Vinod Koul [Wed, 17 Jun 2015 05:50:17 +0000 (11:20 +0530)]
ALSA: hda: add hda link cleanup routine
In HDA extended bus the HDA link objects are created when multilink
capabilities are parsed. We need a routine which free up these link objects
for a bus. So add snd_hdac_link_free_all routine
Vinod Koul [Wed, 17 Jun 2015 05:50:16 +0000 (11:20 +0530)]
ALSA: hda: add hdac_ext stream creation and cleanup routines
HDAC extended core should create streams for an extended bus and also free
up those on cleanup. So introduce snd_hdac_ext_stream_init_all and
snd_hdac_stream_free_all routines
Vinod Koul [Tue, 16 Jun 2015 15:30:22 +0000 (21:00 +0530)]
ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
Since this is common option for HDA driver to specfiy pre-allocated
buffer, we should make this option availble to all HDA driver by
moving this to HDA core
Takashi Iwai [Tue, 16 Jun 2015 10:23:36 +0000 (12:23 +0200)]
ALSA: hda - Fix unused label skip_i915
When CONFIG_SND_HDA_I915=n, we get a compile warning:
sound/pci/hda/hda_intel.c: In function ‘azx_probe_continue’:
sound/pci/hda/hda_intel.c:1882:2: warning: label ‘skip_i915’ defined but not used [-Wunused-label]
Fix it by putting again ifdef to it. Sigh.
Fixes: bf06848bdbe5 ('ALSA: hda - Continue probing even if i915 binding fails') Reported-by: Borislav Petkov <bp@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new Dell XPS13 also requires the similar quirk for fixing the
noisy outputs. (But, as the codec was changed, now the fixup for
Latitude is used instead.)
Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.
ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.
This commit makes this driver to keeps duplex streams always.
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:35 +0000 (12:49 +0900)]
ALSA: bebob: loosen up severity of checking continuity for BeBoB v3 quirk
PrismSound Orpheus, Behringer UFX1604 and FCA610 work with BeBoB v3, and
they're confirmed to transmit discontinuous packets in the beginning of
streaming.
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:34 +0000 (12:49 +0900)]
ALSA: bebob: expand timeout for DM1500 quirk
Behringer FCA610 and UFX1604 is confirmed to require more time till
transmitting packets after establishing connections. This seems to
be a quirk of DM1500 ASIC which ArchWave produced.
For this quirk, this commit extends the time to wait up to 2 seconds.
As a result, in worst cases, below userspace functions require 2 seconds
to return.
- snd_pcm_prepare()
- snd_pcm_hw_params()
- snd_pcm_recover()
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:30 +0000 (12:49 +0900)]
ALSA: bebob: use normalized representation for the type of clock source
This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:29 +0000 (12:49 +0900)]
ALSA: bebob: preparation for replacing string literals by normalized representation for model-dependent structures
Previous commit adds a enumerator as a normalized representation of
clock source, while model-dependent structures still use string literals
for this purpose.
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:27 +0000 (12:49 +0900)]
ALSA: bebob: improve signal mode detection for clock source
With BeBoB version 3, current ALSA BeBoB driver detects the type of
current clock signal source wrongly. This is due to a lack of proper
implementation to parse the information.
This commit renews the parser. As a result, this driver detects
SYT-Match clock signal, thus it can start streams with two modes;
SYT-Match mode and the others. SYT-Match mode will be supported in future
commits.
There's a constrain about detected internal/external clock source.
When detecting external clock source, this driver allows userspace
applications to use current sampling rate only. This is due to consider
abour synchronization to external clock sources such as S/PDIF, ADAT or
word-clock.
According to several information from some devices, I guesss that the
internal clock of most devices synchronize to IEEE 1394 cycle start
packet. In this case, by a usual way, it's detect as 'Sync type
of output Music Sub-Unit' connected to 'Sync type of PCR output Unit
(oPCR)', and this driver judges it as internal clock. Therefore,
userspace applications is allowed to request arbitrary supported sampling
rates.
On the other hand, several devices based on BeBoB version 3 have
additional internal clock. In this case, by a usual way, it's detect as
'Sync/Additional type of External input Unit'. Unfortunately, there's no
way to distinguish this sync type from the other external clock sources
such as word-clock. In this case, this driver handles it as external and
userspace applications is forced to use current sampling rate.
