Takashi Iwai [Thu, 16 Aug 2007 17:32:16 +0000 (19:32 +0200)]
[ALSA] emu10k1 - Fix memory corruption
The number of mixer elements for SPDIF control don't match with the
actual array size (3). This may result in a memory corruption that
overwrites the i2c_capture_source field (ALSA bug#3095).
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Thu, 16 Aug 2007 12:59:45 +0000 (14:59 +0200)]
[ALSA] hda-codec - Add ALC268 acer model
Added model=acer for ALC268 codec support.
The configuration is: headphone = 0x14, speaker = 0x15
needs hp-jack auto-detection. The same routine as alc262-fujitsu model
is used.
Also, added the auto-muting routine for ALC268 model=toshiba.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clemens Ladisch [Thu, 16 Aug 2007 06:44:51 +0000 (08:44 +0200)]
[ALSA] usb-audio: fix parsing of SysEx messages from CME keyboards
When CME keyboards send a SysEx message (e.g. master volume), the USB
packet uses a format different from the standard format. Parsing this
packet according to the specification corrupts the SysEx message itself
and can cause the following MIDI messages to be misinterpreted, too.
This patch adds a workaround for this case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Wed, 15 Aug 2007 20:20:45 +0000 (22:20 +0200)]
[ALSA] hda-codec - Fix Master volume with AD1986A laptop model
Use the bind-control for NID 0x1a and 0x1b as Master volume control
on AD1986 model=laptop as well as model=laptop-eapd. This will fix
the missing output on some devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Wed, 15 Aug 2007 20:18:22 +0000 (22:18 +0200)]
[ALSA] hda-intel - Add flush_scheduled_work() in snd_hda_codec_free()
Added flush_scheduled_work() in snd_hda_codec_free() to make sure that
the all work is gone. Also, optimized the condition to schedule the
delayed work in snd_hda_power_down().
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Tue, 14 Aug 2007 13:18:26 +0000 (15:18 +0200)]
[ALSA] hda-intel - Don't do suspend if already powered down
In the power-saving mode, the suspend is done dynamically at power-down.
So we don't have to call suspend stuff explicitly if it's already
powered down.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Tobin Davis [Mon, 13 Aug 2007 13:50:29 +0000 (15:50 +0200)]
[ALSA] This patch adds more support for Dell systems with Stac9205 codecs.
Tested against a couple of different systems (with different pin
configs), but the others should also work. Also cleaned up some of the
9205 patch code.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] This patch removes memset() from snd_emu10k1_fx8010_info() which apparently
isn't needed there. Upatched code uses:
memset(info, 0, sizeof(info));
where 'info' is a pointer and therefore only first 4 bytes of 'info' gets
cleared on a 32bit machine. Anyway looking at the code zeoring this memory
region isn't needed at all because the snd_emu10k1_fx8010_info() function
initializes all the 'info' fields on its own. So that's why this code works
at all in its original form.
This patch removes this redundant code. Also snd_emu10k1_fx8010_info() can't
fail so lets save some bytes and change its return type to void.
Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Fri, 10 Aug 2007 15:21:45 +0000 (17:21 +0200)]
[ALSA] hda-intel - Add POWER_SAVE option
Added CONFIG_SND_HDA_POWER_SAVE kconfig. It's an experimental option
to achieve an aggressive power-saving. With this option, the driver
will turn on/off the power of each codec and controller chip dynamically
on demand.
The patch introduces a new module option 'power_save'. It specifies
the second of time-out for automatic power-down. As default, it's
10 seconds. Setting 0 means to suppress the power-saving feature.
The codec may have analog-input loopbacks, which are usually represented
by mixer elements such as 'Mic Playback Switch' or 'CD Playback Switch'.
When these are on, we cannot turn off the mixer and the codec chip has
to be kept on. For bookkeeping these states, a new codec-callback is
introduced.
For the bus-controller side, a new callback pm_notify is introduced,
which can be used to turn on/off the contoller appropriately.
Note that this power-saving might cause slight click-noise at
power-on/off. Also, it might take some time to wake up the codec, and
might even drop some tones at the very beginning. This seems to be the
side-effect of turning off the controller chip.
This turn-off of the controller can be disabled by undefining
HDA_POWER_SAVE_RESET_CONTOLLER in hda_intel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Fri, 10 Aug 2007 15:11:07 +0000 (17:11 +0200)]
[ALSA] hda-codec - add snd_hda_codec_stereo() function
Added snd_hda_codec_amp_stereo() function that changes both of stereo
channels with the same mask and value bits. It simplifies most of
amp-handling codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Fri, 10 Aug 2007 15:09:26 +0000 (17:09 +0200)]
[ALSA] hda-codec - optimize resume using caches
So far, the driver looked the table of snd_kcontrol_new used for creating
mixer elements and forces to call each of its put callbacks in PM resume
code. This is too ugly and hackish.
