Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.
This commit renames PCM format setting function to prepare for integrating
AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, MIDI messages are transferred in MIDI conformant data
channel. Essentially, packet streaming layer is not responsible for MIDI
functionality.
This commit moves MIDI trigger helper function from the layer to AM824
layer. The rest of codes related to MIDI functionality will be moved in
later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, several types of data are available in AM824 format. The
data is transferred in each data channel. The position of data channel in
data block differs depending on model.
Current implementation has an array to map the index of data channel in an
data block to the position of actual data channel. The implementation
allows each driver to access the mapping directly.
In later commit, the mapping is in specific structure pushed into an
opaque pointer. Helper functions are required.
This commit adds the helper functions for this purpose. In IEC 61883-6,
AM824 format supports many data types, while this specification easily
causes over-engineering. Current AM824 implementation is allowed to handle
two types of data, Multi Bit Linear Audio data (=PCM samples) and MIDI
conformant data (=MIDI messages).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.
This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value of FDF field in CIP header is protocol-dependent. Thus, it's
better to allow data block processing layer to decide the value in any
timing.
In AM824 data format, the value of FDF field in CIP header indicates
N-flag and Nominal Sampling Frequency Code (sfc). The N-flag is for
switching 'Clock-based rate control mode' and 'Command-based rate control
mode'. In our implementation, 'Clock-based rate control mode' is just
supported. Therefore, When sampling transfer frequency is decided, then
the FDF can be set.
This commit replaces 'amdtp_stream_set_parameters' with
'amdtp_am824_set_parameters' to set the FDF. This is the same timing
to decide the ration between the number of data blocks and the number of
PCM frames.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds data block processing layer for AM824 format. The new
layer initializes streaming layer with its value for fmt field.
Currently, most implementation of data block processing still remains
streaming layer. In later commits, these codes will be moved to the layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, data block processing layer will be newly added. This
layer will be named as 'amdtp-am824'.
This commit renames current amdtp file to amdtp-stream, to distinguish it
from the new layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some vendor specific protocol uses its own value for fmt/fdf fields in
CIP header.
This commit support to set arbitrary values for the fields.
In IEC 61883-6, NO-DATA code is defined for FDF field. A packet with this
code includes no data even if it includes some data blocks. This commit
still leaves a condition to handle this special packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM framework uses PCM buffer with a concept of 'period' to
synchronize userspace operations to hardware for nearly-realtime
processing. Each driver implements snd_pcm_period_elapsed() to tell across
of the period boundary to ALSA PCM middleware. To call the function, some
drivers utilize hardware interrupt handlers, the others count handled PCM
frames.
Drivers for sound units on IEEE 1394 bus are the latter. They use two
buffers; PCM buffer and DMA buffer for IEEE 1394 isochronous packet. PCM
frames are copied between these two buffers and 'amdtp_stream' structure
counts the handled PCM frames. Then, snd_pcm_period_elapsed() is called if
required.
Essentially, packet streaming layer should not be responsible for PCM
frame processing. The PCM frame processing should be handled in each data
block processing layer as a result of handling data blocks. Although, PCM
frame counting is a common work for all of protocols which ALSA firewire
stack is going to support.
This commit adds two new helper functions as interfaces between packet
streaming layer to data block processing layer. In future, each data block
processing layer implements these functions. The packet streaming layer
calls data block processing layer per packet by calling the functions. The
data block processing layer processes data blocks and PCM frames, and
returns the number of processed PCM frames. Then the packet streaming layer
calculates handled PCM frames and calls snd_pcm_period_elapsed().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In future commit, interface between data block processing layer and packet
stream processing layer is defined. These two layers communicate the
number of data blocks and the number of PCM frames.
The data block processing layer has a responsibility for calculating the
number of PCM frames. Therefore, 'dual wire' of Dice quirk should be
handled in data block processing layer.
This commit adds a member of 'frame_multiplier'. This member represents
the ratio of the number of PCM frames against the number of data blocks.
Usually, the value of this member is 1, while it's 2 in Dice's 'dual wire'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, one data block represents one event. In ALSA, the event is
one PCM frame. Therefore, when processing one data block, current
implementation counts one PCM frame.
On the other hand, Dice platform has a quirk called as 'dual wire' at
higher sampling rate. In detail, see comment of commit 6eb6c81eee2a
("ALSA: dice: Split stream functionality into a file").
Currently, to handle this quirk, AMDTP stream structure has a
'double_pcm_frames' member. When this is enabled, two PCM frames are
counted. Each driver set this flag by accessing the structure member
directly.
