It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
ALSA: hda - Merge alc*_parse_auto_config() functions in patch_realtek.c
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function. A good amount of code reduction.
One more code reduction. This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec. Thus one needs to check widget-caps first, then check
the corresponding amp-caps.
Lydia Wang [Fri, 8 Jul 2011 06:02:52 +0000 (14:02 +0800)]
ALSA: hda - Fix VIA output-path init for VT2002P/1802/1812
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Axel Lin [Wed, 6 Jul 2011 13:20:42 +0000 (21:20 +0800)]
ASoC: pxa2xx-pcm: remove unused variable 'dai'
Remove unused variable 'dai' to eliminate below warning.
CC sound/soc/pxa/pxa2xx-pcm.o
sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new':
sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai'
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: hda - Split quirk codes from patch_realtek.c
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model. For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.
The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed. This is just a refactoring.
ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.
In the new code, the following strategy is taken:
- When a cap-src can't handle all input-sources, either skip it, or
switch to the ADC-switching mode. In ADC-switching mode, like the
former dual-ADC mode for ALC275, it changes ADC on the fly according
to the current input source.
- When auto-mic is possible, always assign imux. If the mic pins are
set statically via a quirk, rebuild imux according to the pins.
In the auto-mic mode, the driver always changes the imux (although
the imux isn't exposed as a mixer element).
ALSA: hda - Parse ADCs in alc_auto_create_input_ctls()
Parse ADCs and cap-srcs in alc_auto_create_input_ctls() by itself
instead of passing explicitly from the caller. By this change, all
alc*_auto_create_input_ctls() can be unified to the same calls.
ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers
Use the same common code for auto-parsing the output paths and their
initializations, based on the existing ALC662 code, which is smarter
than the old ALC880/2 code.
ALSA: hda - Parse HP and speaker DACs even for multi connections for ALC662
In alc662_auto_fill_dac_nids(), the HP and speaker DACs aren't parsed
when the corresponding pins aren't fixed with single DACs.
Now check these DACs even for non-fixed pins.
ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Kailang Yang [Wed, 6 Jul 2011 07:51:29 +0000 (09:51 +0200)]
ALSA: hda - Add Realtek ALC269VC codec support
Add the support of ALC269VC codec.
Also delete the unnecessary codec_variant type enum list:
now only three variants (ALC269VA ALC269VB ALC269VC) are needed.
In addition, added some aliases:
- Add ALC269VB alias name ALC277
- Add ALC269VC alias name ALC259 ALC281X
- Add ALC269VC for Lenovo device 0x21f3 name ALC3202
Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stephen Warren [Tue, 5 Jul 2011 16:55:27 +0000 (10:55 -0600)]
ASoC: Tegra: Implement SPDIF CPU DAI
This is a minimal driver for the Tegra SPDIF controller.
In hardware, the SPDIF output signal is always routed to any active HDMI
display controllers, and may also be routed to external pins on Tegra
using the pinmux.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Mon, 4 Jul 2011 17:27:51 +0000 (10:27 -0700)]
ASoC: Manage WM8731 ACTIVE bit as a supply widget
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 4 Jul 2011 17:27:51 +0000 (10:27 -0700)]
ASoC: Manage WM8731 ACTIVE bit as a supply widget
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com> Acked-by: Liam Girdwood <lrg@ti.com>
ASoC: Don't set invalid name string to snd_card->driver field
The snd_card->driver field contains a driver name string, and in
general it shouldn't contain space or special letters. The commit 2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from
card->name, but the long name string may contain such letters, thus
it may still lead to a segfault.
A temporary fix is not to copy the long name string but just keep it
empty as the earlier version did.
Commit af46800 ("ASoC: Implement mux control sharing") revealed that
"Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing
to the same kcontrols and codec registers and thus soc-core falsely detected
them as shared controls. This is actually wrong since there are separate
registers in hardware that configure Line1L to RADC and Line1R to LADC cross
connects so these muxes should not be shared.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
Stephen Warren [Fri, 1 Jul 2011 19:56:14 +0000 (13:56 -0600)]
ASoC: Tegra: I2S: s/clk_get_sys/clk_get/
The clock needed by the I2S driver is associated with the I2S device name
in the standard fashion. Hence, use clk_get(dev) instead of clk_get_sys(clk_name).
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren [Fri, 1 Jul 2011 19:56:13 +0000 (13:56 -0600)]
ASoC: Tegra: I2S: Ensure clock is enabled when writing regs
The I2S controller needs a clock to respond to register writes. Without
this, register writes will at worst hang the CPU. In practice, I've only
observed writes being dropped.
Luckily, the dropped register writes historically had no effect:
TEGRA_I2S_TIMING: The value we wrote was the reset default.
TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data
when one slot was empty. The requested value was for the FIFOs to request
when four slots were empty. The DMA controller in the mainline kernel is
configured to burst a single entry at a time into the FIFO, hence there
was no issue. The only negative effect was on bus efficiency losses due
to an increased number of arbitration attempts.
However, in various non-upstream changes, the DMA controller now bursts
four entries at a time into the FIFO. If there is only space for one
entry, the data is simply dropped. In practice, this resulted in 3/4 of
samples being dropped, and playback at 4x the expected rate and pitch.
By fixing the clocking issue, this is solved.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals. For avoiding this, the aa-mix routes have to be disabled
for surround paths.
Lydia Wang [Mon, 4 Jul 2011 09:01:33 +0000 (17:01 +0800)]
ALSA: hda - Fix invalid multi-channel amplifiers for VT1718S
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;
But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Mon, 4 Jul 2011 08:54:15 +0000 (16:54 +0800)]
ALSA: hda - Fix issue that front can't output sound for VT1718S
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Fri, 1 Jul 2011 02:55:07 +0000 (10:55 +0800)]
ALSA: hda - Fix the silent front with independent-HP for VIA codecs
Unmute DAC on front speaker path when Independent HP is enabled.
When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().
If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 30 Jun 2011 15:24:47 +0000 (17:24 +0200)]
ALSA: hda - Re-implementation of VIA Independent-HP sharing with side stream
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.
When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.
One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.
Takashi Iwai [Thu, 30 Jun 2011 13:08:04 +0000 (15:08 +0200)]
ALSA: sb16 - Fix build errors on MIPS and others with 13bit ioctl size
One of ioctl definition in sound/sb16_csp.h contains the data size
over 8kB, and this causes build errors on architectures like MIPS,
which define _IOC_SIZEBITS=13.
For avoiding this build errors but keeping the compatibility, manually
expand with _IOC() instead of using _IOW() for the problematic ioctl.
Takashi Iwai [Thu, 30 Jun 2011 10:45:36 +0000 (12:45 +0200)]
ALSA: hdspm - Fix compile warnings with PPC
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Takashi Iwai [Thu, 30 Jun 2011 10:31:23 +0000 (12:31 +0200)]
ALSA: cs5535 - Fix invalid big-endian conversions
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
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