr->conf_file_name, uc_mgr->card_long_name, MAX_CARD_LONG_NAME))
+ folder_name = uc_mgr->card_long_name;
+ else
+ folder_name = uc_mgr->card_name;
+
if (env)
snprintf(filename, sizeof(filename), "%s/%s/%s",
- env, uc_mgr->card_name, file);
+ env, folder_name, file);
else
snprintf(filename, sizeof(filename), "%s/ucm/%s/%s",
- snd_config_topdir(), uc_mgr->card_name, file);
+ snd_config_topdir(), folder_name, file);
filename[sizeof(filename)-1] = '\0';
err = uc_mgr_config_load(filename, &cfg);
--
2.13.6
From 81db276f8c2235adc83e9698b0174265f6482655 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 2 May 2018 14:08:06 +0800
Subject: [PATCH 10/26] conf/ucm: increase the input volume for LineIn
Otherwise, the boost value is 0, and the sound captured from that
LineIn jack is too weak for users.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
src/conf/ucm/HDAudio-Lenovo-DualCodecs/HiFi.conf | 1 +
1 file changed, 1 insertion(+)
diff --git a/src/conf/ucm/HDAudio-Lenovo-DualCodecs/HiFi.conf b/src/conf/ucm/HDAudio-Lenovo-DualCodecs/HiFi.conf
index 50967896..ece780da 100644
--- a/src/conf/ucm/HDAudio-Lenovo-DualCodecs/HiFi.conf
+++ b/src/conf/ucm/HDAudio-Lenovo-DualCodecs/HiFi.conf
@@ -99,6 +99,7 @@ SectionDevice."LineIn" {
EnableSequence [
cdev "hw:PCH"
cset "name='Input Source' Line"
+ cset "name='Line Boost Volume' 3"
]
}
--
2.13.6
From f7c38c29d1be8bab9dd4f406aea3b0e9151c1c06 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Sat, 28 Apr 2018 21:51:56 +0200
Subject: [PATCH 11/26] conf/ucm: chtrt5645: Cleanup and playback fixes
Apply cleanup and playback fixes changes from:
https://github.com/plbossart/UCM.git
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[hdegoede@redhat.com: Modify commit msg and paths for merging into alsa-lib]
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/conf/ucm/chtrt5645/HiFi.conf | 157 ++++++++++++++++++++++++++++++++-------
1 file changed, 129 insertions(+), 28 deletions(-)
diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf
index 0c2c83cc..f63392d4 100644
--- a/src/conf/ucm/chtrt5645/HiFi.conf
+++ b/src/conf/ucm/chtrt5645/HiFi.conf
@@ -11,51 +11,146 @@ SectionVerb {
EnableSequence [
cdev "hw:chtrt5645"
- # Enable audio output path
- cset "name='codec_out1 mix 0 pcm0_in Switch' on"
- cset "name='media0_out mix 0 media1_in Switch' on"
+ # media mixer settings
+ # compress
+ cset "name='media0_in Gain 0 Switch' on"
+ cset "name='media0_in Gain 0 Volume' 0"
- cset "name='media1_in Gain 0 Ramp Delay' 50"
+ # normal
cset "name='media1_in Gain 0 Switch' on"
- cset "name='media1_in Gain 0 Volume' 80% 80%"
+ cset "name='media1_in Gain 0 Volume' 0"
+ # swm loopback
+ cset "name='media2_in Gain 0 Switch' off"
+ cset "name='media2_in Gain 0 Volume' 0%"
+ # deep buffer
+ cset "name='media3_in Gain 0 Switch' on"
+ cset "name='media3_in Gain 0 Volume' 0"
+
+ cset "name='media0_out mix 0 media0_in Switch' on"
+ cset "name='media0_out mix 0 media1_in Switch' on"
+ cset "name='media0_out mix 0 media2_in Switch' off"
+ cset "name='media0_out mix 0 media3_in Switch' on"
+
+ cset "name='media1_out mix 0 media0_in Switch' off"
+ cset "name='media1_out mix 0 media1_in Switch' off"
+ cset "name='media1_out mix 0 media2_in Switch' off"
+ cset "name='media1_out mix 0 media3_in Switch' off"
- cset "name='pcm0_in Gain 0 Ramp Delay' 50"
cset "name='pcm0_in Gain 0 Switch' on"
- cset "name='pcm0_in Gain 0 Volume' 80% 80%"
+ cset "name='pcm0_in Gain 0 Volume' 0"
+
+ cset "name='pcm1_in Gain 0 Switch' off"
+ cset "name='pcm1_in Gain 0 Volume' 0%"
+
+ # codec0_out settings (used if ssp2 is connected to aif1)
+ cset "name='codec_out0 mix 0 codec_in0 Switch' off"
+ cset "name='codec_out0 mix 0 codec_in1 Switch' off"
+ cset "name='codec_out0 mix 0 media_loop1_in Switch' off"
+ cset "name='codec_out0 mix 0 media_loop2_in Switch' off"
+ cset "name='codec_out0 mix 0 pcm0_in Switch' on"
+ cset "name='codec_out0 mix 0 pcm1_in Switch' off"
+ cset "name='codec_out0 mix 0 sprot_loop_in Switch' off"
+ cset "name='codec_out0 Gain 0 Switch' on"
+ cset "name='codec_out0 Gain 0 Volume' 0"
+
+ # modem_out settings (used if ssp0 is connected to aif2)
+ cset "name='modem_out mix 0 codec_in0 Switch' off"
+ cset "name='modem_out mix 0 codec_in1 Switch' off"
+ cset "name='modem_out mix 0 media_loop1_in Switch' off"
+ cset "name='modem_out mix 0 media_loop2_in Switch' off"
+ cset "name='modem_out mix 0 pcm0_in Switch' on"
+ cset "name='modem_out mix 0 pcm1_in Switch' off"
+ cset "name='modem_out mix 0 sprot_loop_in Switch' off"
+ cset "name='modem_out Gain 0 Switch' on"
+ cset "name='modem_out Gain 0 Volume' 0"
+
+ # input settings
+ # pcm1_out settings
+
+ # input used when SSP2 is connected
+ cset "name='codec_in0 Gain 0 Switch' on"
+ cset "name='codec_in0 Gain 0 Volume' 0"
- cset "name='codec_out1 Gain 0 Ramp Delay' 50"
- cset "name='codec_out1 Gain 0 Switch' on"
- cset "name='codec_out1 Gain 0 Volume' 70% 70%"
+ # input used when SSP0 is connected
+ cset "name='modem_in Gain 0 Switch' on"
+ cset "name='modem_in Gain 0 Volume' 0"
- # Enable audio input path
- cset "name='pcm1_out mix 0 media_loop2_in Switch' on"
- cset "name='media_loop2_out mix 0 codec_in0 Switch' on"
+ cset "name='pcm1_out mix 0 codec_in0 Switch' on"
+ cset "name='pcm1_out mix 0 modem_in Switch' on"
+ cset "name='pcm1_out mix 0 codec_in1 Switch' off"
+ cset "name='pcm1_out mix 0 media_loop1_in Switch' off"
+ cset "name='pcm1_out mix 0 media_loop2_in Switch' off"
+ cset "name='pcm1_out mix 0 pcm0_in Switch' off"
+ cset "name='pcm1_out mix 0 pcm1_in Switch' off"
+ cset "name='pcm1_out mix 0 sprot_loop_in Switch' off"
- cset "name='codec_in0 Gain 0 Ramp Delay' 50"
- cset "name='codec_in0 Gain 0 Switch' on"
- cset "name='codec_in0 Gain 0 Volume' 80% 80%"
+ cset "name='pcm1_out Gain 0 Switch' on"
+ cset "name='pcm1_out Gain 0 Volume' 0"
- cset "name='media_loop2_out Gain 0 Ramp Delay' 50"
- cset "name='media_loop2_out Gain 0 Switch' on"
- cset "name='media_loop2_out Gain 0 Volume' 80% 80%"
+ # disable codec_out1
+ cset "name='codec_out1 mix 0 codec_in0 Switch' off"
+ cset "name='codec_out1 mix 0 codec_in1 Switch' off"
+ cset "name='codec_out1 mix 0 media_loop1_in Switch' off"
+ cset "name='codec_out1 mix 0 media_loop2_in Switch' off"
+ cset "name='codec_out1 mix 0 pcm0_in Switch' off"
+ cset "name='codec_out1 mix 0 pcm1_in Switch' off"
+ cset "name='codec_out1 mix 0 sprot_loop_in Switch' off"
+ cset "name='codec_out1 Gain 0 Switch' off"
+ cset "name='codec_out1 Gain 0 Volume' 0%"
- cset "name='pcm1_out Gain 0 Ramp Delay' 50"
- cset "name='pcm1_out Gain 0 Switch' on"