Now, the resume is simplified using the codec amp and command register
caches. The driver simply restores the values that have been written
in the cache table. With this simplification, most codec support codes
don't require any special resume callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the cache for codec command registers.
snd_hda_codec_write_cache() and snd_hda_sequence_write_cache() do
the write operations with caching, which values can be resumed via
snd_hda_codec_resume_cache().
The patch introduces only the framework, and no codec code is using
this cache yet. It'll be implemented in the following patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Krzysztof Helt [Fri, 10 Aug 2007 10:04:42 +0000 (12:04 +0200)]
[ALSA] isa libs Makefiles cleanup
This patch uses the Kconfig parameters SND_AD1848_LIB and
SND_CS4231_LIB instead of mentioning each driver that requires
the ad1848-lib or cs4231-lib separately in the Makefiles.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clemens Ladisch [Fri, 10 Aug 2007 07:39:14 +0000 (09:39 +0200)]
[ALSA] seq_midi_event: prevent running status after system messages
Reset the event type after encoding a system message to prevent any
following data bytes from being interpreted as data for a running status
system message, which is not allowed in MIDI.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clemens Ladisch [Fri, 10 Aug 2007 07:38:36 +0000 (09:38 +0200)]
[ALSA] seq_midi_event: fix encoding of data bytes after end of sysex
Create a new state ST_INVALID for the encoder to prevent data bytes at
the beginning of a stream or after a sysex message being interpreted as
note-off parameters.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Mark Hills [Fri, 10 Aug 2007 06:01:54 +0000 (08:01 +0200)]
[ALSA] This patch is a USB quirk to ensure the Stanton Scratchamp v1 is detected
(bugtrack #2932). The interface is two USB devices in the same physical
box. Note that this is the USB ScratchAmp v1 and not the later v2
(firewire) model.
Signed-off-by: Mark Hills <mark@pogo.org.uk> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Wed, 8 Aug 2007 14:49:08 +0000 (16:49 +0200)]
[ALSA] Support 3-bytes 24bit format in PCM OSS emulation
Add the support of 3-bytes 24bit formats in PCM OSS emulation.
Also removed snd_pcm_build_linear_format() function. It's exported
just for OSS emulation, and now the code was changed without calling
this function.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Wed, 8 Aug 2007 13:50:58 +0000 (15:50 +0200)]
[ALSA] Simplify the format conversion in PCM OSS emulation
Simplify the format conversion code in PCM OSS emulation.
This patch also adds the support of 3bytes 24bit formats with linear
and mulaw, but they are not enabled in pcm_plugin.c yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Russ Cox [Mon, 6 Aug 2007 13:37:58 +0000 (15:37 +0200)]
[ALSA] fix selector unit bug affecting some USB speakerphones
Following the suggestion in this thread:
https://bugs.launchpad.net/ubuntu/+source/alsa-lib/+bug/26683
the correct upper bound on desc[0] is 5 + num_ins not 6 + num_ins,
because the index used later is 5+i, not 6+i.
This change makes my Vosky Chatterbox speakerphone work.
Apparently it also helps with the Minivox MV100.
Jesper Juhl [Mon, 6 Aug 2007 12:05:27 +0000 (14:05 +0200)]
[ALSA] au88x0: mem leak fix in snd_vortex_create()
In sound/pci/au88x0/au88x0.c::snd_vortex_create() :
The Coverity checker found that if we allocate storage for 'chip'
but then leave via the regions_out: label, then we end up leaking
the storage allocated for 'chip'.
I believe simply freeing 'chip' before the 'return err;' line is
all we need to fix this, but please double-check me :)
Takashi Iwai [Thu, 2 Aug 2007 13:51:59 +0000 (15:51 +0200)]
[ALSA] hda-intel - Remove invalid __devinit
Some functions in hda_codec.c are called from patch ops, which are
kept in the codec instance even after initialization. Thus they
shouldn't be marked as __devinit.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Rene Herman [Wed, 1 Aug 2007 21:50:21 +0000 (23:50 +0200)]
[ALSA] add the ESS1879 pnpbios ID to the es18xx driver
As reported by Troy Heidner, the 'Gateway Solo 5150' laptop (for one) has an
onboard ESS1879 that identifies itself through PNPBIOS as just that. He also
confirmed that other than not knowing about it, snd-es18xx drives the chip
fine, so this adds the ID to the driver.
Signed-off-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Timur Tabi [Wed, 1 Aug 2007 10:22:07 +0000 (12:22 +0200)]
[ALSA] CS4270 driver does not compile with I2C disabled
Fix compilation errors with the CS4270 when I2C is not enabled. Updated
some comments to indicate that that stand-alone mode is not fully implemented,
because there is no mechanism for the CS4270 driver and the machine driver to
communicate the values of various input pins.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Timur Tabi [Tue, 31 Jul 2007 16:18:44 +0000 (18:18 +0200)]
[ALSA] ASoC CS4270 codec device driver
This patch adds ALSA SoC support for the Cirrus Logic CS4270 codec. The
following features are suppored:
1) Stand-alone and software mode
2) Software mode via I2C only
3) Master mode, not Slave
4) No power management
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clemens Ladisch [Mon, 30 Jul 2007 06:14:31 +0000 (08:14 +0200)]
[ALSA] check for linked substreams of different cards
It is possible to have linked substreams that belong to different cards
and/or different drivers. This patch changes some drivers to make sure
that they do not incorrectly try to handle substreams of a different
card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] hda-codec - Add a generic bind-control helper
Added callbacks for a generic bind-control of mixer elements.
