Jaroslav Kysela [Fri, 13 Nov 2009 17:41:52 +0000 (18:41 +0100)]
ALSA: hda - add beep_mode module parameter
The beep_mode parameter for snd-hda-intel module allows to choose among
different digital beep device registation to the input layer.
0 = do not register to the input layer
1 = register to the input layer all time
2 = use "Beep Switch" control exported to user space mixer applications
Also, introduce CONFIG_SND_HDA_INPUT_BEEP_MODE for default value.
Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jaroslav Kysela [Wed, 21 Oct 2009 12:48:23 +0000 (14:48 +0200)]
ALSA: hda_intel: Digital PC Beep - change behaviour for input layer
Original implementation was keeping registered input device for SND_BEEP
and SND_TONE events all time. This patch changes this behaviour:
If digital PC Beep is turned off using universal control switch,
the input device is unregistered.
Explanation: The kd_mksound() send SND_BEEP and SND_TONE only to last
registered device acceping those events. It means that the HDA Intel
audio driver blocks also the internal PC Speaker device (pcspkr.c
driver) even if the HDA Beep is muted. The user can easy disable
all beeps using 'setterm -blength 0' or 'xset b off' command.
Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Carpenter [Mon, 16 Nov 2009 09:07:17 +0000 (11:07 +0200)]
ALSA: ice1724 - make some bitfields unsigned
This is a clean up and doesn't change the behavior.
Bit fields should always be unsigned. Otherwise pm_suspend_enabled will
be -1 when you want it to be 1. The other bad thing is that the sparse
checker will complain 36 times if they aren't unsigned.
The other bitfields in that struct are unsigned already.
Signed-off-by: Dan Carpenter <error27@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: ice1724 - Patch for suspend/resume for ESI Juli@
Add proper suspend/resume code for Juli@ cards. Based on ice1724
suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
Tested on linux-2.6.31.6
sound/pci/hda/patch_via.c: work around gcc-4.0.2 ICE
sound/pci/hda/patch_via.c: In function 'via_hp_bind_automute':
sound/pci/hda/patch_via.c:2074: internal compiler error: in do_SUBST, at combine.c:462
Please submit a full bug report,
with preprocessed source if appropriate.
See <URL:http://gcc.gnu.org/bugs.html> for instructions.
[added a comment by tiwai]
Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Joonyoung Shim [Thu, 12 Nov 2009 08:14:04 +0000 (17:14 +0900)]
ASoC: Add jack_status_check callback function for GPIO jacks
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Barry Song [Thu, 12 Nov 2009 04:01:47 +0000 (12:01 +0800)]
ASoC: move setting ac97 platformdata earlier than ac97 read/write
While probing, AC97 codec drivers and soc-core generically execute the
following sequence:
snd_soc_new_ac97_codec -> snd_soc_new_pcms -> reset ac-link/read AC97 ID
to detect ->... -> set platform_data to ac97 by soc-core
commit 474828a40f6ddab6e2a3475a19c5c84aa3ec7d60 adds platform_data to
snd_ac97 instance. But ac97 platform data hasn't given to snd_ac97
before actual ac97 operations. Then while ac97_read access platform_data
of snd_ac97 for detecting, NULL pointer oops will fire. That means old
platform_data patch doesn't work in real-life cases.
This patch moves the operation of setting ac97 platform_data earlier
than ac97 reading/writing operations. Then it makes platform_data of
AC97 become practically useful.
Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 10 Nov 2009 16:08:04 +0000 (16:08 +0000)]
ASoC: Add bit clock rate calculator utility functions
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Takashi Iwai [Thu, 12 Nov 2009 08:50:28 +0000 (09:50 +0100)]
ALSA: hda - Don't access invalid substream in proc file
The commit e3303235209c0496b490e10ab131e72a9568c153
"ALSA: hda - proc - show which I/O NID is associated to PCM device"
introduces the access to substream pointer. But, PCMs may have no
substreams in one or both directions, and this results in NULL
dereference. Also, print the first substream number doesn't make
sense.
This patch removes the access to the substream pointer, and reformat
to fit to the standard coding style.
Daniel T Chen [Wed, 11 Nov 2009 19:32:10 +0000 (14:32 -0500)]
ALSA: hda: Use model=mb5 for MacBookPro 5,2
BugLink: https://bugs.launchpad.net/bugs/462098
Until we can look closer at the verbs, let's use ALC885_MB5 for
codec SSID 0x106b4600 to enable playback and capture for MacBookPro
5,2s.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 11 Nov 2009 08:34:25 +0000 (09:34 +0100)]
ALSA: hda - Add power on/off counter
Added the power on/off counter and expose via sysfs files.