- cset "name='pcm1_out Gain 0 Volume' 80% 80%"
+ # disable codec_in1
+ cset "name='codec_in1 Gain 0 Switch' off"
+ cset "name='codec_in1 Gain 0 Volume' 0%"
+
+ # disable all loops
+ cset "name='media_loop1_out mix 0 codec_in0 Switch' off"
+ cset "name='media_loop1_out mix 0 codec_in1 Switch' off"
+ cset "name='media_loop1_out mix 0 media_loop1_in Switch' off"
+ cset "name='media_loop1_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop1_out mix 0 pcm0_in Switch' off"
+ cset "name='media_loop1_out mix 0 pcm1_in Switch' off"
+ cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='media_loop2_out mix 0 codec_in0 Switch' off"
+ cset "name='media_loop2_out mix 0 codec_in1 Switch' off"
+ cset "name='media_loop2_out mix 0 media_loop1_in Switch' off"
+ cset "name='media_loop2_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop2_out mix 0 pcm0_in Switch' off"
+ cset "name='media_loop2_out mix 0 pcm1_in Switch' off"
+ cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='sprot_loop_out mix 0 codec_in0 Switch' off"
+ cset "name='sprot_loop_out mix 0 codec_in1 Switch' off"
+ cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off"
+ cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off"
+ cset "name='sprot_loop_out mix 0 pcm0_in Switch' off"
+ cset "name='sprot_loop_out mix 0 pcm1_in Switch' off"
+ cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off"
# Output Configuration
- cset "name='DAC L2 Mux' 'IF1 DAC'"
- cset "name='DAC R2 Mux' 'IF1 DAC'"
+ cset "name='DAC1 L Mux' IF1 DAC"
+ cset "name='DAC1 R Mux' IF1 DAC"
+ cset "name='DAC1 MIXL DAC1 Switch' 1"
+ cset "name='DAC1 MIXR DAC1 Switch' 1"
+ cset "name='Stereo DAC MIXL DAC L1 Switch' 1"
+ cset "name='Stereo DAC MIXR DAC R1 Switch' 1"
+
+ cset "name='DAC L2 Mux' IF2 DAC"
+ cset "name='DAC R2 Mux' IF2 DAC"
cset "name='Mono DAC MIXL DAC L2 Switch' on"
cset "name='Mono DAC MIXR DAC R2 Switch' on"
cset "name='DAC2 Playback Switch' on"
+ cset "name='HPOVOL MIXL DAC1 Switch' on"
+ cset "name='HPOVOL MIXR DAC1 Switch' on"
cset "name='HPOVOL MIXL DAC2 Switch' on"
cset "name='HPOVOL MIXR DAC2 Switch' on"
cset "name='HPO MIX HPVOL Switch' on"
cset "name='HPOVOL L Switch' on"
cset "name='HPOVOL R Switch' on"
+ cset "name='SPK MIXL DAC L1 Switch' on"
+ cset "name='SPK MIXR DAC R1 Switch' on"
cset "name='SPK MIXL DAC L2 Switch' on"
cset "name='SPK MIXR DAC R2 Switch' on"
cset "name='SPOL MIX SPKVOL L Switch' on"
@@ -105,15 +200,18 @@ SectionDevice."Speaker" {
}
ConflictingDevice [
- "Headphone"
+ "Headphones"
]
EnableSequence [
cdev "hw:chtrt5645"
+ cset "name='Headphone Switch' off"
+ cset "name='Headphone Channel Switch' off"
+
cset "name='Ext Spk Switch' on"
cset "name='Speaker Channel Switch' on"
- cset "name='Speaker Playback Volume' 39"
+ cset "name='Speaker Playback Volume' 31"
]
DisableSequence [
@@ -124,7 +222,7 @@ SectionDevice."Speaker" {
]
}
-SectionDevice."Headphone".0 {
+SectionDevice."Headphones" {
Comment "Headphones"
Value {
@@ -140,9 +238,12 @@ SectionDevice."Headphone".0 {
EnableSequence [
cdev "hw:chtrt5645"
+ cset "name='Ext Spk Switch' off"
+ cset "name='Speaker Channel Switch' off"
+
cset "name='Headphone Switch' on"
cset "name='Headphone Channel Switch' on"
- cset "name='Headphone Playback Volume' 39"
+ cset "name='Headphone Playback Volume' 31"
]
DisableSequence [
--
2.13.6
From 4c0d983d7b4402e7275455ac43f39049b45037ea Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Sat, 28 Apr 2018 21:51:57 +0200
Subject: [PATCH 12/26] conf/ucm: chtrt5645: Microphone recording fixes
Apply microphone recording changes from:
https://github.com/plbossart/UCM.git
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[hdegoede@redhat.com: Drop non generic DMIC changes]
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/conf/ucm/chtrt5645/HiFi.conf | 50 ++++++++++++++++++++++++++++++++--------
1 file changed, 41 insertions(+), 9 deletions(-)
diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf
index f63392d4..b97a9c00 100644
--- a/src/conf/ucm/chtrt5645/HiFi.conf
+++ b/src/conf/ucm/chtrt5645/HiFi.conf
@@ -4,8 +4,8 @@ SectionVerb {
TQ "HiFi"
# ALSA PCM device for HiFi
- PlaybackPCM "hw:chtrt5645,0"
- CapturePCM "hw:chtrt5645,0"
+ PlaybackPCM "hw:chtrt5645"
+ CapturePCM "hw:chtrt5645"
}
EnableSequence [
@@ -255,7 +255,7 @@ SectionDevice."Headphones" {
}
SectionDevice."Mic".0 {
- Comment "Internal Microphone"
+ Comment "Internal Analog Microphones"
Value {
CaptureChannels "2"
@@ -266,15 +266,33 @@ SectionDevice."Mic".0 {
cdev "hw:chtrt5645"
cset "name='Int Mic Switch' on"
- cset "name='Sto1 ADC MIXL ADC2 Switch' on"
- cset "name='Sto1 ADC MIXR ADC2 Switch' on"
+
+ cset "name='Sto1 ADC MIXL ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXL ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+ cset "name='Mono ADC MIXL ADC2 Switch' off"
+ cset "name='Mono ADC MIXR ADC2 Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' on"
+ cset "name='RECMIXR BST1 Switch' on"
+
]
DisableSequence [
cdev "hw:chtrt5645"
- cset "name='Sto1 ADC MIXL ADC2 Switch' off"
- cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXL ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' off"
+ cset "name='Mono ADC MIXL ADC1 Switch' off"
+ cset "name='Mono ADC MIXR ADC1 Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' off"
+ cset "name='RECMIXR BST1 Switch' off"
+
cset "name='Int Mic Switch' off"
]
}
@@ -292,19 +310,33 @@ SectionDevice."HSMic".0 {
cdev "hw:chtrt5645"
cset "name='Headset Mic Switch' on"
- cset "name='RECMIXL BST1 Switch' on"
- cset "name='RECMIXR BST1 Switch' on"
+
cset "name='Sto1 ADC MIXL ADC1 Switch' on"
cset "name='Sto1 ADC MIXR ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXL ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+ cset "name='Mono ADC MIXL ADC2 Switch' off"
+ cset "name='Mono ADC MIXR ADC2 Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' on"
+ cset "name='RECMIXR BST1 Switch' on"
+
]
DisableSequence [
cdev "hw:chtrt5645"
cset "name='Headset Mic Switch' off"
+
cset "name='RECMIXL BST1 Switch' off"
cset "name='RECMIXR BST1 Switch' off"
cset "name='Sto1 ADC MIXL ADC1 Switch' off"
cset "name='Sto1 ADC MIXR ADC1 Switch' off"
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+
]
}
--
2.13.6
From 76bc53e69a12163a86e6746fd3011de6b1652043 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Sat, 28 Apr 2018 21:51:58 +0200
Subject: [PATCH 13/26] conf/ucm: chtrt5645: Fix recording from internal analog
microphone
The internal analog mic switch is called 'Int Analog Mic Switch'
(not 'Int Mic Switch') and is connected to BST2 not BST1.
