Takashi Iwai [Fri, 17 Dec 2010 14:23:41 +0000 (15:23 +0100)]
ALSA: hda - Fix conflict of Mic Boot controls
Due to the recent change for multiple mics assignment, we need to handle
the index of each Mic Boost control respectively. Otherwise the driver
gets the control element conflicts, and gives the unsable state.
BugLink: http://launchpad.net/bugs/690530
The SKU value of this machine dictates that auto-mute should be
disabled. Since the SKU value is similar to the PCI SSID, the most
likely conclusion is that the SKU value should be ignored.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
After coming back from suspend, the timeout waiting for Phoenix
chip to complete its power up sequence is not enough, which leaves
the codec cache value for some registers in an outdated state.
Increase the timeout value to wait for the power up sequence
to correclty complete.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: twl6040: Clear interrupt status at boot time
On Phoenix 1.1, the INTID register default value is an invalid
one, causing the interrupt handler to think the phoenix power on
sequence is ready before it actually finishes.
This causes some i2c errors when trying to configure twl.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Peter Ujfalusi [Tue, 14 Dec 2010 11:45:29 +0000 (13:45 +0200)]
ASoC: TWL4030: Fix 24bit support
twl4030 series of codecs supports S32_LE with msbits=24.
Replace the S24_LE with S32_LE format, and add constraint
for 24msbit in case of 32 S32_LE format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Olaya, Margarita [Sat, 11 Dec 2010 03:11:44 +0000 (21:11 -0600)]
ASoC: dapm: Add output driver widget
In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.
Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 10 Dec 2010 18:54:49 +0000 (20:54 +0200)]
ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38
Fix "ASoC: Fix bias power down of non-DAPM codec" for 3.6.37 will cause a
build error when merging into ASoC for-2.6.38. Fix the issue by doing a
change that commit ce6120c "ASoC: Decouple DAPM from CODECs" would do.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 10 Dec 2010 18:53:55 +0000 (20:53 +0200)]
ASoC: Fix bias power down of non-DAPM codec
Currently bias of non-DAPM codec will be powered down (standby/off) whenever
there is a stream stop. This is wrong in simultaneous playback/capture since
the bias is put down immediately after stopping the first stream.
Fix this by using the codec->active count when figuring out the needed bias
level after stream stop.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* wm8350, wm8753 and soc-core use custom code to cancel a delayed
work, execute it immediately if it was pending and wait for its
completion. This is equivalent to flush_delayed_work_sync(). Use
it instead.
Signed-off-by: Tejun Heo <tj@kernel.org> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Peter Ujfalusi [Fri, 10 Dec 2010 11:26:31 +0000 (13:26 +0200)]
ASoC: tlv320dac33: Power down digital parts, when not needed
If the following scenario has been followed:
1. Enable analog bypass
amixer sset 'Analog Left Bypass' on
amixer sset 'Analog Right Bypass' on
2. Start playback
aplay -fdat -d3 /dev/zero
After the playback stopped (3 sec), and the soc timeout (5 sec),
the digital parts of the codec will remain powered up.
This means that the DAI clocks are continue to run, the
oscillator remain operational, etc.
Use the SND_SOC_DAPM_POST_PMD widget to get notification
about the stopped stream, and power down the digital
part of the codec.
If the analog bypass is enabled, than the codec will remain in
BIAS_ON level, and things will work correctly.
In case, if the bypass is disabled, than the codec will
fall to BIAS_STANDBY than to BIAS_OFF level, as it used
to.
The digital part of DAC33 is initialized at every stream start
(DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec)
will have working DAI.
When the codec is coming out from BIAS_OFF, the full power-up
sequence followed by the same DAPM_PRE widget event will power up
the digital part.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mario Becroft [Fri, 3 Dec 2010 22:51:34 +0000 (11:51 +1300)]
ASoC: Optimise WM9081 FLL performance
Tune the FLL gain for optimal performance according to evaluation
results.
Signed-off-by: Mario Becroft <mb@gem.win.co.nz> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Fri, 10 Dec 2010 09:34:26 +0000 (17:34 +0800)]
ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove()
It does not make sense to set clientdata to onyx in onyx_i2c_remove()
as we are going to kfree onyx.