The sysfs files, power_on_acct and power_off_acct, are created under
each codec hwdep sysfs directory (e.g. /sys/class/sound/hwC0D0).
The files show the msec length of the codec power-on and power-off,
respectively.
Roel Kluin [Tue, 10 Nov 2009 19:11:55 +0000 (20:11 +0100)]
ALSA: hda - possible read past array alc88[02]_parse_auto_config()
The test of index `i' is after the read - too late - and
unsafe: if snd_hda_get_connections() fails in the last
iteration a read beyond the array is possible.
Clemens Ladisch [Tue, 10 Nov 2009 09:13:30 +0000 (10:13 +0100)]
sound: pcm: record a substream's owner process
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 21 Oct 2009 07:12:26 +0000 (09:12 +0200)]
sound: rawmidi: fix opened substreams count
The substream_opened field is to count the number of opened substreams,
not the number of times that any substreams have been opened.
Furthermore, all substreams being opened does not imply that the next
open would fail, due to the possibility of O_APPEND. With this wrong
check, opening a substream multiple times would succeed only if the
device had more unopened substreams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
If only the output substream of a RawMIDI device has been opened and
if this device is then opened with O_RDWR | O_APPEND and if the
initialization of the input substream fails (either because of low
memory or because the device driver's open callback fails), then the
runtime structure of the already open output substream will be freed
and all following writes through the first handle will cause
snd_rawmidi_write() to use the NULL runtime pointer.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 21 Oct 2009 07:10:16 +0000 (09:10 +0200)]
sound: rawmidi: fix checking of O_APPEND when opening MIDI device
Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 dropped the
check that a substream must already have been opened with O_APPEND to be
able to open it a second time.
This would make it possible for a substream to be switched to append
mode, which would mean that non-atomic writes would fail unexpectedly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 21 Oct 2009 07:09:38 +0000 (09:09 +0200)]
sound: rawmidi: fix double init when opening MIDI device with O_APPEND
Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 moved the
substream initialization code to where it would be executed every time
the substream is opened.
This had the consequence that any further opening would drop and leak
the data in the existing buffer, and that the device driver's open
callback would be called multiple times, unexpectedly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 10 Nov 2009 15:08:45 +0000 (16:08 +0100)]
ALSA: hda - Avoid quirk for HP dc5750
The present quirk for HP dc5750 seems broken and maps the pins wrongly.
Since the auto-parser works well for this device, set the default entry
to use model=auto.
The TX and RX irq handlers are identical. Merge them
Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Wed, 4 Nov 2009 07:58:20 +0000 (09:58 +0200)]
ASoC: TWL4030: Do not modify the APLL_CTL register
APLL_CTL register is configured by the twl4030-codec MFD
driver.
Remove code, which makes changes in the APLL_CTL register,
and replace those with checks against the configured
audio_mclk configuration done in the MFD driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Wed, 4 Nov 2009 07:58:19 +0000 (09:58 +0200)]
MFD: twl4030-codec: APLL_INFREQ handling in the MFD driver
Configure the APLL_INFREQ field in the APLL_CTL register
based on the platform data.
Provide also a function for childs to query the audio_mclk
frequency.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Wed, 4 Nov 2009 07:58:18 +0000 (09:58 +0200)]
OMAP: Configure audio_mclk for twl4030-codec MFD
audio_mclk value is going to be handled by the
twl4030-codec MFD driver, configure the correct
value for boards, which is using the twl4030 audio.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Wed, 4 Nov 2009 07:58:17 +0000 (09:58 +0200)]
MFD: TWL4030: Add audio_mclk to the codec platform data
Add audio_mclk to the platform data struct for the
twl4030-codec MFD driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Graeme Gregory [Mon, 9 Nov 2009 19:02:15 +0000 (19:02 +0000)]
ASoC: omap-mcbsp - add support for upto 16 channels.
This patch increases the number of supported audio channels from 4
to 16 and has been sponsored by Shotspotter Inc. It also fixes a
FSYNC rate calculation bug when McBSP is FSYNC master.
Signed-off-by: Graeme Gregory <gg@slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Drake [Mon, 9 Nov 2009 15:17:24 +0000 (15:17 +0000)]
ALSA: hda - Tweak OLPC XO-1.5 microphone bias
Our contacts at Conexant suggested that we reduce the external
microphone bias to 50% in order to center the input signal with
the DC input range of the codec. This is because the microphone
port is DC coupled for potential use with sensors.
