From b7d4de7ff03085fda8310b2983b907166dd40f43 Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Wed, 8 Jul 2009 19:24:26 +0200
Subject: ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free
Check for rtd->params->drcmr != NULL before accessing it.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/arm/pxa2xx-pcm-lib.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 108b643229b..6205f37d547 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
- if (rtd && rtd->params)
+ if (rtd && rtd->params && rtd->params->drcmr)
*rtd->params->drcmr = 0;
snd_pcm_set_runtime_buffer(substream, NULL);
--
cgit v1.2.3-18-g5258
From 9d30937accf2c01e8b0bd59787409a7348cbbcb7 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Fri, 10 Jul 2009 12:27:31 +0200
Subject: ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid
checking
On some IbexPeak systems with ALC889A errors like "azx_get_response
timeout, switching to polling mode: last cmd=0xaf9f000b" are produced,
because non-existent codec #10 is wrongly accessed.
The problem is that snd_hda_get_connections() returns out-of-range result
for NID 0x1c (something like 0xf8f9 or 0xffff).
This patch adds a check to alc880_parse_auto_config() to avoid using
of this out-of-range NIDs. A better fix maybe to improve
snd_hda_get_connections() routine to check for valid NID ranges if
NIDs are expected as result.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 6 ++++++
1 file changed, 6 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bbb9b42e260..7e99763ca52 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
&dig_nid, 1);
if (err < 0)
continue;
+ if (dig_nid > 0x7f) {
+ printk(KERN_ERR "alc880_auto: invalid dig_nid "
+ "connection 0x%x for NID 0x%x\n", dig_nid,
+ spec->autocfg.dig_out_pins[i]);
+ continue;
+ }
if (!i)
spec->multiout.dig_out_nid = dig_nid;
else {
--
cgit v1.2.3-18-g5258
From 864e1e8db436dfd784340d0aef18ea303d157bcb Mon Sep 17 00:00:00 2001
From: Greg Kroah-Hartman <gregkh@suse.de>
Date: Mon, 22 Jun 2009 15:46:03 -0700
Subject: Sound: remove direct access of driver_data
This is the last in-kernel direct usage of driver_data, replace it with
the proper dev_get/set_drvdata() calls.
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
---
sound/soc/codecs/wm8988.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index c05f71803aa..8c0fdf84aac 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -1037,14 +1037,14 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi)
codec->control_data = spi;
codec->dev = &spi->dev;
- spi->dev.driver_data = wm8988;
+ dev_set_drvdata(&spi->dev, wm8988);
return wm8988_register(wm8988);
}
static int __devexit wm8988_spi_remove(struct spi_device *spi)
{
- struct wm8988_priv *wm8988 = spi->dev.driver_data;
+ struct wm8988_priv *wm8988 = dev_get_drvdata(&spi->dev);
wm8988_unregister(wm8988);
--
cgit v1.2.3-18-g5258
From 8886f33f25083a47d5fa24ad7b57bb708c5c5403 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 13 Jul 2009 13:21:58 +0200
Subject: sound: usb-audio: add workaround for Blue Microphones devices
Blue Microphones USB devices have an alternate setting that sends two
channels of data to the computer. Unfortunately, the descriptors of
that altsetting have a wrong channel setting, which means that any
recorded data from such a device has twice the sample rate from what
would be expected.
This patch adds a workaround to ignore that altsetting. Since these
devices have only one actual channel, no data is lost.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/usbaudio.c | 14 +++++++++++++-
1 file changed, 13 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c7b902358b7..44b9cdc8a83 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2661,7 +2661,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
struct usb_interface_descriptor *altsd;
int i, altno, err, stream;
int format;
- struct audioformat *fp;
+ struct audioformat *fp = NULL;
unsigned char *fmt, *csep;
int num;
@@ -2734,6 +2734,18 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
continue;
}
+ /*
+ * Blue Microphones workaround: The last altsetting is identical
+ * with the previous one, except for a larger packet size, but
+ * is actually a mislabeled two-channel setting; ignore it.
+ */
+ if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
+ fp && fp->altsetting == 1 && fp->channels == 1 &&
+ fp->format == SNDRV_PCM_FORMAT_S16_LE &&
+ le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
+ fp->maxpacksize * 2)
+ continue;
+
csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
/* Creamware Noah has this descriptor after the 2nd endpoint */
if (!csep && altsd->bNumEndpoints >= 2)
--
cgit v1.2.3-18-g5258
From cb65c8732a50f8a145d36dbdac026a1789ad1587 Mon Sep 17 00:00:00 2001
From: Jaswinder Singh Rajput <jaswinder@kernel.org>
Date: Wed, 15 Jul 2009 16:45:40 +0530
Subject: ALSA: riptide - proper handling of pci_register_driver for joystick
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
We need to check returning error for pci_register_driver(&joystick_driver)
On failure, we should unregister formerly registered audio drivers
This also fixed the compiler warning :
CC [M] sound/pci/riptide/riptide.o
sound/pci/riptide/riptide.c: In function ‘alsa_card_riptide_init’:
sound/pci/riptide/riptide.c:2200: warning: ignoring return value of ‘__pci_register_driver’, declared with attribute warn_unused_result
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/riptide/riptide.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 235a71e5ac8..b5ca02e2038 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2197,9 +2197,12 @@ static int __init alsa_card_riptide_init(void)
if (err < 0)
return err;
#if defined(SUPPORT_JOYSTICK)
- pci_register_driver(&joystick_driver);
+ err = pci_register_driver(&joystick_driver);
+ /* On failure unregister formerly registered audio driver */
+ if (err < 0)
+ pci_unregister_driver(&driver);
#endif
- return 0;
+ return err;
}
static void __exit alsa_card_riptide_exit(void)
--
cgit v1.2.3-18-g5258
From 2e9bf247066a293ebcd4672ddd487808ab5f2d1b Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Sat, 18 Jul 2009 11:48:19 +0200
Subject: ALSA: hda_codec: Check for invalid zero connections
To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_codec.c | 6 ++++++
1 file changed, 6 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d255de6be..88480c0c58a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
AC_VERB_GET_CONNECT_LIST, i);
range_val = !!(parm & (1 << (shift-1))); /* ranges */
val = parm & mask;
+ if (val == 0) {
+ snd_printk(KERN_WARNING "hda_codec: "
+ "invalid CONNECT_LIST verb %x[%i]:%x\n",
+ nid, i, parm);
+ return 0;
+ }
parm >>= shift;
if (range_val) {
/* ranges between the previous and this one */
--
cgit v1.2.3-18-g5258
From fcb2954b9621dfeaca92f6a11dac69cfdfaa6705 Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Sat, 18 Jul 2009 17:26:14 +0200
Subject: ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
If spin_lock_irqsave is called twice in a row with the same second
argument, the interrupt state at the point of the second call overwrites
the value saved by the first call. Indeed, the second call does not need
to save the interrupt state, so it is changed to a simple spin_lock.
