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 19:56:13 +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
Mark Brown [Mon, 13 Jun 2011 11:14:07 +0000 (12:14 +0100)]
ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Wu Fengguang [Wed, 29 Jun 2011 06:26:07 +0000 (14:26 +0800)]
ALSA: HDMI - fix ELD monitor name length
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Lydia Wang [Wed, 29 Jun 2011 05:52:02 +0000 (13:52 +0800)]
ALSA: hda - Fix jack-detection on non-VT1708 VIA codecs
Move codec init verb which is only applicatable for VT1708.
I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 28 Jun 2011 10:45:47 +0000 (12:45 +0200)]
ALSA: hda - Create snd_hda_get_conn_index() helper function
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Takashi Iwai [Tue, 28 Jun 2011 10:45:47 +0000 (12:45 +0200)]
ALSA: hda - Create snd_hda_get_conn_index() helper function
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Takashi Iwai [Tue, 28 Jun 2011 06:59:30 +0000 (08:59 +0200)]
ALSA: hda - Fix warnings with CONFIG_SND_POWER_SAVE=n
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Takashi Iwai [Mon, 27 Jun 2011 14:17:07 +0000 (16:17 +0200)]
ALSA: hda - More volume-init fixes for ALC267 codec
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Takashi Iwai [Mon, 27 Jun 2011 13:53:38 +0000 (15:53 +0200)]
ALSA: hda - Fix volume-init for ALC259 with invalid widget caps
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Takashi Iwai [Mon, 27 Jun 2011 13:48:17 +0000 (15:48 +0200)]
ALSA: hda - Fix volume-init of ALC299 & co
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Takashi Iwai [Mon, 27 Jun 2011 10:34:01 +0000 (12:34 +0200)]
ALSA: hda - Check hard-wired DACs at first for ALC662 & co
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Takashi Iwai [Mon, 27 Jun 2011 09:32:07 +0000 (11:32 +0200)]
ALSA: hda - Call proper DAC-filler function for Realtek auto-parser
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Takashi Iwai [Fri, 24 Jun 2011 12:10:28 +0000 (14:10 +0200)]
ALSA: hda - Add snd_hda_get_conn_list() helper function
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Tim Blechmann [Fri, 24 Jun 2011 15:36:20 +0000 (17:36 +0200)]
ALSA: lx6464es - include mac address in device name
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jesper Juhl [Thu, 23 Jun 2011 21:54:40 +0000 (23:54 +0200)]
ALSA: Remove unneeded version.h includes from sound/
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Takashi Iwai [Fri, 24 Jun 2011 09:03:58 +0000 (11:03 +0200)]
ALSA: hda - Clean up multi-channel mixer name assignment in patch_realtek.c
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Takashi Iwai [Fri, 24 Jun 2011 08:43:03 +0000 (10:43 +0200)]
ALSA: hda - Add a workaround for invalid line-out setups
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Daniel Mack [Wed, 22 Jun 2011 18:48:25 +0000 (20:48 +0200)]
ASoC: pxa-ssp: Correct check for stream presence
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Mark Brown [Wed, 22 Jun 2011 19:07:24 +0000 (20:07 +0100)]
ASoC: Add basic WM8918 support
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure] Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure] Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Wed, 22 Jun 2011 13:23:25 +0000 (15:23 +0200)]
ALSA: hda - Implement dynamic-ADC switching for VIA codecs
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Taylor Hutt [Mon, 20 Jun 2011 18:54:32 +0000 (11:54 -0700)]
ASoC: codecs: Max98095: Fix logging of hardware revision.
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.
Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().
Signed-off-by: Taylor Hutt <thutt@chromium.org> Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Tue, 21 Jun 2011 12:22:14 +0000 (14:22 +0200)]
ALSA: hda - Fix surround-volume parsing for VT1708B codecs
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.