This can be used for creating a mixer element controlling multiple
widgets at the same time. Two macros, HDA_BIND_VOL() and HDA_BIND_SW(),
are introduced for creating bind-volume and bind-switch, respectively.
It taks the mixer element name and struct hda_bind_ctls pointer, which
contains the real control callbacks in ops field and long array for
private_value of each bound widget.
All widgets have to be the same type (i.e. the same amp capability).
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a hwdep interface for each codec (enabled per kconfig).
This interface can be used for reading/writing HD-audio verbs
and other purposes as future extensions.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
input_free_device()'s comment says:
input_free_device() should only be used if input_register_device() was
not called yet or if it failed. Once device was registered
use input_unregister_device() and memory will be freed once last
refrence to the device is dropped.
[ALSA] hda-codec - Fix the initial mixer state of ALC262 sony-assamd model
Many of ALC262 codes don't call the automute function at the beginning,
which may keep the silence until the HP jack is replugged. Now proper
init_hook is added.
Also, sony-assamd model doesn't handle the widget 0x14 properly, thus
calling automute isn't enough. Now Front switch handles both widgets.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] ca0106: remove extra commands in SPI DAC init sequence
The init sequence set a number of registers more than once to different
values. It's only necessary to set them once to their final values.
It also never actually updated the digital attenuation settings.
[ALSA] ca0106: Add more symbol SPI register names and use them
Add more symbol name for SPI register values. Change the SPI_XXX_BIT defines
from the bit number to a mask. Saves having to write (1<<SPI_XXX_BIT) all the
time to convert to mask. We never end up wanting the bit number.
Use all the symbol names for the SPI DAC init sequence. The sequence is
exactly the same as it was before.
[ALSA] ca0106: power down SPI DAC channels when not in use
For cards with an SPI DAC (SB Live 24-bit / Audigy SE), power down channels
0-2 when not in use. They are powered up on PCM open and down again on PCM
close. Channel 4 (== Front) is not powered down, as it is used for capture
feedback. Powering it down would effectively kill line in pass-through.
The SPDIF output on AD1988 had some problems due to the wrongly routed
analog loopback to SPDIF. This patch fixes the implementation of
'IEC958 Playback Source' mixer to handle the amp bits of mixer widget
0x1d correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Harald Welte [Tue, 24 Jul 2007 10:49:39 +0000 (12:49 +0200)]
[ALSA] s3c24xx-pcm: fix hw_params dma handling
Since the PCM emulation can call multiple times to hw_setup(), but we
can only once allocate/request the DMA channel, we have to handle
this gracefully.
Signed-off-by: Harald Welte <laforge@openmoko.org> Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] ca0106: Add analog mute controls for cards with SPI DAC
Add four mute controls for the analog output channels for cards that use
an SPI DAC, like the SB0570 SB Live! 24-bit / Audigy SE. The Wolfson DAC
doesn't support muting left/right so the controls are mono.
The chip state struct gets a 32-byte array to act as a shadow of the spi
dac registers. Only two registers are used for mute, but more would be
needed for analog gain, de-emphasis, DAC power down, phase inversion, and
other features.
Clemens Ladisch [Mon, 23 Jul 2007 15:38:44 +0000 (17:38 +0200)]
[ALSA] ymfpci: fix volume handling of the 44.1 kHz slot
The existing code for handling the 44.1 slot's volume has two problems:
the volume is not affected by the 'Wave Playback Volume' mixer control,
and the BUF441OUTVOL register, which is used to control the per-
substream volume for this slot, uses a different scale than the gain
fields of the other slots.
This patch makes the BUF441OUTVOL register a shadow of the
NATIVEDACOUTVOL register so that the Wave volume is consistent for all
substreams.
As a consequence of this, the per-substream PCM volume control gets no
longer activated for the substream using this slot. The code for
(de)activating the mixer control is moved from the open/close to the
prepare/trigger_stop callbacks so that it is able to determine the
substream's slot.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] ALSA sound driver for the AT73C213 DAC using Atmel SSC driver
This patch adds support for the AT73C213 DAC using the misc Atmel SSC driver in
I2S mode. The driver also requires a SPI to setup the registers and control
volume.
It has been tested with an AT32AP7000 on the ATSTK1000 development board. The
driver should also work with any Atmel device with an SSC module supported by
the Atmel SSC driver (atmel-ssc).
The atmel-ssc driver is just submitted to the Linux kernel. Please see mail
thread http://lkml.org/lkml/2007/7/16/32
projects / mirror_ubuntu-kernels.git / log