The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
expression lock1,lock2;
expression flags;
@@
*spin_lock_irqsave(lock1,flags)
... when != flags
*spin_lock_irqsave(lock2,flags)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/isa/gus/gus_pcm.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index edb11eefdfe..2dcf45bf729 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
continue;
/* load real volume - better precision */
- spin_lock_irqsave(&gus->reg_lock, flags);
+ spin_lock(&gus->reg_lock);
snd_gf1_select_voice(gus, pvoice->number);
snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
pcmp->final_volume = 1;
- spin_unlock_irqrestore(&gus->reg_lock, flags);
+ spin_unlock(&gus->reg_lock);
}
spin_unlock_irqrestore(&gus->voice_alloc, flags);
return change;
--
cgit v1.2.3-18-g5258
From f96e0808212ca284cc9398d7cd3f573786c1d890 Mon Sep 17 00:00:00 2001
From: Jaswinder Singh Rajput <jaswinder@kernel.org>
Date: Sun, 19 Jul 2009 21:58:34 +0530
Subject: ALSA: OSS sequencer should be initialized after
snd_seq_system_client_init
When build SND_SEQUENCER in kernel then OSS sequencer(alsa_seq_oss_init)
is initialized before System (snd_seq_system_client_init) which leads to
memory leak :
unreferenced object 0xf6b0e680 (size 256):
comm "swapper", pid 1, jiffies 4294670753
backtrace:
[<c108ac5c>] create_object+0x135/0x204
[<c108adfe>] kmemleak_alloc+0x26/0x4c
[<c1087de2>] kmem_cache_alloc+0x72/0xff
[<c126d2ac>] seq_create_client1+0x22/0x160
[<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
[<c1485a05>] snd_seq_oss_create_client+0x86/0x142
[<c1485920>] alsa_seq_oss_init+0xf6/0x155
[<c1001059>] do_one_initcall+0x4f/0x111
[<c14655be>] kernel_init+0x115/0x166
[<c10032af>] kernel_thread_helper+0x7/0x10
[<ffffffff>] 0xffffffff
unreferenced object 0xf688a580 (size 64):
comm "swapper", pid 1, jiffies 4294670753
backtrace:
[<c108ac5c>] create_object+0x135/0x204
[<c108adfe>] kmemleak_alloc+0x26/0x4c
[<c1087de2>] kmem_cache_alloc+0x72/0xff
[<c126f964>] snd_seq_pool_new+0x1c/0xb8
[<c126d311>] seq_create_client1+0x87/0x160
[<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
[<c1485a05>] snd_seq_oss_create_client+0x86/0x142
[<c1485920>] alsa_seq_oss_init+0xf6/0x155
[<c1001059>] do_one_initcall+0x4f/0x111
[<c14655be>] kernel_init+0x115/0x166
[<c10032af>] kernel_thread_helper+0x7/0x10
[<ffffffff>] 0xffffffff
unreferenced object 0xf6b0e480 (size 256):
comm "swapper", pid 1, jiffies 4294670754
backtrace:
[<c108ac5c>] create_object+0x135/0x204
[<c108adfe>] kmemleak_alloc+0x26/0x4c
[<c1087de2>] kmem_cache_alloc+0x72/0xff
[<c12725a0>] snd_seq_create_port+0x51/0x21c
[<c126de50>] snd_seq_ioctl_create_port+0x57/0x13c
[<c126d07a>] snd_seq_do_ioctl+0x4a/0x69
[<c126d0de>] snd_seq_kernel_client_ctl+0x33/0x49
[<c1485a74>] snd_seq_oss_create_client+0xf5/0x142
[<c1485920>] alsa_seq_oss_init+0xf6/0x155
[<c1001059>] do_one_initcall+0x4f/0x111
[<c14655be>] kernel_init+0x115/0x166
[<c10032af>] kernel_thread_helper+0x7/0x10
[<ffffffff>] 0xffffffff
The correct order should be :
System (snd_seq_system_client_init) should be initialized before
OSS sequencer(alsa_seq_oss_init) which is equivalent to :
1. insmod sound/core/seq/snd-seq-device.ko
2. insmod sound/core/seq/snd-seq.ko
3. insmod sound/core/seq/snd-seq-midi-event.ko
4. insmod sound/core/seq/oss/snd-seq-oss.ko
Including sound/core/seq/oss/Makefile after other seq modules
fixes the ordering and memory leak.