Also change the analog mic volume levels so that we get better
audio / less noise.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/conf/ucm/chtrt5645/HiFi.conf | 38 ++++++++++++++++++++++++--------------
1 file changed, 24 insertions(+), 14 deletions(-)
diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf
index b97a9c00..e81866cf 100644
--- a/src/conf/ucm/chtrt5645/HiFi.conf
+++ b/src/conf/ucm/chtrt5645/HiFi.conf
@@ -160,15 +160,24 @@ SectionVerb {
# Input Configuration
cset "name='Stereo1 DMIC Mux' 0"
+ cset "name='Stereo1 ADC1 Mux' 1"
cset "name='Stereo1 ADC2 Mux' 1"
cset "name='ADC Capture Switch' on"
- cset "name='ADC Capture Volume' 31"
- cset "name='ADC Boost Capture Volume' 3"
- cset "name='Mono ADC Capture Volume' 63"
- cset "name='Mono ADC Boost Capture Volume' 2"
- cset "name='IN Capture Volume' 63"
+ # 55/63
+ cset "name='ADC Capture Volume' 55"
+ # set ADC Boost to 0/3, higher vals cause a lot of white noise
+ cset "name='ADC Boost Capture Volume' 0"
+ # 55/63
+ cset "name='Mono ADC Capture Volume' 55"
+ # 0/3
+ cset "name='Mono ADC Boost Capture Volume' 0"
+ # 27/31
+ cset "name='IN Capture Volume' 27"
cset "name='I2S2 Func Switch' on"
-
+ # 3/12 the headphone mic tends to be quite loud
+ cset "name='IN1 Boost' 3"
+ # 8/8 the internal analog mic tends to be quite soft
+ cset "name='IN2 Boost' 8"
]
DisableSequence [
@@ -243,7 +252,8 @@ SectionDevice."Headphones" {
cset "name='Headphone Switch' on"
cset "name='Headphone Channel Switch' on"
- cset "name='Headphone Playback Volume' 31"
+ # 25/39 higher values cause crackling on some boards
+ cset "name='Headphone Playback Volume' 25"
]
DisableSequence [
@@ -255,7 +265,7 @@ SectionDevice."Headphones" {
}
SectionDevice."Mic".0 {
- Comment "Internal Analog Microphones"
+ Comment "Internal Analog Microphone"
Value {
CaptureChannels "2"
@@ -265,7 +275,7 @@ SectionDevice."Mic".0 {
EnableSequence [
cdev "hw:chtrt5645"
- cset "name='Int Mic Switch' on"
+ cset "name='Int Analog Mic Switch' on"
cset "name='Sto1 ADC MIXL ADC1 Switch' on"
cset "name='Sto1 ADC MIXR ADC1 Switch' on"
@@ -277,8 +287,8 @@ SectionDevice."Mic".0 {
cset "name='Mono ADC MIXL ADC2 Switch' off"
cset "name='Mono ADC MIXR ADC2 Switch' off"
- cset "name='RECMIXL BST1 Switch' on"
- cset "name='RECMIXR BST1 Switch' on"
+ cset "name='RECMIXL BST2 Switch' on"
+ cset "name='RECMIXR BST2 Switch' on"
]
@@ -290,10 +300,10 @@ SectionDevice."Mic".0 {
cset "name='Mono ADC MIXL ADC1 Switch' off"
cset "name='Mono ADC MIXR ADC1 Switch' off"
- cset "name='RECMIXL BST1 Switch' off"
- cset "name='RECMIXR BST1 Switch' off"
+ cset "name='RECMIXL BST2 Switch' off"
+ cset "name='RECMIXR BST2 Switch' off"
- cset "name='Int Mic Switch' off"
+ cset "name='Int Analog Mic Switch' off"
]
}
--
2.13.6
From c22a7f423ddef2a1376bc84f2aafc0a167192ab6 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Sat, 28 Apr 2018 21:51:59 +0200
Subject: [PATCH 14/26] conf/ucm: chtrt5645: At config for the Asus T100HA
The Asus T100HA uses a digital mic rather then an analog one, add
long-name config specific for the T100HA, which is a copy of the standard
chtrt5645 config with the internal analog mic section replaced with one
for the digital mic found on the Asus T100HA.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
configure.ac | 1 +
.../ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN.conf | 5 +
.../HiFi.conf | 348 +++++++++++++++++++++
.../Makefile.am | 4 +
src/conf/ucm/Makefile.am | 1 +
5 files changed, 359 insertions(+)
create mode 100644 src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN.conf
create mode 100644 src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/HiFi.conf
create mode 100644 src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile.am
diff --git a/configure.ac b/configure.ac
index 5bc1994f..94baf055 100644
--- a/configure.ac
+++ b/configure.ac
@@ -717,6 +717,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \
src/conf/cards/Makefile \
src/conf/pcm/Makefile \
src/conf/ucm/Makefile \
+ src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile \
src/conf/ucm/broadwell-rt286/Makefile \
src/conf/ucm/broxton-rt298/Makefile \
src/conf/ucm/bytcr-rt5651/Makefile \
diff --git a/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN.conf b/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN.conf
new file mode 100644
index 00000000..5afe5f0b
--- /dev/null
+++ b/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN.conf
@@ -0,0 +1,5 @@
+Comment "Intel SoC Audio Device"
+SectionUseCase."HiFi" {
+ File "../ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/HiFi.conf"
+ Comment "Default"
+}
diff --git a/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/HiFi.conf b/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/HiFi.conf
new file mode 100644
index 00000000..78bf9823
--- /dev/null
+++ b/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/HiFi.conf
@@ -0,0 +1,348 @@
+SectionVerb {
+ # ALSA PCM
+ Value {
+ TQ "HiFi"
+
+ # ALSA PCM device for HiFi
+ PlaybackPCM "hw:chtrt5645"
+ CapturePCM "hw:chtrt5645"
+ }
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ # media mixer settings
+ # compress
+ cset "name='media0_in Gain 0 Switch' on"
+ cset "name='media0_in Gain 0 Volume' 0"
+
+ # normal
+ cset "name='media1_in Gain 0 Switch' on"
+ cset "name='media1_in Gain 0 Volume' 0"
+ # swm loopback
+ cset "name='media2_in Gain 0 Switch' off"
+ cset "name='media2_in Gain 0 Volume' 0%"
+ # deep buffer
+ cset "name='media3_in Gain 0 Switch' on"
+ cset "name='media3_in Gain 0 Volume' 0"
+
+ cset "name='media0_out mix 0 media0_in Switch' on"
+ cset "name='media0_out mix 0 media1_in Switch' on"
+ cset "name='media0_out mix 0 media2_in Switch' off"
+ cset "name='media0_out mix 0 media3_in Switch' on"
+
+ cset "name='media1_out mix 0 media0_in Switch' off"
+ cset "name='media1_out mix 0 media1_in Switch' off"
+ cset "name='media1_out mix 0 media2_in Switch' off"
+ cset "name='media1_out mix 0 media3_in Switch' off"
+
+ cset "name='pcm0_in Gain 0 Switch' on"
+ cset "name='pcm0_in Gain 0 Volume' 0"
+
+ cset "name='pcm1_in Gain 0 Switch' off"
+ cset "name='pcm1_in Gain 0 Volume' 0%"
+
+ # codec0_out settings (used if ssp2 is connected to aif1)
+ cset "name='codec_out0 mix 0 codec_in0 Switch' off"
+ cset "name='codec_out0 mix 0 codec_in1 Switch' off"
+ cset "name='codec_out0 mix 0 media_loop1_in Switch' off"
+ cset "name='codec_out0 mix 0 media_loop2_in Switch' off"
+ cset "name='codec_out0 mix 0 pcm0_in Switch' on"
+ cset "name='codec_out0 mix 0 pcm1_in Switch' off"
+ cset "name='codec_out0 mix 0 sprot_loop_in Switch' off"
+ cset "name='codec_out0 Gain 0 Switch' on"
+ cset "name='codec_out0 Gain 0 Volume' 0"
+
+ # modem_out settings (used if ssp0 is connected to aif2)
+ cset "name='modem_out mix 0 codec_in0 Switch' off"
+ cset "name='modem_out mix 0 codec_in1 Switch' off"
+ cset "name='modem_out mix 0 media_loop1_in Switch' off"
+ cset "name='modem_out mix 0 media_loop2_in Switch' off"
+ cset "name='modem_out mix 0 pcm0_in Switch' on"
+ cset "name='modem_out mix 0 pcm1_in Switch' off"
+ cset "name='modem_out mix 0 sprot_loop_in Switch' off"
+ cset "name='modem_out Gain 0 Switch' on"
+ cset "name='modem_out Gain 0 Volume' 0"
+
+ # input settings
+ # pcm1_out settings
+
+ # input used when SSP2 is connected
+ cset "name='codec_in0 Gain 0 Switch' on"
+ cset "name='codec_in0 Gain 0 Volume' 0"
+
+ # input used when SSP0 is connected
+ cset "name='modem_in Gain 0 Switch' on"
+ cset "name='modem_in Gain 0 Volume' 0"
+
+ cset "name='pcm1_out mix 0 codec_in0 Switch' on"
+ cset "name='pcm1_out mix 0 modem_in Switch' on"
+ cset "name='pcm1_out mix 0 codec_in1 Switch' off"
+ cset "name='pcm1_out mix 0 media_loop1_in Switch' off"
+ cset "name='pcm1_out mix 0 media_loop2_in Switch' off"
+ cset "name='pcm1_out mix 0 pcm0_in Switch' off"
+ cset "name='pcm1_out mix 0 pcm1_in Switch' off"
+ cset "name='pcm1_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='pcm1_out Gain 0 Switch' on"
+ cset "name='pcm1_out Gain 0 Volume' 0"
+
+ # disable codec_out1
+ cset "name='codec_out1 mix 0 codec_in0 Switch' off"
+ cset "name='codec_out1 mix 0 codec_in1 Switch' off"
+ cset "name='codec_out1 mix 0 media_loop1_in Switch' off"
+ cset "name='codec_out1 mix 0 media_loop2_in Switch' off"
+ cset "name='codec_out1 mix 0 pcm0_in Switch' off"
+ cset "name='codec_out1 mix 0 pcm1_in Switch' off"
+ cset "name='codec_out1 mix 0 sprot_loop_in Switch' off"
+ cset "name='codec_out1 Gain 0 Switch' off"
+ cset "name='codec_out1 Gain 0 Volume' 0%"
+
+ # disable codec_in1
+ cset "name='codec_in1 Gain 0 Switch' off"
+ cset "name='codec_in1 Gain 0 Volume' 0%"
+
+ # disable all loops
+ cset "name='media_loop1_out mix 0 codec_in0 Switch' off"
+ cset "name='media_loop1_out mix 0 codec_in1 Switch' off"
+ cset "name='media_loop1_out mix 0 media_loop1_in Switch' off"
+ cset "name='media_loop1_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop1_out mix 0 pcm0_in Switch' off"
+ cset "name='media_loop1_out mix 0 pcm1_in Switch' off"
+ cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='media_loop2_out mix 0 codec_in0 Switch' off"
+ cset "name='media_loop2_out mix 0 codec_in1 Switch' off"
+ cset "name='media_loop2_out mix 0 media_loop1_in Switch' off"
+ cset "name='media_loop2_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop2_out mix 0 pcm0_in Switch' off"
+ cset "name='media_loop2_out mix 0 pcm1_in Switch' off"
+ cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='sprot_loop_out mix 0 codec_in0 Switch' off"
+ cset "name='sprot_loop_out mix 0 codec_in1 Switch' off"
+ cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off"
+ cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off"
+ cset "name='sprot_loop_out mix 0 pcm0_in Switch' off"
+ cset "name='sprot_loop_out mix 0 pcm1_in Switch' off"
+ cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off"
+
+ # Output Configuration
+ cset "name='DAC1 L Mux' IF1 DAC"
+ cset "name='DAC1 R Mux' IF1 DAC"
+ cset "name='DAC1 MIXL DAC1 Switch' 1"
+ cset "name='DAC1 MIXR DAC1 Switch' 1"
+ cset "name='Stereo DAC MIXL DAC L1 Switch' 1"
+ cset "name='Stereo DAC MIXR DAC R1 Switch' 1"
+
+ cset "name='DAC L2 Mux' IF2 DAC"
+ cset "name='DAC R2 Mux' IF2 DAC"
+ cset "name='Mono DAC MIXL DAC L2 Switch' on"
+ cset "name='Mono DAC MIXR DAC R2 Switch' on"
+ cset "name='DAC2 Playback Switch' on"
+
+ cset "name='HPOVOL MIXL DAC1 Switch' on"
+ cset "name='HPOVOL MIXR DAC1 Switch' on"
+ cset "name='HPOVOL MIXL DAC2 Switch' on"
+ cset "name='HPOVOL MIXR DAC2 Switch' on"
+ cset "name='HPO MIX HPVOL Switch' on"
+ cset "name='HPOVOL L Switch' on"
+ cset "name='HPOVOL R Switch' on"
+
+ cset "name='SPK MIXL DAC L1 Switch' on"
+ cset "name='SPK MIXR DAC R1 Switch' on"
+ cset "name='SPK MIXL DAC L2 Switch' on"
+ cset "name='SPK MIXR DAC R2 Switch' on"
+ cset "name='SPOL MIX SPKVOL L Switch' on"
+ cset "name='SPOR MIX SPKVOL R Switch' on"
+ cset "name='SPKVOL L Switch' on"
+ cset "name='SPKVOL R Switch' on"
+
+ # Input Configuration
+ cset "name='Stereo1 DMIC Mux' 0"
+ cset "name='Stereo1 ADC1 Mux' 1"
+ cset "name='Stereo1 ADC2 Mux' 1"
+ cset "name='ADC Capture Switch' on"
+ # 55/63
+ cset "name='ADC Capture Volume' 55"
+ # set ADC Boost to 0/3, higher vals cause a lot of white noise
+ cset "name='ADC Boost Capture Volume' 0"
+ # 55/63
+ cset "name='Mono ADC Capture Volume' 55"
+ # 0/3
+ cset "name='Mono ADC Boost Capture Volume' 0"
+ # 27/31
+ cset "name='IN Capture Volume' 27"
+ cset "name='I2S2 Func Switch' on"
+ # 3/12 the headphone mic tends to be quite loud
+ cset "name='IN1 Boost' 3"
+ # 8/8 the internal analog mic tends to be quite soft
+ cset "name='IN2 Boost' 8"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ # Disable audio output path
+ cset "name='codec_out1 mix 0 pcm0_in Switch' off"
+ cset "name='media0_out mix 0 media1_in Switch' off"
+
+ cset "name='media1_in Gain 0 Switch' off"
+ cset "name='pcm0_in Gain 0 Switch' off"
+ cset "name='codec_out1 Gain 0 Switch' off"
+
+ # Disable audio input path
+ cset "name='pcm1_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop2_out mix 0 codec_in0 Switch' off"
+
+ cset "name='media_loop2_out Gain 0 Switch' off"
+ cset "name='pcm1_out Gain 0 Switch' off"
+ cset "name='codec_in0 Gain 0 Switch' off"
+ ]
+}
+
+SectionDevice."Speaker" {
+ Comment "Speaker"
+
+ Value {
+ PlaybackChannels "2"
+ }
+
+ ConflictingDevice [
+ "Headphones"
+ ]
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headphone Switch' off"
+ cset "name='Headphone Channel Switch' off"
+
+ cset "name='Ext Spk Switch' on"
+ cset "name='Speaker Channel Switch' on"
+ cset "name='Speaker Playback Volume' 31"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Ext Spk Switch' off"
+ cset "name='Speaker Channel Switch' off"
+ ]
+}
+
+SectionDevice."Headphones" {
+ Comment "Headphones"
+
+ Value {
+ PlaybackChannels "2"
+ JackControl "Headphone Jack"
+ JackHWMute "Speaker"
+ }
+
+ ConflictingDevice [
+ "Speaker"
+ ]
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Ext Spk Switch' off"
+ cset "name='Speaker Channel Switch' off"
+
+ cset "name='Headphone Switch' on"
+ cset "name='Headphone Channel Switch' on"
+ # 25/39 higher values cause crackling on some boards
+ cset "name='Headphone Playback Volume' 25"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headphone Switch' off"
+ cset "name='Headphone Channel Switch' off"
+ ]
+}
+
+SectionDevice."DMic".0 {
+ Comment "Internal Microphone"
+
+ Value {
+ CaptureChannels "2"
+ CapturePriority "150"
+ }
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Int Mic Switch' on"
+
+ cset "name='Stereo1 DMIC Mux' DMIC1"
+ cset "name='Stereo1 ADC2 Mux' DMIC"
+ cset "name='Mono ADC L2 Mux' DMIC"
+ cset "name='Mono ADC R2 Mux' DMIC"
+
+ cset "name='Sto1 ADC MIXL ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXL ADC2 Switch' on"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' on"
+ cset "name='Mono ADC MIXL ADC1 Switch' off"
+ cset "name='Mono ADC MIXR ADC1 Switch' off"
+ cset "name='Mono ADC MIXL ADC2 Switch' on"
+ cset "name='Mono ADC MIXR ADC2 Switch' on"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Sto1 ADC MIXL ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+ cset "name='Mono ADC MIXL ADC2 Switch' off"
+ cset "name='Mono ADC MIXR ADC2 Switch' off"
+ cset "name='Int Mic Switch' off"
+ ]
+}
+
+SectionDevice."HSMic".0 {
+ Comment "Headset Microphone"
+
+ Value {
+ CaptureChannels "2"
+ JackControl "Headset Mic Jack"
+ JackHWMute "DMic"
+ }
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headset Mic Switch' on"
+
+ cset "name='Sto1 ADC MIXL ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXL ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+ cset "name='Mono ADC MIXL ADC2 Switch' off"
+ cset "name='Mono ADC MIXR ADC2 Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' on"
+ cset "name='RECMIXR BST1 Switch' on"
+
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headset Mic Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' off"
+ cset "name='RECMIXR BST1 Switch' off"
+ cset "name='Sto1 ADC MIXL ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' off"
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+
+ ]
+}
diff --git a/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile.am b/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile.am
new file mode 100644
index 00000000..dd8b372a
--- /dev/null
+++ b/src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile.am
@@ -0,0 +1,4 @@
+alsaconfigdir = @ALSA_CONFIG_DIR@
+ucmdir = $(alsaconfigdir)/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN
+ucm_DATA = ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN.conf HiFi.conf
+EXTRA_DIST = $(ucm_DATA)
diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am
index 3b4f13a8..e496ca89 100644
--- a/src/conf/ucm/Makefile.am
+++ b/src/conf/ucm/Makefile.am
@@ -1,4 +1,5 @@
SUBDIRS=\
+ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN \
broadwell-rt286 \
broxton-rt298 \
bytcr-rt5651 \
--
2.13.6
From 5a2df9449d0b17e3579bde60ba48244ba24ea604 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Sat, 28 Apr 2018 21:52:00 +0200
Subject: [PATCH 15/26] conf/ucm: chtrt5645: At config for the Lenovo Ideapad
Miix 320
The Lenovo Ideapad Miix 320 uses a digital mic connected to the DMIC2 input
(unlike the Asus T100HA which has it connected to the DMIC1 input), add a
long-name config specific for the Miix 320, which is a copy of the standard
chtrt5645 config with the internal analog mic section replaced with one
for a digital mic connected to the DMIC2 input.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
configure.ac | 1 +
.../HiFi.conf | 350 +++++++++++++++++++++
...ENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf | 5 +
.../Makefile.am | 4 +
src/conf/ucm/Makefile.am | 1 +
5 files changed, 361 insertions(+)
create mode 100644 src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/HiFi.conf
create mode 100644 src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf
create mode 100644 src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile.am
diff --git a/configure.ac b/configure.ac
index 94baf055..3ee989eb 100644
--- a/configure.ac
+++ b/configure.ac
@@ -729,6 +729,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \
src/conf/ucm/gpd-win-pocket-rt5645/Makefile \
src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \
src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \
+ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \
src/conf/ucm/PandaBoard/Makefile \
src/conf/ucm/PandaBoardES/Makefile \
src/conf/ucm/PAZ00/Makefile \
diff --git a/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/HiFi.conf b/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/HiFi.conf
new file mode 100644
index 00000000..8cc9c7f2
--- /dev/null
+++ b/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/HiFi.conf
@@ -0,0 +1,350 @@
+SectionVerb {
+ # ALSA PCM
+ Value {
+ TQ "HiFi"
+
+ # ALSA PCM device for HiFi
+ PlaybackPCM "hw:chtrt5645"
+ CapturePCM "hw:chtrt5645"