What we really want here is i2c_set_clientdata(client, NULL);
Since the i2c core will take care of it now, so this patch just removes it.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changes to both I2S and PCM code:
- Rates list extended up to 96kHz, it's tested on EDB9302 and works for both capture and
playback.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: fix deemphasis control in wm8904/55/60 codecs
Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Peter Ujfalusi [Wed, 8 Dec 2010 13:12:56 +0000 (15:12 +0200)]
ASoC: tlv320dac33: Fix compillation error
Fix the compilation error introduced by patch:
ASoC: tlv320dac33: Avoid multiple soft power up
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Peter Ujfalusi [Wed, 8 Dec 2010 14:04:33 +0000 (16:04 +0200)]
ASoC: tlv320dac33: Move DAC LR power on to a supply widget
The power for the DACs need to be enabled, even when only
the analog bypass is in use with the codec, otherwise
the audio is going to be distorted.
Make sure that the DACs are powered all the time, when
there is audio activity.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Peter Ujfalusi [Wed, 8 Dec 2010 14:04:32 +0000 (16:04 +0200)]
ASoC: tlv320dac33: Rename outpup amplifier widget
Use better name for the widget, and remove the 'Power'
from it's name.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Brian Bloniarz [Wed, 8 Dec 2010 20:45:20 +0000 (12:45 -0800)]
ALSA: ice1712 - working M-Audio Delta 66E support
Rev. E of the M-Audio Delta 66 is partially supported (commit ef2cd2ccad66b4aba518eca7514eface267ee0f3), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.
ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Uk Kim [Tue, 7 Dec 2010 13:58:40 +0000 (13:58 +0000)]
ASoC: Add ADC high pass filter support to WM8994
Signed-off-by: Uk Kim <w0806.kim@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 7 Dec 2010 17:14:56 +0000 (17:14 +0000)]
ASoC: Support WM8994 mono AIF configurations
The WM8994 supports mono signals - enable this in the driver. With DSP
mode an automatic data channel selector is available, activate this
when in mono mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In case the codec driver did not provide a read/write function,
codec->driver->read|write will be NULL. Ensure that we use the one
specified in codec->read|write to avoid oopsing when we access
the debugfs entries. This is achieved by using snd_soc_read() and
snd_soc_write().
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
Some newer chips have more than one HDMI output, but usually not
all of them are exposed as physical jacks. Removing the unused
PCM devices (as indicated by BIOS in the pin config default) will
reduce user confusion as they currently have to choose between
several HDMI devices, some of them not working anyway.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Anssi Hannula [Tue, 7 Dec 2010 19:19:23 +0000 (21:19 +0200)]
ALSA: hda - Reset sample sizes and max bitrates when reading ELD
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.
The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.
The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.
Fix that by always clearing sample_bits and max_bitrate when reading
SADs.
Anssi Hannula [Tue, 7 Dec 2010 18:56:19 +0000 (20:56 +0200)]
ALSA: hda - Always allow basic audio irrespective of ELD info
Commit bbbe33900d1f3c added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, according to CEA-861-D no SAD is needed for basic audio
(32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a
basic audio flag in the CEA EDID Extension.
The flag is not present in ELD. However, as all audio capable sinks are
required to support basic audio, we can assume it to be always
available.
Fix allowed audio formats with sinks that have SADs (Short Audio
Descriptors) which do not completely overlap with the basic audio
formats (there are no reports of affected devices so far) by always
assuming that basic audio is supported.
Anssi Hannula [Tue, 7 Dec 2010 16:41:35 +0000 (18:41 +0200)]
ALSA: hda - Do not wrongly restrict min_channels based on ELD
Commit bbbe33900d1f3c added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, it wrongly assumes that the bits 0-2 of the first byte of
CEA Short Audio Descriptors mean a supported number of channels. In
reality, they mean the maximum number of channels (as per CEA-861-D
7.5.2). This means that the channel count can only be used to restrict
max_channels, not min_channels.
Restricting min_channels causes us to deny opening the device in stereo
mode if the sink only has SADs that declare larger numbers of channels
(like Primare SP32 AV Processor does).
Fix that by not restricting min_channels based on ELD information.
Jassi Brar [Tue, 7 Dec 2010 10:23:07 +0000 (19:23 +0900)]
ASoC: WM8580: Debug BCLK and sample size
In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32,
rather than WM8580_AIF_LENGTH_24.
Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
projects / karo-tx-linux.git / log