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/seq/Makefile | 7 ++-----
1 file changed, 2 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 1bcb360330e..941f64a853e 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -3,10 +3,6 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
-ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
- obj-$(CONFIG_SND_SEQUENCER) += oss/
-endif
-
snd-seq-device-objs := seq_device.o
snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
seq_fifo.o seq_prioq.o seq_timer.o \
@@ -19,7 +15,8 @@ snd-seq-virmidi-objs := seq_virmidi.o
obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+ obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+ obj-$(CONFIG_SND_SEQUENCER) += oss/
endif
obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
--
cgit v1.2.3-18-g5258
From 42b95f0c6b524b5a670dd17533a3522db368f600 Mon Sep 17 00:00:00 2001
From: Hao Song <baritono.tux@gmail.com>
Date: Mon, 20 Jul 2009 15:01:16 +0800
Subject: ALSA: hda - Add quirk for Gateway T6834c laptop
Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.
Signed-off-by: Hao Song <baritono.tux@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a18c1..d9b89ba2b65 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2378,6 +2378,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
"Dell Vostro 1500", STAC_9205_DELL_M42),
/* Gateway */
+ SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
{} /* terminator */
};
--
cgit v1.2.3-18-g5258
From b04add956616b6d89ff21da749b46ad2bd58ef32 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Jul 2009 08:01:36 +0200
Subject: ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS. Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.
This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides. Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
sound/pci/hda/patch_sigmatel.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d9b89ba2b65..da7f9f65c04 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5855,6 +5855,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
};
static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+ SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
+ "Sony VAIO F/S", STAC_9872_VAIO),
{} /* terminator */
};
@@ -5867,6 +5869,8 @@ static int patch_stac9872(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+ spec->pin_nids = stac9872_pin_nids;
spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
stac9872_models,
@@ -5878,8 +5882,6 @@ static int patch_stac9872(struct hda_codec *codec)
stac92xx_set_config_regs(codec,
stac9872_brd_tbl[spec->board_config]);
- spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
- spec->pin_nids = stac9872_pin_nids;
spec->multiout.dac_nids = spec->dac_nids;
spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
spec->adc_nids = stac9872_adc_nids;
--
cgit v1.2.3-18-g5258
From 34fdeb2d07102e07ecafe79dec170bd6733f2e56 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Jul 2009 15:42:51 +0200
Subject: ALSA: ca0106 - Fix the max capture buffer size
The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.
This patch restricts the max buffer size less than that to make life
a bit easier.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
sound/pci/ca0106/ca0106_main.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 57b992a5c05..700f15ea16d 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
.rate_max = 192000,
.channels_min = 2,
.channels_max = 2,
- .buffer_bytes_max = ((65536 - 64) * 8),
+ .buffer_bytes_max = 65536 - 128,
.period_bytes_min = 64,
- .period_bytes_max = (65536 - 64),
+ .period_bytes_max = 32768 - 64,
.periods_min = 2,
.periods_max = 2,
.fifo_size = 0,
--
cgit v1.2.3-18-g5258
From 55fe27f7e2c9d24ce870136bd99ae67b020122d1 Mon Sep 17 00:00:00 2001
From: Frank Roth <frashman@freenet.de>
Date: Mon, 20 Jul 2009 17:00:14 +0200
Subject: ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong.
I double checked it with connector colors and creative soundblaster
windows drivers.
So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.
Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/ctxfi/ctdaio.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 082e35c08c0..deb6cfa7360 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
[LINEO1] = {.left = 0x40, .right = 0x41},
- [LINEO2] = {.left = 0x70, .right = 0x71},
+ [LINEO2] = {.left = 0x60, .right = 0x61},
[LINEO3] = {.left = 0x50, .right = 0x51},
- [LINEO4] = {.left = 0x60, .right = 0x61},
+ [LINEO4] = {.left = 0x70, .right = 0x71},
[LINEIM] = {.left = 0x45, .right = 0xc5},
[SPDIFOO] = {.left = 0x00, .right = 0x01},
[SPDIFIO] = {.left = 0x05, .right = 0x85},
--
cgit v1.2.3-18-g5258
From 79452f0a28aa5a40522c487b42a5fc423647ad98 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 12:51:51 +0200
Subject: ALSA: pcm - Fix regressions with VMware
VMware tends to report PCM positions and period updates at utterly
wrong timing. This screws up the recent PCM core code that tries
to correct the position based on the irq timing.
Now, when a backward irq position is detected, skip the update
instead of rebasing. (This is almost the old behavior before
2.6.30.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_lib.c | 11 ++++++++++-
1 file changed, 10 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 333e4dd2945..3b673e2f991 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -244,18 +244,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
delta = new_hw_ptr - hw_ptr_interrupt;
}
if (delta < 0) {
- delta += runtime->buffer_size;
+ if (runtime->periods == 1)
+ delta += runtime->buffer_size;
if (delta < 0) {
hw_ptr_error(substream,
"Unexpected hw_pointer value "
"(stream=%i, pos=%ld, intr_ptr=%ld)\n",
substream->stream, (long)pos,
(long)hw_ptr_interrupt);
+#if 1
+ /* simply skipping the hwptr update seems more
+ * robust in some cases, e.g. on VMware with
+ * inaccurate timer source
+ */
+ return 0; /* skip this update */
+#else
/* rebase to interrupt position */
hw_base = new_hw_ptr = hw_ptr_interrupt;
/* align hw_base to buffer_size */
hw_base -= hw_base % runtime->buffer_size;
delta = 0;
+#endif
} else {
hw_base += runtime->buffer_size;
if (hw_base >= runtime->boundary)
--
cgit v1.2.3-18-g5258
From 2cf313ee75ddf6220b5d623b749b1bb79458307f Mon Sep 17 00:00:00 2001
From: Alexey Fisher <bug-track@fisher-privat.net>
Date: Wed, 22 Jul 2009 14:57:54 +0200
Subject: ALSA: usb-audio - Volume control quirk for QuickCam E 3500
- E3500 report cval->max more than it actually can handel, so if you
set 95% capture level it will be silently muted.
- Betwen cval->min and cval-max(real) is 2940 control units,
but real are only 7 with cval->res = 384.