+ }
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ # media mixer settings
+ # compress
+ cset "name='media0_in Gain 0 Switch' on"
+ cset "name='media0_in Gain 0 Volume' 0"
+
+ # normal
+ cset "name='media1_in Gain 0 Switch' on"
+ cset "name='media1_in Gain 0 Volume' 0"
+ # swm loopback
+ cset "name='media2_in Gain 0 Switch' off"
+ cset "name='media2_in Gain 0 Volume' 0%"
+ # deep buffer
+ cset "name='media3_in Gain 0 Switch' on"
+ cset "name='media3_in Gain 0 Volume' 0"
+
+ cset "name='media0_out mix 0 media0_in Switch' on"
+ cset "name='media0_out mix 0 media1_in Switch' on"
+ cset "name='media0_out mix 0 media2_in Switch' off"
+ cset "name='media0_out mix 0 media3_in Switch' on"
+
+ cset "name='media1_out mix 0 media0_in Switch' off"
+ cset "name='media1_out mix 0 media1_in Switch' off"
+ cset "name='media1_out mix 0 media2_in Switch' off"
+ cset "name='media1_out mix 0 media3_in Switch' off"
+
+ cset "name='pcm0_in Gain 0 Switch' on"
+ cset "name='pcm0_in Gain 0 Volume' 0"
+
+ cset "name='pcm1_in Gain 0 Switch' off"
+ cset "name='pcm1_in Gain 0 Volume' 0%"
+
+ # codec0_out settings (used if ssp2 is connected to aif1)
+ cset "name='codec_out0 mix 0 codec_in0 Switch' off"
+ cset "name='codec_out0 mix 0 codec_in1 Switch' off"
+ cset "name='codec_out0 mix 0 media_loop1_in Switch' off"
+ cset "name='codec_out0 mix 0 media_loop2_in Switch' off"
+ cset "name='codec_out0 mix 0 pcm0_in Switch' on"
+ cset "name='codec_out0 mix 0 pcm1_in Switch' off"
+ cset "name='codec_out0 mix 0 sprot_loop_in Switch' off"
+ cset "name='codec_out0 Gain 0 Switch' on"
+ cset "name='codec_out0 Gain 0 Volume' 0"
+
+ # modem_out settings (used if ssp0 is connected to aif2)
+ cset "name='modem_out mix 0 codec_in0 Switch' off"
+ cset "name='modem_out mix 0 codec_in1 Switch' off"
+ cset "name='modem_out mix 0 media_loop1_in Switch' off"
+ cset "name='modem_out mix 0 media_loop2_in Switch' off"
+ cset "name='modem_out mix 0 pcm0_in Switch' on"
+ cset "name='modem_out mix 0 pcm1_in Switch' off"
+ cset "name='modem_out mix 0 sprot_loop_in Switch' off"
+ cset "name='modem_out Gain 0 Switch' on"
+ cset "name='modem_out Gain 0 Volume' 0"
+
+ # input settings
+ # pcm1_out settings
+
+ # input used when SSP2 is connected
+ cset "name='codec_in0 Gain 0 Switch' on"
+ cset "name='codec_in0 Gain 0 Volume' 0"
+
+ # input used when SSP0 is connected
+ cset "name='modem_in Gain 0 Switch' on"
+ cset "name='modem_in Gain 0 Volume' 0"
+
+ cset "name='pcm1_out mix 0 codec_in0 Switch' on"
+ cset "name='pcm1_out mix 0 modem_in Switch' on"
+ cset "name='pcm1_out mix 0 codec_in1 Switch' off"
+ cset "name='pcm1_out mix 0 media_loop1_in Switch' off"
+ cset "name='pcm1_out mix 0 media_loop2_in Switch' off"
+ cset "name='pcm1_out mix 0 pcm0_in Switch' off"
+ cset "name='pcm1_out mix 0 pcm1_in Switch' off"
+ cset "name='pcm1_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='pcm1_out Gain 0 Switch' on"
+ cset "name='pcm1_out Gain 0 Volume' 0"
+
+ # disable codec_out1
+ cset "name='codec_out1 mix 0 codec_in0 Switch' off"
+ cset "name='codec_out1 mix 0 codec_in1 Switch' off"
+ cset "name='codec_out1 mix 0 media_loop1_in Switch' off"
+ cset "name='codec_out1 mix 0 media_loop2_in Switch' off"
+ cset "name='codec_out1 mix 0 pcm0_in Switch' off"
+ cset "name='codec_out1 mix 0 pcm1_in Switch' off"
+ cset "name='codec_out1 mix 0 sprot_loop_in Switch' off"
+ cset "name='codec_out1 Gain 0 Switch' off"
+ cset "name='codec_out1 Gain 0 Volume' 0%"
+
+ # disable codec_in1
+ cset "name='codec_in1 Gain 0 Switch' off"
+ cset "name='codec_in1 Gain 0 Volume' 0%"
+
+ # disable all loops
+ cset "name='media_loop1_out mix 0 codec_in0 Switch' off"
+ cset "name='media_loop1_out mix 0 codec_in1 Switch' off"
+ cset "name='media_loop1_out mix 0 media_loop1_in Switch' off"
+ cset "name='media_loop1_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop1_out mix 0 pcm0_in Switch' off"
+ cset "name='media_loop1_out mix 0 pcm1_in Switch' off"
+ cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='media_loop2_out mix 0 codec_in0 Switch' off"
+ cset "name='media_loop2_out mix 0 codec_in1 Switch' off"
+ cset "name='media_loop2_out mix 0 media_loop1_in Switch' off"
+ cset "name='media_loop2_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop2_out mix 0 pcm0_in Switch' off"
+ cset "name='media_loop2_out mix 0 pcm1_in Switch' off"
+ cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off"
+
+ cset "name='sprot_loop_out mix 0 codec_in0 Switch' off"
+ cset "name='sprot_loop_out mix 0 codec_in1 Switch' off"
+ cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off"
+ cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off"
+ cset "name='sprot_loop_out mix 0 pcm0_in Switch' off"
+ cset "name='sprot_loop_out mix 0 pcm1_in Switch' off"
+ cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off"
+
+ # Output Configuration
+ cset "name='DAC1 L Mux' IF1 DAC"
+ cset "name='DAC1 R Mux' IF1 DAC"
+ cset "name='DAC1 MIXL DAC1 Switch' 1"
+ cset "name='DAC1 MIXR DAC1 Switch' 1"
+ cset "name='Stereo DAC MIXL DAC L1 Switch' 1"
+ cset "name='Stereo DAC MIXR DAC R1 Switch' 1"
+
+ cset "name='DAC L2 Mux' IF2 DAC"
+ cset "name='DAC R2 Mux' IF2 DAC"
+ cset "name='Mono DAC MIXL DAC L2 Switch' on"
+ cset "name='Mono DAC MIXR DAC R2 Switch' on"
+ cset "name='DAC2 Playback Switch' on"
+
+ cset "name='HPOVOL MIXL DAC1 Switch' on"
+ cset "name='HPOVOL MIXR DAC1 Switch' on"
+ cset "name='HPOVOL MIXL DAC2 Switch' on"
+ cset "name='HPOVOL MIXR DAC2 Switch' on"
+ cset "name='HPO MIX HPVOL Switch' on"
+ cset "name='HPOVOL L Switch' on"
+ cset "name='HPOVOL R Switch' on"
+
+ cset "name='SPK MIXL DAC L1 Switch' on"
+ cset "name='SPK MIXR DAC R1 Switch' on"
+ cset "name='SPK MIXL DAC L2 Switch' on"
+ cset "name='SPK MIXR DAC R2 Switch' on"
+ cset "name='SPOL MIX SPKVOL L Switch' on"
+ cset "name='SPOR MIX SPKVOL R Switch' on"
+ cset "name='SPKVOL L Switch' on"
+ cset "name='SPKVOL R Switch' on"
+
+ # Input Configuration
+ cset "name='Stereo1 DMIC Mux' 0"
+ cset "name='Stereo1 ADC1 Mux' 1"
+ cset "name='Stereo1 ADC2 Mux' 1"
+ cset "name='ADC Capture Switch' on"
+ # 55/63
+ cset "name='ADC Capture Volume' 55"
+ # set ADC Boost to 0/3, higher vals cause a lot of white noise
+ cset "name='ADC Boost Capture Volume' 0"
+ # 55/63
+ cset "name='Mono ADC Capture Volume' 55"
+ # 0/3
+ cset "name='Mono ADC Boost Capture Volume' 0"
+ # 27/31
+ cset "name='IN Capture Volume' 27"
+ cset "name='I2S2 Func Switch' on"
+ # 3/12 the headphone mic tends to be quite loud
+ cset "name='IN1 Boost' 3"
+ # 8/8 the internal analog mic tends to be quite soft
+ cset "name='IN2 Boost' 8"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ # Disable audio output path
+ cset "name='codec_out1 mix 0 pcm0_in Switch' off"
+ cset "name='media0_out mix 0 media1_in Switch' off"
+
+ cset "name='media1_in Gain 0 Switch' off"
+ cset "name='pcm0_in Gain 0 Switch' off"
+ cset "name='codec_out1 Gain 0 Switch' off"
+
+ # Disable audio input path
+ cset "name='pcm1_out mix 0 media_loop2_in Switch' off"
+ cset "name='media_loop2_out mix 0 codec_in0 Switch' off"
+
+ cset "name='media_loop2_out Gain 0 Switch' off"
+ cset "name='pcm1_out Gain 0 Switch' off"
+ cset "name='codec_in0 Gain 0 Switch' off"
+ ]
+}
+
+SectionDevice."Speaker" {
+ Comment "Speaker"
+
+ Value {
+ PlaybackChannels "2"
+ }
+
+ ConflictingDevice [
+ "Headphones"
+ ]
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headphone Switch' off"
+ cset "name='Headphone Channel Switch' off"
+
+ cset "name='Ext Spk Switch' on"
+ cset "name='Speaker Channel Switch' on"
+ cset "name='Speaker Playback Volume' 31"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Ext Spk Switch' off"
+ cset "name='Speaker Channel Switch' off"
+ ]
+}
+
+SectionDevice."Headphones" {
+ Comment "Headphones"
+
+ Value {
+ PlaybackChannels "2"
+ JackControl "Headphone Jack"
+ JackHWMute "Speaker"
+ }
+
+ ConflictingDevice [
+ "Speaker"
+ ]
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Ext Spk Switch' off"
+ cset "name='Speaker Channel Switch' off"
+
+ cset "name='Headphone Switch' on"
+ cset "name='Headphone Channel Switch' on"
+ # 25/39 higher values cause crackling on some boards
+ cset "name='Headphone Playback Volume' 25"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headphone Switch' off"
+ cset "name='Headphone Channel Switch' off"
+ ]
+}
+
+SectionDevice."DMic".0 {
+ Comment "Internal Microphone"
+
+ Value {
+ CaptureChannels "2"
+ CapturePriority "150"
+ }
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Int Mic Switch' on"
+
+ cset "name='Stereo1 DMIC Mux' DMIC2"
+ cset "name='Stereo1 ADC2 Mux' DMIC"
+ cset "name='Mono DMIC L Mux' DMIC2"
+ cset "name='Mono DMIC R Mux' DMIC2"
+ cset "name='Mono ADC L2 Mux' DMIC"
+ cset "name='Mono ADC R2 Mux' DMIC"
+
+ cset "name='Sto1 ADC MIXL ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXL ADC2 Switch' on"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' on"
+ cset "name='Mono ADC MIXL ADC1 Switch' off"