I note that when the source of clock is detected as 'Isochronous stream
type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the
synchronization clock is generated according to SYT-series in received
packets. In this case, this driver generates the series by myself. I
experienced this mode often make the device silent suddenly during
playbacking. This means that the mode is easy to lost synchronization.
Takashi Iwai [Mon, 15 Jun 2015 09:59:32 +0000 (11:59 +0200)]
ALSA: hda - Fix audio crackles on Dell Latitude E7x40
We still got a report that the audio crackles and noises occur with
the recent 4.1 kernels on Dell machines. These machines seem to need
similar workarounds that have been applied to the recent Dell XPS 13
models. Since the codec of these machines (Dell Latitute E7240 and
E7440) is different from XPS 13's one, we need a new fixup entry.
Also, it was confirmed that the previous workaround to disable the
widget power-save (commit [219f47e4f964: ALSA: hda - Disable widget
power-saving for ALC292 & co]) is no longer needed after this fix.
So, this patch includes the partial revert of the commit, too.
Reported-and-tested-by: Mihai Donțu <mihai.dontu@gmail.com> Tested-by: Jonathan McDowell <noodles@earth.li> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hui Wang [Mon, 15 Jun 2015 09:43:39 +0000 (17:43 +0800)]
ALSA: hda - adding a DAC/pin preference map for a HP Envy TS machine
On a HP Envy TouchSmart laptop, there are 2 speakers (main speaker
and subwoofer speaker), 1 headphone and 2 DACs, without this fixup,
the headphone will be assigned to a DAC and the 2 speakers will be
assigned to another DAC, this assignment makes the surround-2.1
channels invalid.
To fix it, here using a DAC/pin preference map to bind the main
speaker to 1 DAC and the subwoofer speaker will be assigned to another
DAC.
Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 12 Jun 2015 05:53:58 +0000 (07:53 +0200)]
ALSA: hda - Abort the probe without i915 binding for HSW/BDW
The previous patch tried to continue the probe if i915 binding fails.
For for simplicity reason, we haven't implemented abort even for
controller chips that are dedicated for HDMI/DP on HSW and BDW.
However, Mengdong suggested that this can be dangerous; BIOS may
disable gfx power well although the PCI entry for HD-audio is left,
and this may result in the unexpected behavior, kernel errors, etc.
For avoiding this situation, abort the probe at i915 binding failure
only for HSW/BDW chips selectively. For other chips, it still
continues.
Fixes: bf06848bdbe5 ('ALSA: hda - Continue probing even if i915 binding fails') Reported-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 11 Jun 2015 12:02:49 +0000 (14:02 +0200)]
ALSA: hda - Fix link power unbalance at device removal
snd_hdac_link_power() has to be called after unregistering the codec
device. Otherwise the device might be already runtime-suspended, thus
the refcount goes under zero, triggering a warning like:
Jeeja KP [Thu, 11 Jun 2015 08:41:49 +0000 (14:11 +0530)]
ALSA: hdac_ext: add extended stream capabilities
Now we have the bus and controller code added to find and initialize
the extended capabilities. Now we need to use them in stream code to
decouple stream, manage links etc
So this patch adds the stream handling code for extended capabilities
introduced in preceding patches
Jeeja KP [Thu, 11 Jun 2015 08:41:47 +0000 (14:11 +0530)]
ALSA: hdac_ext: add extended HDA bus
The new HDA controllers from Intel support new capabilities like
multilink, pipe processing, SPIB, GTS etc In order to use them we
create an extended HDA bus which embed the hdac bus and contains the
fields for extended configurations
Johan Rastén [Thu, 11 Jun 2015 08:04:51 +0000 (10:04 +0200)]
ALSA: usb-audio: Set correct type for some UAC2 mixer controls.
Changed ctl type for Input Gain Control and Input Gain Pad Control to
USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0
definition.
Signed-off-by: Johan Rastén <johan@oljud.se> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 11 Jun 2015 08:51:28 +0000 (10:51 +0200)]
ALSA: hda - Re-add the lost fake mute support
Yet another regression by the transition to regmap cache; for better
usability, we had the fake mute control using the zero amp value for
Conexant codecs, and this was forgotten in the new hda core code.