- Alsa can't handel less than 10 controls, so make it more
and set cval->res = 192.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/usbmixer.c | 25 ++++++++++++++++++++-----
1 file changed, 20 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 4bd3a7a0edc..ec9cdf98692 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
break;
}
- /* quirk for UDA1321/N101 */
- /* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
- /* is not very clear from datasheets */
- /* I hope that the min value is -15360 for newer firmware --jk */
+ /* volume control quirks */
switch (state->chip->usb_id) {
case USB_ID(0x0471, 0x0101):
case USB_ID(0x0471, 0x0104):
case USB_ID(0x0471, 0x0105):
case USB_ID(0x0672, 0x1041):
+ /* quirk for UDA1321/N101.
+ * note that detection between firmware 2.1.1.7 (N101)
+ * and later 2.1.1.21 is not very clear from datasheets.
+ * I hope that the min value is -15360 for newer firmware --jk
+ */
if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
cval->min == -15616) {
- snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n");
+ snd_printk(KERN_INFO
+ "set volume quirk for UDA1321/N101 chip\n");
cval->max = -256;
}
+ break;
+
+ case USB_ID(0x046d, 0x09a4):
+ if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+ snd_printk(KERN_INFO
+ "set volume quirk for QuickCam E3500\n");
+ cval->min = 6080;
+ cval->max = 8768;
+ cval->res = 192;
+ }
+ break;
+
}
snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
--
cgit v1.2.3-18-g5258
From 86de7416600e93835eeacee379aea939b6a0917a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 16:02:46 +0200
Subject: ALSA: hda - Use snprintf() to be safer
Use snprint() for creating the jack name string instead of sprintf()
in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index da7f9f65c04..512f3b9b9a4 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4066,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
jack->nid = nid;
jack->type = type;
- sprintf(name, "%s at %s %s Jack",
+ snprintf(name, sizeof(name), "%s at %s %s Jack",
snd_hda_get_jack_type(def_conf),
snd_hda_get_jack_connectivity(def_conf),
snd_hda_get_jack_location(def_conf));
--
cgit v1.2.3-18-g5258
From 68110661e86868cd107955ec7c077e1f34519f78 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 17:05:15 +0200
Subject: ALSA: ctxfi - Fix uninitialized error checks
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/ctxfi/ctamixer.c | 14 ++++++--------
sound/pci/ctxfi/ctsrc.c | 7 +++----
2 files changed, 9 insertions(+), 12 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a1db51b3ead..a7f4a671f7b 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,
/* Allocate mem for amixer resource */
amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
- if (NULL == amixer) {
- err = -ENOMEM;
- return err;
- }
+ if (!amixer)
+ return -ENOMEM;
/* Check whether there are sufficient
* amixer resources to meet request. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,
/* Allocate mem for sum resource */
sum = kzalloc(sizeof(*sum), GFP_KERNEL);
- if (NULL == sum) {
- err = -ENOMEM;
- return err;
- }
+ if (!sum)
+ return -ENOMEM;
/* Check whether there are sufficient sum resources to meet request. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index e1c145d8b70..df43a5cd393 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,
/* Allocate mem for SRCIMP resource */
srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
- if (NULL == srcimp) {
- err = -ENOMEM;
- return err;
- }
+ if (!srcimp)
+ return -ENOMEM;
/* Check whether there are sufficient SRCIMP resources. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
--
cgit v1.2.3-18-g5258
From 4012ade9338c05428162e85cc9b149dcadf1ce85 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 18:15:10 +0200
Subject: ALSA: hda - Restore GPIO1 properly at resume with AD1984A
The commit 099db17e66294b02814dee01c81d9abbbeece93e introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.
The fix is simple, use the cached write for storing GPIO data.
Reference: Novell bnc#522764
https://bugzilla.novell.com/show_bug.cgi?id=522764
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_analog.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index f795ee588cc..e8e6a43865c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3742,7 +3742,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
int mute = (!ucontrol->value.integer.value[0] &&
!ucontrol->value.integer.value[1]);
/* toggle GPIO1 according to the mute state */
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
mute ? 0x02 : 0x0);
return ret;
}
--
cgit v1.2.3-18-g5258
From 06c71282a90470184a78f7f0ab0f7ce0fc1f69c8 Mon Sep 17 00:00:00 2001
From: Chaithrika U S <chaithrika@ti.com>
Date: Wed, 22 Jul 2009 07:45:04 -0400
Subject: ASoC: tlv320aic3x: Enable PLL when not bypassed
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
sound/soc/codecs/tlv320aic3x.c | 11 ++++++++++-
1 file changed, 10 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ab099f48248..cb0d1bf34b5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 pll_d = 1;
+ u8 reg;
/* select data word length */
data =
@@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
pll_q &= 0xf;
aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
- } else
+ /* disable PLL if it is bypassed */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE);
+
+ } else {
aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+ /* enable PLL when it is used */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE);
+ }
/* Route Left DAC to left channel input and
* right DAC to right channel input */
--
cgit v1.2.3-18-g5258
From cedb8118e8cef21a2b73fd9cb70660ac19124c16 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 23 Jul 2009 11:04:13 +0200
Subject: ALSA: pcm - Add logging of hwptr updates and interrupt updates
Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.