+ cset "name='Mono ADC MIXR ADC1 Switch' off"
+ cset "name='Mono ADC MIXL ADC2 Switch' on"
+ cset "name='Mono ADC MIXR ADC2 Switch' on"
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Sto1 ADC MIXL ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+ cset "name='Mono ADC MIXL ADC2 Switch' off"
+ cset "name='Mono ADC MIXR ADC2 Switch' off"
+ cset "name='Int Mic Switch' off"
+ ]
+}
+
+SectionDevice."HSMic".0 {
+ Comment "Headset Microphone"
+
+ Value {
+ CaptureChannels "2"
+ JackControl "Headset Mic Jack"
+ JackHWMute "DMic"
+ }
+
+ EnableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headset Mic Switch' on"
+
+ cset "name='Sto1 ADC MIXL ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' on"
+ cset "name='Sto1 ADC MIXL ADC2 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC2 Switch' off"
+
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+ cset "name='Mono ADC MIXL ADC2 Switch' off"
+ cset "name='Mono ADC MIXR ADC2 Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' on"
+ cset "name='RECMIXR BST1 Switch' on"
+
+ ]
+
+ DisableSequence [
+ cdev "hw:chtrt5645"
+
+ cset "name='Headset Mic Switch' off"
+
+ cset "name='RECMIXL BST1 Switch' off"
+ cset "name='RECMIXR BST1 Switch' off"
+ cset "name='Sto1 ADC MIXL ADC1 Switch' off"
+ cset "name='Sto1 ADC MIXR ADC1 Switch' off"
+ cset "name='Mono ADC MIXL ADC1 Switch' on"
+ cset "name='Mono ADC MIXR ADC1 Switch' on"
+
+ ]
+}
diff --git a/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf b/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf
new file mode 100644
index 00000000..50bda000
--- /dev/null
+++ b/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf
@@ -0,0 +1,5 @@
+Comment "Intel SoC Audio Device"
+SectionUseCase."HiFi" {
+ File "../LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/HiFi.conf"
+ Comment "Default"
+}
diff --git a/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile.am b/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile.am
new file mode 100644
index 00000000..18963c21
--- /dev/null
+++ b/src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile.am
@@ -0,0 +1,4 @@
+alsaconfigdir = @ALSA_CONFIG_DIR@
+ucmdir = $(alsaconfigdir)/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216
+ucm_DATA = LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216.conf HiFi.conf
+EXTRA_DIST = $(ucm_DATA)
diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am
index e496ca89..02df25f0 100644
--- a/src/conf/ucm/Makefile.am
+++ b/src/conf/ucm/Makefile.am
@@ -11,6 +11,7 @@ GoogleNyan \
gpd-win-pocket-rt5645 \
HDAudio-Gigabyte-ALC1220DualCodecs \
HDAudio-Lenovo-DualCodecs \
+LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \
PandaBoard \
PandaBoardES \
PAZ00 \
--
2.13.6
From 8ebb40c96970c913719a75deb2fe82ba2e257386 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 May 2018 16:23:21 +0200
Subject: [PATCH 16/26] conf/ucm: Add a UCM profile for Dell WD15 Dock
USB-audio
USB-audio device on Dell WD15 docking station provides two individual
PCM streams, one for headphone and another for line out. A UCM
profile gives the proper roles for these.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf | 5 +++++
src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 26 +++++++++++++++++++++++++
src/conf/ucm/Dell-WD15-Dock/Makefile.am | 4 ++++
src/conf/ucm/Makefile.am | 1 +
4 files changed, 36 insertions(+)
create mode 100644 src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf
create mode 100644 src/conf/ucm/Dell-WD15-Dock/HiFi.conf
create mode 100644 src/conf/ucm/Dell-WD15-Dock/Makefile.am
diff --git a/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf b/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf
new file mode 100644
index 00000000..f4f0a8d9
--- /dev/null
+++ b/src/conf/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf
@@ -0,0 +1,5 @@
+Comment "USB-audio on Dell WD15 docking station"
+SectionUseCase."HiFi" {
+ File "HiFi.conf"
+ Comment "Default"
+}
diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf
new file mode 100644
index 00000000..b07f3440
--- /dev/null
+++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf
@@ -0,0 +1,26 @@
+SectionDevice."Headphone" {
+ Comment "Headphone"
+
+ Value {
+ PlaybackChannels "2"
+ PlaybackPCM "hw:WD15Dock,0"
+ }
+}
+
+SectionDevice."LineOut" {
+ Comment "Line Out"
+
+ Value {
+ PlaybackChannels "2"
+ PlaybackPCM "hw:WD15Dock,1"
+ }
+}
+
+SectionDevice."Mic" {
+ Comment "Microphone"
+
+ Value {
+ CaptureChannels "2"
+ CapturePCM "hw:WD15Dock,0"
+ }
+}
diff --git a/src/conf/ucm/Dell-WD15-Dock/Makefile.am b/src/conf/ucm/Dell-WD15-Dock/Makefile.am
new file mode 100644
index 00000000..7ab58730
--- /dev/null
+++ b/src/conf/ucm/Dell-WD15-Dock/Makefile.am
@@ -0,0 +1,4 @@
+alsaconfigdir = @ALSA_CONFIG_DIR@
+ucmdir = $(alsaconfigdir)/ucm/Dell-WD15-Dock
+ucm_DATA = Dell-WD15-Dock.conf HiFi.conf
+EXTRA_DIST = $(ucm_DATA)
diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am
index 02df25f0..38fc6b25 100644
--- a/src/conf/ucm/Makefile.am
+++ b/src/conf/ucm/Makefile.am
@@ -7,6 +7,7 @@ chtrt5645 \
chtrt5645-mono-speaker-analog-mic \
DAISY-I2S \
DB410c \
+Dell-WD15-Dock \
GoogleNyan \
gpd-win-pocket-rt5645 \
HDAudio-Gigabyte-ALC1220DualCodecs \
--
2.13.6
From 44f499bb22f3923f966e11a234455e3d06936b8b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 3 May 2018 08:40:20 +0200
Subject: [PATCH 17/26] configure: Fix forgotten ucm entry
The previous commit forgot to add to configure.ac. Fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
configure.ac | 1 +
1 file changed, 1 insertion(+)
diff --git a/configure.ac b/configure.ac
index 3ee989eb..693b5d53 100644
--- a/configure.ac
+++ b/configure.ac
@@ -725,6 +725,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \
src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \
src/conf/ucm/DAISY-I2S/Makefile \
src/conf/ucm/DB410c/Makefile \
+ src/conf/ucm/Dell-WD15-Dock/Makefile \
src/conf/ucm/GoogleNyan/Makefile \
src/conf/ucm/gpd-win-pocket-rt5645/Makefile \
src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \
--
2.13.6
From 5eb78219f6bc0114cbb6518dae1c9256a63ec23a Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 15 May 2018 21:36:34 +0900
Subject: [PATCH 18/26] control: add a series of macro for offset of several
types of TLV
In development period for Linux v4.18, a series of SNDRV_CTL_TLVO_XXX
macro was introduced to kernel stuffs for position offset of TLV data.
This commit adds these macros to backport header in this library.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/sound/tlv.h | 16 ++++++++++++++++
1 file changed, 16 insertions(+)
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index b4df440c..2ea47685 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -41,6 +41,10 @@
#define SNDRV_CTL_TLVD_LENGTH(...) \
((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+/* Accessor offsets for TLV data items */
+#define SNDRV_CTL_TLVO_TYPE 0
+#define SNDRV_CTL_TLVO_LEN 1
+
#define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
#define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
@@ -60,6 +64,10 @@
SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
}
+/* Accessor offsets for min, mute and step items in dB scale type TLV */
+#define SNDRV_CTL_TLVO_DB_SCALE_MIN 2
+#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP 3
+
/* dB scale specified with min/max values instead of step */
#define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
@@ -74,6 +82,10 @@
SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
}
+/* Accessor offsets for min, max items in db-minmax types of TLV. */
+#define SNDRV_CTL_TLVO_DB_MINMAX_MIN 2
+#define SNDRV_CTL_TLVO_DB_MINMAX_MAX 3
+
/* linear volume between min_dB and max_dB (.01dB unit) */
#define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
@@ -82,6 +94,10 @@
SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
}
+/* Accessor offsets for min, max items in db-linear type of TLV. */
+#define SNDRV_CTL_TLVO_DB_LINEAR_MIN 2
+#define SNDRV_CTL_TLVO_DB_LINEAR_MAX 3
+
/* dB range container:
* Items in dB range container must be ordered by their values and by their
* dB values. This implies that larger values must correspond with larger
--
2.13.6
From f61193c96c52cbd99e22d9cbb94937374ce2cb39 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 15 May 2018 21:36:35 +0900
Subject: [PATCH 19/26] control: use position offset macro of TLV data
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/control/control.c | 12 ++++----
src/control/control_hw.c | 6 ++--
src/control/control_local.h | 1 +
src/control/tlv.c | 72 ++++++++++++++++++++++++---------------------
4 files changed, 51 insertions(+), 40 deletions(-)
diff --git a/src/control/control.c b/src/control/control.c
index 11f7815a..82cd1a05 100644
--- a/src/control/control.c
+++ b/src/control/control.c
@@ -938,10 +938,10 @@ int snd_ctl_elem_tlv_read(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id,
* and compare the returned value after ioctl for checking
* the validity of TLV.