Since the bits 4-7 are unused for the amp registers (as we follow the
syntax of AMP_GET verb), the bit 4 is now used to indicate the fake
mute. For setting this flag, snd_hda_codec_amp_update() becomes a
function from a simple macro. The bonus is that it gained a proper
function description.
Takashi Iwai [Wed, 10 Jun 2015 10:03:49 +0000 (12:03 +0200)]
ALSA: hda - Continue probing even if i915 binding fails
Currently snd-hda-intel driver aborts the probing of Intel HD-audio
controller with i915 power well management when binding with i915
driver via hda_i915_init() fails. This is no big problem for Haswell
and Broadwell where the HD-audio controllers are dedicated to
HDMI/DP, thus i915 link is mandatory. However, Skylake, Baytrail and
Braswell have only one controller and both HDMI/DP and analog codecs
share the same bus. Thus, even if HDMI/DP isn't usable, we should
keep the controller working for other codecs.
For fixing this, this patch simply allows continuing the probing even
if hda_i915_init() call fails. This may leave stale sound components
for HDMI/DP devices that are unbound with graphics. We could abort
the probing selectively, but from the code simplicity POV, it's better
to continue in all cases.
Reported-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 10 Jun 2015 08:27:00 +0000 (10:27 +0200)]
ALSA: hda - Don't actually write registers for caps overwrites
Along with the transition to regmap for managing the cached parameter
reads, the caps overwrite was also moved to regmap cache. The cache
change itself works, but it still tries to write the non-existing verb
(the HDA parameter is read-only) wrongly. It's harmless in most
cases, but some chips are picky and may result in the codec
communication stall.
This patch avoids it just by adding the missing flag check in
reg_write ops.
Lu, Han [Tue, 9 Jun 2015 08:50:38 +0000 (16:50 +0800)]
ALSA: hda: Intel: enable automatic runtime pm for HDMI codecs by default
Enable runtime PM of the HDMI audio codec on the latest Intel platforms.
So the HD-A controller or HDMI codec can suspend when idle timeout by
default and release the GFX power well.
The patch influences HSW/BDW/BYT/BSW/SKL. Eariler platforms and third
party analog codecs will not be influenced.
Signed-off-by: Lu, Han <han.lu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 8 Jun 2015 19:04:24 +0000 (21:04 +0200)]
ASoC: intel: Remove unused variable hsw
The recent fix left a variable declaration without usage.
sound/soc/intel/haswell/sst-haswell-pcm.c:1349:18: warning: unused variable ‘hsw’ [-Wunused-variable]
Takashi Iwai [Mon, 8 Jun 2015 18:47:53 +0000 (20:47 +0200)]
Merge tag 'asoc-v4.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
Peter Ujfalusi [Thu, 4 Jun 2015 13:04:30 +0000 (16:04 +0300)]
ASoC: tas2552: Configure the WCLK frequency based on the stream
Instead of hard wiring the WCLK frequency at probe time do it runtime.
The hard wired 88_96KHz was not even setting the correct bits since it was
defined as (1 << 6) which will change the I2S_OUT_SEL bit and will leave
the amplifier configured for 8KHz.
At the same time clean up and fix the CFG3 register bits.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 4 Jun 2015 13:04:28 +0000 (16:04 +0300)]
ASoC: tas2552: Implement startup/stop sequence as per TRM
Certain sequence need to be followed in order to have smooth power up and
power down performance.
Execute this sequence via DAPM_POST widget.
Remove patching the RESERVED_0D register at probe time since it has to be
handled every time when we stop or start the amplifier.
In order to be able to execute the sequence at the correct time, the driver
need to request to ignore the pmdown time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 4 Jun 2015 13:04:27 +0000 (16:04 +0300)]
ASoC: tas2552: Correct the Speaker Driver Playback Volume (PGA_GAIN)
The last parameter for DECLARE_TLV_DB_SCALE() is to tell if the gain will
be muted or not when it is set to raw 0. IN this case it is not muted.