For example,
# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_lib.c | 25 +++++++++++++++++++++++++
1 file changed, 25 insertions(+)
(limited to 'sound')
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 3b673e2f991..065eaf0a386 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
xrun(substream);
return -EPIPE;
}
+ if (xrun_debug(substream, 8)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, "
+ "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+ name, pos,
+ (int)runtime->period_size,
+ (int)runtime->buffer_size,
+ (long)old_hw_ptr,
+ (long)runtime->hw_ptr_base,
+ (long)runtime->hw_ptr_interrupt);
+ }
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;
hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
@@ -353,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
xrun(substream);
return -EPIPE;
}
+ if (xrun_debug(substream, 16)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, "
+ "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+ name, pos,
+ (int)runtime->period_size,
+ (int)runtime->buffer_size,
+ (long)old_hw_ptr,
+ (long)runtime->hw_ptr_base,
+ (long)runtime->hw_ptr_interrupt);
+ }
+
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;
--
cgit v1.2.3-18-g5258
From 89350640439e0160056de26995d52deb18202b3e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 23 Jul 2009 14:28:37 +0200
Subject: ALSA: pcm - Fix warnings in debug loggings
Add proper cast.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_lib.c | 24 ++++++++++++------------
1 file changed, 12 insertions(+), 12 deletions(-)
(limited to 'sound')
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 065eaf0a386..d315f72949f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -238,12 +238,12 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
pcm_debug_name(substream, name, sizeof(name));
snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, "
"hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
- name, pos,
- (int)runtime->period_size,
- (int)runtime->buffer_size,
- (long)old_hw_ptr,
- (long)runtime->hw_ptr_base,
- (long)runtime->hw_ptr_interrupt);
+ name, (unsigned int)pos,
+ (unsigned int)runtime->period_size,
+ (unsigned int)runtime->buffer_size,
+ (unsigned long)old_hw_ptr,
+ (unsigned long)runtime->hw_ptr_base,
+ (unsigned long)runtime->hw_ptr_interrupt);
}
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;
@@ -370,12 +370,12 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
pcm_debug_name(substream, name, sizeof(name));
snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, "
"hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
- name, pos,
- (int)runtime->period_size,
- (int)runtime->buffer_size,
- (long)old_hw_ptr,
- (long)runtime->hw_ptr_base,
- (long)runtime->hw_ptr_interrupt);
+ name, (unsigned int)pos,
+ (unsigned int)runtime->period_size,
+ (unsigned int)runtime->buffer_size,
+ (unsigned long)old_hw_ptr,
+ (unsigned long)runtime->hw_ptr_base,
+ (unsigned long)runtime->hw_ptr_interrupt);
}
hw_base = runtime->hw_ptr_base;
--
cgit v1.2.3-18-g5258
From 947ca210f1df7656e19890832cb71fc3bdd88707 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 23 Jul 2009 16:21:08 +0200
Subject: ALSA: pcm - Fix hwptr buffer-size overlap bug
The fix 79452f0a28aa5a40522c487b42a5fc423647ad98 introduced another
bug due to the missing offset for the overlapped hwptr.
When the hwptr goes back to zero, the delta value has to be corrected
with the buffer size. Otherwise this causes looping sounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_lib.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index d315f72949f..72cfd47af6b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -256,7 +256,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
delta = new_hw_ptr - hw_ptr_interrupt;
}
if (delta < 0) {
- if (runtime->periods == 1)
+ if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr)
delta += runtime->buffer_size;
if (delta < 0) {
hw_ptr_error(substream,
--
cgit v1.2.3-18-g5258
From b30c4947735f9d76da3d194923efd38ed18ad651 Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Wed, 22 Jul 2009 14:13:35 +0200
Subject: ALSA: snd_usb_caiaq: add support for Audio2DJ
This adds support for Native Instrument's freshly announced Audio2DJ
sound device hardware. Version number bumped to 1.3.19.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/Kconfig | 1 +
sound/usb/caiaq/audio.c | 1 +
sound/usb/caiaq/device.c | 8 +++++++-
sound/usb/caiaq/device.h | 1 +
4 files changed, 10 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 523aec188cc..73525c048e7 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -48,6 +48,7 @@ config SND_USB_CAIAQ
* Native Instruments Kore Controller
* Native Instruments Kore Controller 2
* Native Instruments Audio Kontrol 1
+ * Native Instruments Audio 2 DJ
* Native Instruments Audio 4 DJ
* Native Instruments Audio 8 DJ
* Native Instruments Guitar Rig Session I/O
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 8f9b60c5d74..121af0644fd 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
dev->samplerates |= SNDRV_PCM_RATE_192000;
/* fall thru */
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
dev->samplerates |= SNDRV_PCM_RATE_88200;
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index de38108f0b2..83e6c1312d4 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,13 +35,14 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
"{Native Instruments, Kore Controller},"
"{Native Instruments, Kore Controller 2},"
"{Native Instruments, Audio Kontrol 1},"
+ "{Native Instruments, Audio 2 DJ},"
"{Native Instruments, Audio 4 DJ},"
"{Native Instruments, Audio 8 DJ},"
"{Native Instruments, Session I/O},"
@@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = {
.idVendor = USB_VID_NATIVEINSTRUMENTS,
.idProduct = USB_PID_AUDIO4DJ
},
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_AUDIO2DJ
+ },
{ /* terminator */ }
};
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index ece73514854..44e3edf88be 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -10,6 +10,7 @@
#define USB_PID_KORECONTROLLER 0x4711
#define USB_PID_KORECONTROLLER2 0x4712
#define USB_PID_AK1 0x0815
+#define USB_PID_AUDIO2DJ 0x041c
#define USB_PID_AUDIO4DJ 0x0839
#define USB_PID_AUDIO8DJ 0x1978
#define USB_PID_SESSIONIO 0x1915
--
cgit v1.2.3-18-g5258
From 8de56b7deb2534a586839eda52843c1dae680dc5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 24 Jul 2009 16:51:47 +0200
Subject: ALSA: hda - Fix mute control with some ALC262 models
The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted. This patch fixes
the issue.