Signed-off-by: Daniel Drake <dsd@laptop.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jarkko Nikula [Mon, 9 Nov 2009 06:44:32 +0000 (08:44 +0200)]
ASoC: Pandora: Pass SRG input clock frequency to the OMAP McBSP DAI
Upcoming change to omap-mcbsp.c require that machine drivers using OMAP
as a DAI master to pass sample rate generator input clock frequency to
the omap-mcbsp.c DAI driver.
Pandora is using 256*Fs output from the TWL4030 codec as an input clock to
the McBSP sample rate generator.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Tested-by: Grazvydas Ignotas <notasas@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Krzysztof Helt [Sun, 8 Nov 2009 10:58:08 +0000 (11:58 +0100)]
ALSA: es18xx: code improvements
1. Set the third argument of the snd_device_new to not NULL, so there is
no warning about bug during chip detection. The third argument is not
used in this driver. It was changed in my previous patch.
2. Remove the fm_port and mpu_port fields from the snd_es18xx structure.
They can be converted to function arguments.
3. Remove the dmaN_size fields from the snd_es18xx structure. These
values are used only in pointer functions and can be easily calculated.
4. Remove the ctrl_lock spinlock which is used only in one read function
which is called once during chip initialization. There are many
writes to the same register and they are not protected on purpose
(see the comment ina the snd_es18xx_config_write()).
5. Use the first part of the text5Sources string table as the text4Soruces
table (they are the same).
6. Merge the same cases for the ES1887 and ES1888 when setting chip's caps.
7. Move the snd_es18xx_reset() to __devinit section.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Grant Likely [Sat, 7 Nov 2009 08:34:43 +0000 (01:34 -0700)]
ASoC/mpc5200: fix enable/disable of AC97 slots
The MPC5200 AC97 driver is disabling the slots when a stop
trigger is received, but not reenabling them if the stream
is started again without processing the hw_params again.
This patch fixes the problem by caching the slot enable bit
settings calculated at hw_params time so that they can be
reapplied every time the start trigger is received.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Grant Likely [Sat, 7 Nov 2009 08:34:31 +0000 (01:34 -0700)]
ASoC/mpc5200: add to_psc_dma_stream() helper
Move the resolving of the psc_dma_stream pointer to a helper function
to reduce duplicate code
Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Grant Likely [Sat, 7 Nov 2009 08:34:18 +0000 (01:34 -0700)]
ASoC/mpc5200: Improve printk debug output for trigger
Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Grant Likely [Sat, 7 Nov 2009 08:34:05 +0000 (01:34 -0700)]
ASoC/mpc5200: get rid of the appl_ptr tracking nonsense
Sound drivers PCM DMA is supposed to free-run until told to stop
by the trigger callback. The current code tries to track appl_ptr,
to avoid stale buffer data getting played out at the end of the
data stream. Unfortunately it also results in race conditions
which can cause the audio to stall.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Grant Likely [Sat, 7 Nov 2009 08:33:53 +0000 (01:33 -0700)]
ASoC/mpc5200: Track DMA position by period number instead of bytes
All DMA blocks are lined up to period boundaries, but the DMA
handling code tracks bytes instead. This patch reworks the code
to track the period index into the DMA buffer instead of the
physical address pointer. Doing so makes the code simpler and
easier to understand.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Sat, 7 Nov 2009 08:49:04 +0000 (09:49 +0100)]
ALSA: hda - Don't initialize CORB/RIRB for single_cmd mode
So far, CORB/RIRB still remains even if the driver is switched to the
single_cmd mode. The specification says that this should be disabled,
but I hoped this isn't the case; indeed most devices worked together with
CORB/RIRB.
However, Poulsbo (US15W) seems problematic with this setup, and it
requires to disable CORB/RIRB when single_cmd is used.
Now this patch disables CORB/RIRB initialization when the single_cmd
mode is used. Also the unsolicited event is disabled because it can't
work without RIRB.
Fix combine_word problem where first octet is not
read properly. The only affected place seems to be the
INPUT_TERMINAL type. Before now, sound controls can be created
with the output terminal's name which is a fallback mechanism
used only for unknown input terminal types. For example,
Line can wrongly appear as Speaker. After the change it
should appear as Line.
The side effect of this change can be that users
can expect the wrong control name in their scripts or
programs while now we return the correct one.
Probably, these defines should use get_unaligned_le16 and
friends.
Takashi Iwai [Fri, 6 Nov 2009 14:47:50 +0000 (15:47 +0100)]
ALSA: hda - Reset pins of IDT/STAC codecs at free
Some laptops cause annoying clicks or noises at shutdown/reboot since
the speaker pin is set still high. Apply the same procedure used for
the suspend to avoid such clicks/noises for freeing the codec, too.
projects / karo-tx-linux.git / log