*/
- tlv[0] = -1;
- tlv[1] = 0;
+ tlv[SNDRV_CTL_TLVO_TYPE] = -1;
+ tlv[SNDRV_CTL_TLVO_LEN] = 0;
err = snd_ctl_tlv_do(ctl, 0, id, tlv, tlv_size);
- if (err >= 0 && tlv[0] == (unsigned int)-1)
+ if (err >= 0 && tlv[SNDRV_CTL_TLVO_TYPE] == (unsigned int)-1)
err = -ENXIO;
return err;
}
@@ -967,7 +967,8 @@ int snd_ctl_elem_tlv_write(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id,
const unsigned int *tlv)
{
assert(ctl && id && (id->name[0] || id->numid) && tlv);
- return snd_ctl_tlv_do(ctl, 1, id, (unsigned int *)tlv, tlv[1] + 2 * sizeof(unsigned int));
+ return snd_ctl_tlv_do(ctl, 1, id, (unsigned int *)tlv,
+ tlv[SNDRV_CTL_TLVO_LEN] + 2 * sizeof(unsigned int));
}
/**
@@ -991,7 +992,8 @@ int snd_ctl_elem_tlv_command(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id,
const unsigned int *tlv)
{
assert(ctl && id && (id->name[0] || id->numid) && tlv);
- return snd_ctl_tlv_do(ctl, -1, id, (unsigned int *)tlv, tlv[1] + 2 * sizeof(unsigned int));
+ return snd_ctl_tlv_do(ctl, -1, id, (unsigned int *)tlv,
+ tlv[SNDRV_CTL_TLVO_LEN] + 2 * sizeof(unsigned int));
}
/**
diff --git a/src/control/control_hw.c b/src/control/control_hw.c
index 4cbd306f..68eca522 100644
--- a/src/control/control_hw.c
+++ b/src/control/control_hw.c
@@ -240,11 +240,13 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag,
return -errno;
}
if (op_flag == 0) {
- if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) {
+ unsigned int size;
+ size = xtlv->tlv[SNDRV_CTL_TLVO_LEN] + 2 * sizeof(unsigned int);
+ if (size > tlv_size) {
free(xtlv);
return -EFAULT;
}
- memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int));
+ memcpy(tlv, xtlv->tlv, size);
}
free(xtlv);
return 0;
diff --git a/src/control/control_local.h b/src/control/control_local.h
index 30218c6c..9568968e 100644
--- a/src/control/control_local.h
+++ b/src/control/control_local.h
@@ -20,6 +20,7 @@
*/
#include "local.h"
+#include <sound/tlv.h>
typedef struct _snd_ctl_ops {
int (*close)(snd_ctl_t *handle);
diff --git a/src/control/tlv.c b/src/control/tlv.c
index 467023b3..d6944b52 100644
--- a/src/control/tlv.c
+++ b/src/control/tlv.c
@@ -65,8 +65,8 @@ int snd_tlv_parse_dB_info(unsigned int *tlv,
int err;
*db_tlvp = NULL;
- type = tlv[0];
- size = tlv[1];
+ type = tlv[SNDRV_CTL_TLVO_TYPE];
+ size = tlv[SNDRV_CTL_TLVO_LEN];
tlv_size -= 2 * sizeof(int);
if (size > tlv_size) {
SNDERR("TLV size error");
@@ -83,7 +83,7 @@ int snd_tlv_parse_dB_info(unsigned int *tlv,
return err; /* error */
if (err > 0)
return err; /* found */
- len = int_index(tlv[1]) + 2;
+ len = int_index(tlv[SNDRV_CTL_TLVO_LEN]) + 2;
size -= len * sizeof(int);
tlv += len;
}
@@ -131,10 +131,10 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax,
{
int err;
- switch (tlv[0]) {
+ switch (tlv[SNDRV_CTL_TLVO_TYPE]) {
case SND_CTL_TLVT_DB_RANGE: {
unsigned int pos, len;
- len = int_index(tlv[1]);
+ len = int_index(tlv[SNDRV_CTL_TLVO_LEN]);
if (len > MAX_TLV_RANGE_SIZE)
return -EINVAL;
pos = 2;
@@ -167,22 +167,23 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax,
}
case SND_CTL_TLVT_DB_SCALE: {
int step;
- if (tlv[3] & 0x10000)
+ if (tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & 0x10000)
*min = SND_CTL_TLV_DB_GAIN_MUTE;
else
- *min = (int)tlv[2];
- step = (tlv[3] & 0xffff);
- *max = (int)tlv[2] + step * (rangemax - rangemin);
+ *min = (int)tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN];
+ step = (tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & 0xffff);
+ *max = (int)tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] +
+ step * (rangemax - rangemin);
return 0;
}
case SND_CTL_TLVT_DB_MINMAX:
case SND_CTL_TLVT_DB_LINEAR:
- *min = (int)tlv[2];
- *max = (int)tlv[3];
+ *min = (int)tlv[SNDRV_CTL_TLVO_DB_LINEAR_MIN];
+ *max = (int)tlv[SNDRV_CTL_TLVO_DB_LINEAR_MAX];
return 0;
case SND_CTL_TLVT_DB_MINMAX_MUTE:
*min = SND_CTL_TLV_DB_GAIN_MUTE;
- *max = (int)tlv[3];
+ *max = (int)tlv[SNDRV_CTL_TLVO_DB_MINMAX_MAX];
return 0;
}
return -EINVAL;
@@ -200,10 +201,12 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax,
int snd_tlv_convert_to_dB(unsigned int *tlv, long rangemin, long rangemax,
long volume, long *db_gain)
{
- switch (tlv[0]) {
+ unsigned int type = tlv[SNDRV_CTL_TLVO_TYPE];
+
+ switch (type) {
case SND_CTL_TLVT_DB_RANGE: {
unsigned int pos, len;
- len = int_index(tlv[1]);
+ len = int_index(tlv[SNDRV_CTL_TLVO_LEN]);
if (len > MAX_TLV_RANGE_SIZE)
return -EINVAL;
pos = 2;
@@ -220,9 +223,9 @@ int snd_tlv_convert_to_dB(unsigned int *tlv, long rangemin, long rangemax,
}
case SND_CTL_TLVT_DB_SCALE: {
int min, step, mute;
- min = tlv[2];
- step = (tlv[3] & 0xffff);
- mute = (tlv[3] >> 16) & 1;
+ min = tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN];
+ step = (tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & 0xffff);
+ mute = (tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] >> 16) & 1;
if (mute && volume <= rangemin)
*db_gain = SND_CTL_TLV_DB_GAIN_MUTE;
else
@@ -232,10 +235,10 @@ int snd_tlv_convert_to_dB(unsigned int *tlv, long rangemin, long rangemax,
case SND_CTL_TLVT_DB_MINMAX:
case SND_CTL_TLVT_DB_MINMAX_MUTE: {
int mindb, maxdb;
- mindb = tlv[2];
- maxdb = tlv[3];
+ mindb = tlv[SNDRV_CTL_TLVO_DB_MINMAX_MIN];
+ maxdb = tlv[SNDRV_CTL_TLVO_DB_MINMAX_MAX];
if (volume <= rangemin || rangemax <= rangemin) {
- if (tlv[0] == SND_CTL_TLVT_DB_MINMAX_MUTE)
+ if (type == SND_CTL_TLVT_DB_MINMAX_MUTE)
*db_gain = SND_CTL_TLV_DB_GAIN_MUTE;
else
*db_gain = mindb;
@@ -248,8 +251,8 @@ int snd_tlv_convert_to_dB(unsigned int *tlv, long rangemin, long rangemax,
}
#ifndef HAVE_SOFT_FLOAT
case SND_CTL_TLVT_DB_LINEAR: {
- int mindb = tlv[2];
- int maxdb = tlv[3];
+ int mindb = tlv[SNDRV_CTL_TLVO_DB_LINEAR_MIN];
+ int maxdb = tlv[SNDRV_CTL_TLVO_DB_LINEAR_MAX];
if (volume <= rangemin || rangemax <= rangemin)
*db_gain = mindb;
else if (volume >= rangemax)
@@ -289,11 +292,13 @@ int snd_tlv_convert_to_dB(unsigned int *tlv, long rangemin, long rangemax,
int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax,
long db_gain, long *value, int xdir)
{
- switch (tlv[0]) {
+ unsigned int type = tlv[SNDRV_CTL_TLVO_TYPE];
+
+ switch (type) {
case SND_CTL_TLVT_DB_RANGE: {
long dbmin, dbmax, prev_submax;
unsigned int pos, len;
- len = int_index(tlv[1]);
+ len = int_index(tlv[SNDRV_CTL_TLVO_LEN]);
if (len < 6 || len > MAX_TLV_RANGE_SIZE)
return -EINVAL;
pos = 2;
@@ -324,13 +329,14 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax,
return 0;
}
case SND_CTL_TLVT_DB_SCALE: {
- int min, step, max;
- min = tlv[2];
- step = (tlv[3] & 0xffff);
+ int min, step, max, mute;
+ min = tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN];
+ step = tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & 0xffff;
+ mute = tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & 0x10000;
max = min + (int)(step * (rangemax - rangemin));
if (db_gain <= min)
if (db_gain > SND_CTL_TLV_DB_GAIN_MUTE && xdir > 0 &&
- (tlv[3] & 0x10000))
+ mute)
*value = rangemin + 1;
else
*value = rangemin;
@@ -348,11 +354,11 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax,
case SND_CTL_TLVT_DB_MINMAX:
case SND_CTL_TLVT_DB_MINMAX_MUTE: {
int min, max;
- min = tlv[2];
- max = tlv[3];
+ min = tlv[SNDRV_CTL_TLVO_DB_MINMAX_MIN];
+ max = tlv[SNDRV_CTL_TLVO_DB_MINMAX_MAX];
if (db_gain <= min)
if (db_gain > SND_CTL_TLV_DB_GAIN_MUTE && xdir > 0 &&
- tlv[0] == SND_CTL_TLVT_DB_MINMAX_MUTE)
+ type == SND_CTL_TLVT_DB_MINMAX_MUTE)
*value = rangemin + 1;
else
*value = rangemin;