Reference: Novell bnc#404873
https://bugzilla.novell.com/show_bug.cgi?id=404873
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
sound/pci/hda/patch_realtek.c | 33 ++++++++++++++++-----------------
1 file changed, 16 insertions(+), 17 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7e99763ca52..8c8b273116f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10631,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
alc262_lenovo_3000_automute(codec, 1);
}
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx, long *valp)
+{
+ int i, change = 0;
+
+ for (i = 0; i < 2; i++, valp++)
+ change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+ return change;
+}
+
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -10639,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+ change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_fujitsu_automute(codec, 0);
return change;
@@ -10680,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_lenovo_3000_automute(codec, 0);
return change;
@@ -11854,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
if (change)
alc268_acer_automute(codec, 0);
return change;
--
cgit v1.2.3-18-g5258
From 626f5cefc60b281a00db1402b82deff82080c70a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jul 2009 00:54:39 +0200
Subject: ALSA: hda - Add quirk for Dell Studio 1555
Added a quirk entry for Dell Studio 1555.
Reference: Novell bnc#525244
https://bugzilla.novell.com/show_bug.cgi?id=525244
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_sigmatel.c | 2 ++
1 file changed, 2 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 512f3b9b9a4..5383d8cff88 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1809,6 +1809,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"Dell Studio 1537", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
"Dell Studio 17", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
+ "Dell Studio 1555", STAC_DELL_M6_DMIC),
{} /* terminator */
};
--
cgit v1.2.3-18-g5258
From 78735cffc2d9ab0dec32f1ba7cbc1d84b45bbf29 Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Wed, 29 Jul 2009 14:35:20 +0200
Subject: ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_eld.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index fcad5ec3177..9446a5abea1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -508,7 +508,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
char name[64];
char *sname;
long long val;
- int n;
+ unsigned int n;
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%s %llx", name, &val) != 2)
@@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
sname++;
n = 10 * n + name[4] - '0';
}
- if (n < 0 || n > 31) /* double the CEA limit */
+ if (n >= ELD_MAX_SAD)
continue;
if (!strcmp(sname, "_coding_type"))
e->sad[n].format = val;
--
cgit v1.2.3-18-g5258
From c45ec06c74512265969aef40b00f320c6afb7a90 Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Wed, 29 Jul 2009 11:46:59 +0200
Subject: sound: aedsp16: Buffer overflow
DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes
including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/oss/aedsp16.c | 9 +++++----
1 file changed, 5 insertions(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c
index 3ee9900ffd7..35b5912cf3f 100644
--- a/sound/oss/aedsp16.c
+++ b/sound/oss/aedsp16.c
@@ -325,8 +325,9 @@
/*
* Size of character arrays that store name and version of sound card
*/
-#define CARDNAMELEN 15 /* Size of the card's name in chars */
-#define CARDVERLEN 2 /* Size of the card's version in chars */
+#define CARDNAMELEN 15 /* Size of the card's name in chars */
+#define CARDVERLEN 10 /* Size of the card's version in chars */
+#define CARDVERDIGITS 2 /* Number of digits in the version */
#if defined(CONFIG_SC6600)
/*
@@ -410,7 +411,7 @@
static int soft_cfg __initdata = 0; /* bitmapped config */
static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */
-static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver:
+static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver:
hi->ver[0] lo->ver[1] */
#if defined(CONFIG_SC6600)
@@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port)
* string is finished.
*/
ver[len++] = ret;
- } while (len < CARDVERLEN);
+ } while (len < CARDVERDIGITS);
sprintf(DSPVersion, "%d.%d", ver[0], ver[1]);
DBG(("success.\n"));
--
cgit v1.2.3-18-g5258
From a987004fbcf163b100d227284999602f83044d7e Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Wed, 29 Jul 2009 12:12:09 +0200
Subject: sound: mpu401.c: Buffer overflow
mpu_synth_info[m].name is a char[30], and the minimum length of the data
written by sprintf is 31 bytes including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/oss/mpu401.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c
index 1b2316f35b1..734b8f9e2f7 100644
--- a/sound/oss/mpu401.c
+++ b/sound/oss/mpu401.c
@@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner)
sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name);
else
sprintf(mpu_synth_info[m].name,
- "MPU-401 %d.%d%c Midi interface #%d",
+ "MPU-401 %d.%d%c MIDI #%d",
(int) (devc->version & 0xf0) >> 4,
devc->version & 0x0f,
revision_char,
--
cgit v1.2.3-18-g5258
From aa563af763373a7e67a7b8fdb427d2a2fcbeab3b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 31 Jul 2009 10:05:11 +0200
Subject: ALSA: hda - Increase PCM stream name buf in patch_realtek.c
The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo". Now the size is doubled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c8b273116f..b95df5d5dcc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,13 +275,13 @@ struct alc_spec {
*/
unsigned int num_init_verbs;
- char stream_name_analog[16]; /* analog PCM stream */
+ char stream_name_analog[32]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
- char stream_name_digital[16]; /* digital PCM stream */
+ char stream_name_digital[32]; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
--
cgit v1.2.3-18-g5258
From f065fabc864f4c98857bf67caa2365e9f8545751 Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Fri, 31 Jul 2009 08:32:03 +0200
Subject: ALSA: sound/aoa: Add kmalloc NULL tests
Check that the result of kzalloc is not NULL before a dereference.
The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
expression *x;
identifier f;
constant char *C;
@@
x = \(kmalloc\|kcalloc\|kzalloc\)(...);
... when != x == NULL
when != x != NULL
when != (x || ...)
(
kfree(x)
|
f(...,C,...,x,...)
|
*f(...,x,...)
|
*x->f
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/aoa/core/gpio-pmf.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 5ca2220eac7..1dd0c28d1fb 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -182,6 +182,10 @@ static int pmf_set_notify(struct gpio_runtime *rt,
if (!old && notify) {
irq_client = kzalloc(sizeof(struct pmf_irq_client),
GFP_KERNEL);
+ if (!irq_client) {
+ err = -ENOMEM;
+ goto out_unlock;
+ }
irq_client->data = notif;
irq_client->handler = pmf_handle_notify_irq;
irq_client->owner = THIS_MODULE;
--
cgit v1.2.3-18-g5258
From ce577e8cf5ddb4216553c9d563a9835d6de70ffa Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 3 Aug 2009 08:23:52 +0200
Subject: ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.