The PGA_GAIN is in 0-4 bits in the register. Fix the offset in the
SOC_SINGLE_TLV() for this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 4 Jun 2015 13:04:26 +0000 (16:04 +0300)]
ASoC: tas2552: Clean up the Digital - Analog DAPM route definition
The strings should be: 'static const char * const tas2552_input_texts[]'
SOC_DAPM_ENUM should have "Route" in place of xname and no need to have it
as an array.
Also align the parameters.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 4 Jun 2015 13:04:23 +0000 (16:04 +0300)]
ASoC: tas2552: Correct dai format support
DSP_A mode require one bit delay from the FS, DSP_B is without data delay.
When checking the requested format, also match the bit and fs inversion
flag along with the format since it is not possible to change inversion.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 4 Jun 2015 13:04:16 +0000 (16:04 +0300)]
ASoC: tas2552: Correct PDM configuration register bit definitions
The PDM clock can be selected via bit0-1.
PDM_DATA_ES bit is at bit2.
The code were trying to select BCLK as PDM reference clock but instead
it was selecting PLL and set the DATA_ES bit to 1.
Selecting the PLL output as reference clock as default does make sense,
but the driver should not change the PDM data edge.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Liam Girdwood [Thu, 4 Jun 2015 14:13:09 +0000 (15:13 +0100)]
ASoC: dapm: fix build errors for missing snd_soc_dapm_new_control symbol
Fix the following error:-
All error/warnings (new ones prefixed by >>):
>
> sound/built-in.o: In function `soc_tplg_dapm_widget_create':
> >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control'
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Liam Girdwood [Fri, 29 May 2015 18:06:14 +0000 (19:06 +0100)]
ASoC: topology: Add topology core
The topology core parses the FW topology file for known block types and
instanciates any common ALSA/ASoC objects that it discovers. The core
also passes any block that is does not understand to client component
drivers for enumeration.
The core exports some APIs to client drivers in order to load and unload
firmware topology data as use case require.
Currently the core deals with the following object types :-
o kcontrols. This includes TLV, enumerated and bytes controls.
o DAPM widgets. All types with any associated kcontrol.
o DAPM graph.
o FE PCM. FE PCM capabilities and configuration can be defined.
o BE DAI Link. BE DAI link capabilities and configuration can be defined.
o Codec <-> codec style links capabilities and configuration.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Liam Girdwood [Fri, 29 May 2015 18:06:13 +0000 (19:06 +0100)]
ASoC: topology: Add topology UAPI header
The ASoC topology UAPI header defines the structures
required to define any DSP firmware audio topology and control objects from
userspace.
The following objects are supported :-
o kcontrols including TLV controls.
o DAPM widgets and graph elements
o Vendor bespoke objects.
o Coefficient data
o FE PCM capabilities and config.
o BE link capabilities and config.
o Codec <-> codec link capabilities and config.
o Topology object manifest.
The file format is simple and divided into blocks for each object type and
each block has a header that defines it's size and type. Blocks can be in
any order of type and can either all be in a single file or spread across
more than one file. Blocks also have a group identifier ID so that they can
be loaded and unloaded by ID.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jeeja KP [Wed, 3 Jun 2015 06:54:32 +0000 (12:24 +0530)]
ALSA: hda - add new HDA registers
This patch adds new registers as per HD audio Spec like capability registers
for processing pipe, software position based FIFO, Multiple Links and Global
Time Synchronization.
Vinod Koul [Wed, 3 Jun 2015 06:54:31 +0000 (12:24 +0530)]
ALSA: hda - add HDA default codec match function
HDA codec drivers can be matched using vendor id and revision id typically.
So provide a match function which does this and is loaded when driver hasn't
provided one (default behaviour)
Clemens Ladisch [Wed, 3 Jun 2015 09:36:51 +0000 (11:36 +0200)]
ALSA: usb-audio: fix missing input volume controls in MAYA44 USB(+)
The driver worked around an error in the MAYA44 USB(+)'s mixer unit
descriptor by aborting before parsing the missing field. However,
aborting parsing too early prevented parsing of the other units
connected to this unit, so the capture mixer controls would be missing.
Fix this by moving the check for this descriptor error after the parsing
of the unit's input pins.