@@ -370,8 +376,8 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax,
#ifndef HAVE_SOFT_FLOAT
case SND_CTL_TLVT_DB_LINEAR: {
int min, max;
- min = tlv[2];
- max = tlv[3];
+ min = tlv[SNDRV_CTL_TLVO_DB_LINEAR_MIN];
+ max = tlv[SNDRV_CTL_TLVO_DB_LINEAR_MAX];
if (db_gain <= min)
*value = rangemin;
else if (db_gain >= max)
--
2.13.6
From 33f0888f0da191af49f17916f6548076d7125ea9 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 15 May 2018 21:36:36 +0900
Subject: [PATCH 20/26] hcontrol: use position offset macro of TLV data
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/control/hcontrol.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/src/control/hcontrol.c b/src/control/hcontrol.c
index 7e775248..0cac8956 100644
--- a/src/control/hcontrol.c
+++ b/src/control/hcontrol.c
@@ -870,7 +870,7 @@ int snd_hctl_elem_tlv_write(snd_hctl_elem_t *elem, const unsigned int *tlv)
{
assert(elem);
assert(tlv);
- assert(tlv[1] >= 4);
+ assert(tlv[SNDRV_CTL_TLVO_LEN] >= 4);
return snd_ctl_elem_tlv_write(elem->hctl->ctl, &elem->id, tlv);
}
@@ -886,7 +886,7 @@ int snd_hctl_elem_tlv_command(snd_hctl_elem_t *elem, const unsigned int *tlv)
{
assert(elem);
assert(tlv);
- assert(tlv[1] >= 4);
+ assert(tlv[SNDRV_CTL_TLVO_LEN] >= 4);
return snd_ctl_elem_tlv_command(elem->hctl->ctl, &elem->id, tlv);
}
--
2.13.6
From b0b720513ee35ec0a5afada0b240310f2f2e1b30 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 15 May 2018 21:36:37 +0900
Subject: [PATCH 21/26] pcm: hw: use position offset macro of TLV data
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/pcm/pcm_hw.c | 10 ++++++----
1 file changed, 6 insertions(+), 4 deletions(-)
diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c
index 65b198c5..a728b23b 100644
--- a/src/pcm/pcm_hw.c
+++ b/src/pcm/pcm_hw.c
@@ -1199,6 +1199,7 @@ snd_pcm_query_chmaps_from_hw(int card, int dev, int subdev,
snd_ctl_t *ctl;
snd_ctl_elem_id_t id = {0};
unsigned int tlv[2048], *start;
+ unsigned int type;
snd_pcm_chmap_query_t **map;
int i, ret, nums;
@@ -1223,9 +1224,10 @@ snd_pcm_query_chmaps_from_hw(int card, int dev, int subdev,
/* FIXME: the parser below assumes that the TLV only contains
* chmap-related blocks
*/
- if (tlv[0] != SND_CTL_TLVT_CONTAINER) {
- if (!is_chmap_type(tlv[0])) {
- SYSMSG("Invalid TLV type %d\n", tlv[0]);
+ type = tlv[SNDRV_CTL_TLVO_TYPE];
+ if (type != SND_CTL_TLVT_CONTAINER) {
+ if (!is_chmap_type(type)) {
+ SYSMSG("Invalid TLV type %d\n", type);
return NULL;
}
start = tlv;
@@ -1234,7 +1236,7 @@ snd_pcm_query_chmaps_from_hw(int card, int dev, int subdev,
unsigned int *p;
int size;
start = tlv + 2;
- size = tlv[1];
+ size = tlv[SNDRV_CTL_TLVO_LEN];
nums = 0;
for (p = start; size > 0; ) {
if (!is_chmap_type(p[0])) {
--
2.13.6
From 6dc2ed4090bee0e5516839173166c6a35c3d80be Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 15 May 2018 21:36:38 +0900
Subject: [PATCH 22/26] pcm: softvol: use position offset macro of TLV data
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/pcm/pcm_softvol.c | 11 +++++++----
1 file changed, 7 insertions(+), 4 deletions(-)
diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c
index 0eaeacef..f08208f8 100644
--- a/src/pcm/pcm_softvol.c
+++ b/src/pcm/pcm_softvol.c
@@ -31,6 +31,8 @@
#include "pcm_local.h"
#include "pcm_plugin.h"
+#include <sound/tlv.h>
+
#ifndef PIC
/* entry for static linking */
const char *_snd_module_pcm_softvol = "";
@@ -708,10 +710,11 @@ static void snd_pcm_softvol_dump(snd_pcm_t *pcm, snd_output_t *out)
static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo)
{
unsigned int tlv[4];
- tlv[0] = SND_CTL_TLVT_DB_SCALE;
- tlv[1] = 2 * sizeof(int);
- tlv[2] = (int)(svol->min_dB * 100);
- tlv[3] = (int)((svol->max_dB - svol->min_dB) * 100 / svol->max_val);
+ tlv[SNDRV_CTL_TLVO_TYPE] = SND_CTL_TLVT_DB_SCALE;
+ tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(int);
+ tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = (int)(svol->min_dB * 100);
+ tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] =
+ (int)((svol->max_dB - svol->min_dB) * 100 / svol->max_val);
return snd_ctl_elem_tlv_write(svol->ctl, &cinfo->id, tlv);
}
--
2.13.6
From 275a438d425c3477be21e1929061f0c1605876f2 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Tue, 15 May 2018 21:36:39 +0900
Subject: [PATCH 23/26] test: use position offset macro of TLV data
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
test/user-ctl-element-set.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c
index e94152b9..fee130e2 100644
--- a/test/user-ctl-element-set.c
+++ b/test/user-ctl-element-set.c
@@ -660,7 +660,7 @@ static int check_tlv(struct elem_set_trial *trial)
if (err < 0)
return err;
- len = tlv[1] + sizeof(unsigned int) * 2;
+ len = tlv[SNDRV_CTL_TLVO_LEN] + sizeof(unsigned int) * 2;
curr = malloc(len);
if (curr == NULL) {
free(tlv);
--
2.13.6
From 9ee3e4338f46f18a5cddc267bbb19c7a222c5801 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 22 May 2018 14:10:59 +0200
Subject: [PATCH 24/26] conf: USB-audio: Fix for Xonar U7 SPDIF device
Add the entry for Xonar U7 to make SPDIF working on it.
Reported-by: Steve Banks <eassbank@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
src/conf/cards/USB-Audio.conf | 1 +
1 file changed, 1 insertion(+)
diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf
index 2b8f9a29..05779c47 100644
--- a/src/conf/cards/USB-Audio.conf
+++ b/src/conf/cards/USB-Audio.conf
@@ -38,6 +38,7 @@ USB-Audio.pcm.surround40_type {
USB-Audio.pcm.iec958_device {
# "NoiseBlaster 3000" 42
"USB Sound Blaster HD" 1
+ "Xonar U7" 1
# The below don't have digital in/out, so prevent them from being opened.
"Andrea PureAudio USB-SA Headset" 999
--
2.13.6
From 4740dd97bf54d67f0edef80e1b9e86c04c8ed402 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 23 May 2018 10:33:40 +0200
Subject: [PATCH 25/26] pcm: add missing flags initialization for the fallback
control data
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
src/pcm/pcm_hw.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c
index a728b23b..59a24200 100644
--- a/src/pcm/pcm_hw.c
+++ b/src/pcm/pcm_hw.c
@@ -1018,7 +1018,7 @@ static int map_status_and_control_data(snd_pcm_t *pcm, bool force_fallback)
snd_pcm_set_appl_ptr(pcm, &hw->mmap_control->appl_ptr, hw->fd,
SNDRV_PCM_MMAP_OFFSET_CONTROL);
if (hw->mmap_control_fallbacked) {
- unsigned int flags;
+ unsigned int flags = 0;
/* read appl_ptr and avail_min from kernel when device opened
* with SND_PCM_APPEND flag
*/
--
2.13.6
From ad5aea89226f131f27790c98c4638e4596060f81 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 23 May 2018 10:36:17 +0200
Subject: [PATCH 26/26] pcm ioplug: fix some coverity issues (switch, missing
unlock in snd_pcm_ioplug_drain())
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
src/pcm/pcm_ioplug.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c
index db64853b..4d44ae22 100644
--- a/src/pcm/pcm_ioplug.c
+++ b/src/pcm/pcm_ioplug.c
@@ -533,6 +533,7 @@ static int snd_pcm_ioplug_drain(snd_pcm_t *pcm)
case SND_PCM_STATE_OPEN:
case SND_PCM_STATE_DISCONNECTED:
case SND_PCM_STATE_SUSPENDED:
+ snd_pcm_unlock(pcm);
return -EBADFD;
case SND_PCM_STATE_PREPARED:
if (pcm->stream == SND_PCM_STREAM_PLAYBACK) {
@@ -545,6 +546,8 @@ static int snd_pcm_ioplug_drain(snd_pcm_t *pcm)
case SND_PCM_STATE_RUNNING:
io->data->state = SND_PCM_STATE_DRAINING;
break;
+ default:
+ break;
}
if (io->data->state == SND_PCM_STATE_DRAINING) {
--
2.13.6