Reference: Novell bnc#526325
https://bugzilla.novell.com/show_bug.cgi?id=526325
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b95df5d5dcc..f6b4cbf1ead 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15157,7 +15157,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
- SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
+ SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
--
cgit v1.2.3-18-g5258
From deadff1665491afce124a8ff83f00f784161f660 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:45:16 +0800
Subject: ALSA: hda: track CIRB/CORB command/response states for each codec
Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.
Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_codec.c | 2 +-
sound/pci/hda/hda_codec.h | 2 +-
sound/pci/hda/hda_intel.c | 76 +++++++++++++++++++++++++++++++++--------------
3 files changed, 56 insertions(+), 24 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88480c0c58a..c7df01b72ca 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -174,7 +174,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
mutex_lock(&bus->cmd_mutex);
err = bus->ops.command(bus, cmd);
if (!err && res)
- *res = bus->ops.get_response(bus);
+ *res = bus->ops.get_response(bus, codec->addr);
mutex_unlock(&bus->cmd_mutex);
snd_hda_power_down(codec);
if (res && *res == -1 && bus->rirb_error) {
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index cad79efaabc..1b75f28ed09 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -568,7 +568,7 @@ struct hda_bus_ops {
/* send a single command */
int (*command)(struct hda_bus *bus, unsigned int cmd);
/* get a response from the last command */
- unsigned int (*get_response)(struct hda_bus *bus);
+ unsigned int (*get_response)(struct hda_bus *bus, unsigned int addr);
/* free the private data */
void (*private_free)(struct hda_bus *);
/* attach a PCM stream */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 77c1b840ca8..19e67a1b602 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -253,7 +253,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
/* STATESTS int mask: S3,SD2,SD1,SD0 */
#define AZX_MAX_CODECS 4
-#define STATESTS_INT_MASK 0x0f
+#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1)
/* SD_CTL bits */
#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */
@@ -361,8 +361,8 @@ struct azx_rb {
dma_addr_t addr; /* physical address of CORB/RIRB buffer */
/* for RIRB */
unsigned short rp, wp; /* read/write pointers */
- int cmds; /* number of pending requests */
- u32 res; /* last read value */
+ int cmds[AZX_MAX_CODECS]; /* number of pending requests */
+ u32 res[AZX_MAX_CODECS]; /* last read value */
};
struct azx {
@@ -531,7 +531,8 @@ static void azx_init_cmd_io(struct azx *chip)
/* RIRB set up */
chip->rirb.addr = chip->rb.addr + 2048;
chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
- chip->rirb.wp = chip->rirb.rp = chip->rirb.cmds = 0;
+ chip->rirb.wp = chip->rirb.rp = 0;
+ memset(chip->rirb.cmds, 0, sizeof(chip->rirb.cmds));
azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr);
azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr));
@@ -552,10 +553,35 @@ static void azx_free_cmd_io(struct azx *chip)
azx_writeb(chip, CORBCTL, 0);
}
+static unsigned int azx_command_addr(u32 cmd)
+{
+ unsigned int addr = cmd >> 28;
+
+ if (addr >= AZX_MAX_CODECS) {
+ snd_BUG();
+ addr = 0;
+ }
+
+ return addr;
+}
+
+static unsigned int azx_response_addr(u32 res)
+{
+ unsigned int addr = res & 0xf;
+
+ if (addr >= AZX_MAX_CODECS) {
+ snd_BUG();
+ addr = 0;
+ }
+
+ return addr;
+}
+
/* send a command */
static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
{
struct azx *chip = bus->private_data;
+ unsigned int addr = azx_command_addr(val);
unsigned int wp;
/* add command to corb */
@@ -564,7 +590,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
wp %= ICH6_MAX_CORB_ENTRIES;
spin_lock_irq(&chip->reg_lock);
- chip->rirb.cmds++;
+ chip->rirb.cmds[addr]++;
chip->corb.buf[wp] = cpu_to_le32(val);
azx_writel(chip, CORBWP, wp);
spin_unlock_irq(&chip->reg_lock);
@@ -578,13 +604,14 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
static void azx_update_rirb(struct azx *chip)
{
unsigned int rp, wp;
+ unsigned int addr;
u32 res, res_ex;
wp = azx_readb(chip, RIRBWP);
if (wp == chip->rirb.wp)
return;
chip->rirb.wp = wp;
-
+
while (chip->rirb.rp != wp) {
chip->rirb.rp++;
chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES;
@@ -592,18 +619,20 @@ static void azx_update_rirb(struct azx *chip)
rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
res = le32_to_cpu(chip->rirb.buf[rp]);
+ addr = azx_response_addr(res_ex);
if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
snd_hda_queue_unsol_event(chip->bus, res, res_ex);
- else if (chip->rirb.cmds) {
- chip->rirb.res = res;
+ else if (chip->rirb.cmds[addr]) {
+ chip->rirb.res[addr] = res;
smp_wmb();
- chip->rirb.cmds--;
+ chip->rirb.cmds[addr]--;
}
}
}
/* receive a response */
-static unsigned int azx_rirb_get_response(struct hda_bus *bus)
+static unsigned int azx_rirb_get_response(struct hda_bus *bus,
+ unsigned int addr)
{
struct azx *chip = bus->private_data;
unsigned long timeout;
@@ -616,10 +645,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
azx_update_rirb(chip);
spin_unlock_irq(&chip->reg_lock);
}
- if (!chip->rirb.cmds) {
+ if (!chip->rirb.cmds[addr]) {
smp_rmb();
bus->rirb_error = 0;
- return chip->rirb.res; /* the last value */
+ return chip->rirb.res[addr]; /* the last value */
}
if (time_after(jiffies, timeout))
break;
@@ -692,7 +721,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
*/
/* receive a response */
-static int azx_single_wait_for_response(struct azx *chip)
+static int azx_single_wait_for_response(struct azx *chip, unsigned int addr)
{
int timeout = 50;
@@ -700,7 +729,7 @@ static int azx_single_wait_for_response(struct azx *chip)
/* check IRV busy bit */
if (azx_readw(chip, IRS) & ICH6_IRS_VALID) {
/* reuse rirb.res as the response return value */
- chip->rirb.res = azx_readl(chip, IR);
+ chip->rirb.res[addr] = azx_readl(chip, IR);
return 0;
}
udelay(1);
@@ -708,7 +737,7 @@ static int azx_single_wait_for_response(struct azx *chip)
if (printk_ratelimit())
snd_printd(SFX "get_response timeout: IRS=0x%x\n",
azx_readw(chip, IRS));
- chip->rirb.res = -1;
+ chip->rirb.res[addr] = -1;
return -EIO;
}
@@ -716,6 +745,7 @@ static int azx_single_wait_for_response(struct azx *chip)
static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
{
struct azx *chip = bus->private_data;
+ unsigned int addr = azx_command_addr(val);
int timeout = 50;
bus->rirb_error = 0;
@@ -728,7 +758,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
azx_writel(chip, IC, val);
azx_writew(chip, IRS, azx_readw(chip, IRS) |
ICH6_IRS_BUSY);
- return azx_single_wait_for_response(chip);
+ return azx_single_wait_for_response(chip, addr);
}
udelay(1);
}
@@ -739,10 +769,11 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
}
/* receive a response */
-static unsigned int azx_single_get_response(struct hda_bus *bus)
+static unsigned int azx_single_get_response(struct hda_bus *bus,
+ unsigned int addr)
{
struct azx *chip = bus->private_data;
- return chip->rirb.res;
+ return chip->rirb.res[addr];
}
/*
@@ -765,13 +796,14 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
}
/* get a response */
-static unsigned int azx_get_response(struct hda_bus *bus)
+static unsigned int azx_get_response(struct hda_bus *bus,
+ unsigned int addr)
{
struct azx *chip = bus->private_data;
if (chip->single_cmd)
- return azx_single_get_response(bus);
+ return azx_single_get_response(bus, addr);
else
- return azx_rirb_get_response(bus);
+ return azx_rirb_get_response(bus, addr);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -1245,7 +1277,7 @@ static int probe_codec(struct azx *chip, int addr)
chip->probing = 1;
azx_send_cmd(chip->bus, cmd);
- res = azx_get_response(chip->bus);
+ res = azx_get_response(chip->bus, addr);
chip->probing = 0;
if (res == -1)
return -EIO;
--
cgit v1.2.3-18-g5258
From a678cdee25a387c8fc3b2754974695412baf1d85 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:46:46 +0800
Subject: ALSA: hda: take cmd_mutex in probe_codec()
Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.
So cmd_mutex would be a safe addition to probe_codec().
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 2 ++
1 file changed, 2 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 19e67a1b602..ddabc827ac4 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1275,10 +1275,12 @@ static int probe_codec(struct azx *chip, int addr)
(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
unsigned int res;
+ mutex_lock(&chip->bus->cmd_mutex);
chip->probing = 1;
azx_send_cmd(chip->bus, cmd);
res = azx_get_response(chip->bus, addr);
chip->probing = 0;
+ mutex_unlock(&chip->bus->cmd_mutex);
if (res == -1)
return -EIO;
snd_printdd(SFX "codec #%d probed OK\n", addr);
--
cgit v1.2.3-18-g5258
From cdb1fbf23181c133fb24f12ad14ccea7dc399599 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:47:41 +0800
Subject: ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ddabc827ac4..b6e6314d006 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -513,6 +513,7 @@ static int azx_alloc_cmd_io(struct azx *chip)
static void azx_init_cmd_io(struct azx *chip)
{
+ spin_lock_irq(&chip->reg_lock);
/* CORB set up */
chip->corb.addr = chip->rb.addr;
chip->corb.buf = (u32 *)chip->rb.area;
@@ -544,13 +545,16 @@ static void azx_init_cmd_io(struct azx *chip)
azx_writew(chip, RINTCNT, 1);
/* enable rirb dma and response irq */
azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN);
+ spin_unlock_irq(&chip->reg_lock);
}
static void azx_free_cmd_io(struct azx *chip)
{
+ spin_lock_irq(&chip->reg_lock);
/* disable ringbuffer DMAs */
azx_writeb(chip, RIRBCTL, 0);
azx_writeb(chip, CORBCTL, 0);
+ spin_unlock_irq(&chip->reg_lock);
}
static unsigned int azx_command_addr(u32 cmd)
--
cgit v1.2.3-18-g5258
From c32649feb4573b31f0a2bfdf35cbe1351256c764 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:48:12 +0800
Subject: ALSA: hda: read CORBWP inside reg_lock
This converts the last CORBWP access outside of reg_lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 4 +++-
1 file changed, 3 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b6e6314d006..df6d9820efa 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -588,15 +588,17 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
unsigned int addr = azx_command_addr(val);
unsigned int wp;
+ spin_lock_irq(&chip->reg_lock);
+
/* add command to corb */
wp = azx_readb(chip, CORBWP);
wp++;
wp %= ICH6_MAX_CORB_ENTRIES;
- spin_lock_irq(&chip->reg_lock);
chip->rirb.cmds[addr]++;
chip->corb.buf[wp] = cpu_to_le32(val);
azx_writel(chip, CORBWP, wp);
+
spin_unlock_irq(&chip->reg_lock);
return 0;
--
cgit v1.2.3-18-g5258
From feb273404f15d86098cb0e81e46330d5c1e22b1b Mon Sep 17 00:00:00 2001
From: Wu Fengguang <f