In future commit, some members related to AM824 data block will be moved
to specific structure, to separate packet streaming layer and data block
processing layer. The access will be limited by opaque pointer.
For this reason, this commit adds an argument into
amdtp_stream_set_parameter() to set the flag.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, amdtp_stream_set_parameters() returns no error even if wrong
arguments are given. This is not good for streaming layer because drivers
can continue processing ignoring capability of streaming layer.
This commit changes this function to return error code.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, some members related to AM824 data format will be moved
from AMDTP stream structure to data block structure. This commit is a
preparation for it. Additionally, current layout of AMDTP stream structure
is a bit mess by several extensions. This commit also arranges the layout.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpads with recent Realtek codecs seem suffering from click
noises at power transition since the introduction of widget power
saving in 4.1 kernel. Although this might be solved by some delays in
appropriate points, as a quick workaround, just disable the
power_save_node feature for now. The gain it gives is relatively
small, and this makes the situation back to pre 4.1 time.
This patch ended up with a bit more code changes than usual because
the existing fixup for Thinkpads is highly chained. Instead of adding
yet another chain, combine a few of them into a single fixup entry, as
a gratis cleanup.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
The Tegra HD-audio controller driver causes deadlocks when loaded as a
module since the driver invokes request_module() at binding with the
codec driver. This patch works around it by deferring the probe in a
work like Intel HD-audio controller driver does. Although hovering
the codec probe stuff into udev would be a better solution, it may
cause other regressions, so let's try this band-aid fix until the more
proper solution gets landed.
Reported-by: Thierry Reding <treding@nvidia.com>
Tested-by: Thierry Reding <treding@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.
The commit below shows the inversion in the requestor lines.
Fixes: d65a14587a9b ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the error path so that we can free the allocated memory on the error
path instead of releasing them individually on each error.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have requested for the firmware but we have missed releasing it both
on success and on error path.
While checking the code it turned out that the requested firmware is not
even used. More over the same firmware is being loaded by
wm0010_stage2_load().
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the max register value of mic boost pga should be 3.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm8962 can't support 64k sample rate. When playing a 64KHz wave file,
'Unsupported rate 64000Hz' will be prompted.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The '\n' at the end of the format string is not needed. It adds an extra
line break when doing
cat /proc/interrupts
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous fix of pxa library support, which was introduced to fix the
library dependency, broke the previous SoC behavior, where a machine
code binding pxa2xx-ac97 with a coded relied on :
- sound/soc/pxa/pxa2xx-ac97.c
- sound/soc/codecs/XXX.c
For example, the mioa701_wm9713.c machine code is currently broken. The
"select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for
compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared
in sound/arm/Kconfig and sound/soc/pxa/Kconfig.
Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct
pxa2xx-ac97 compilation.
Fixes: 846172dfe33c ("ASoC: fix SND_PXA2XX_LIB Kconfig warning")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TX and RX direction share the same bit clock and frame sync, so
the samplerate must be the same to both directions.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix below build warning:
sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe':
sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable]
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In capture, there is chance that hw_ptr reported at IRQ is
a little smaller than period_size due to internal AFE buffer.
In the case of ping-pong buffer:
|xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------|
hw_ptr < period_size
This available buffer will not be read since its size is smaller than
avail_min (which is period_size by default), and read thread continues
to sleep. If the next hw_ptr is just a little larger than buffer_size,
overrun occurs. One more period can hold the possible unread buffer.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value
for the refining should meet this requirement as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of small fixes since the last update: the HD-audio
quirks as usual with a USB-audio fix and a trivial fix for the
old sparc driver.
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Merge tag 'sound-fix-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes since the last update: the HD-audio quirks
as usual with a USB-audio fix and a trivial fix for the old sparc
driver"
* tag 'sound-fix-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Change internal PCM order
ALSA: hda - Fix white noise on Dell M3800
ALSA: hda - Use ALC880_FIXUP_FUJITSU for FSC Amilo M1437
ALSA: hda - Enable headphone jack detect on old Fujitsu laptops
ALSA: sparc: amd7930: Fix module autoload for OF platform driver
ALSA: hda - Add some FIXUP quirks for white noise on Dell laptop.
The input PGAs have a gain range from -17.25dB to +30dB in 0.75dB steps.
The boost stage can provide additional gain. For line inputs, -12dB to
+6dB gain is available on the boost mixer. For micphone inputs, it can
provide up to +29dB additional gain from the microphone PGA.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The min gain is the corresponding gain value when the register value is 0
instead of 1, just correct it.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch removes the incorrect settings to avoid the pop sound in the
first playback with headphone after boot.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Incase of an unknown event we were directly returning but we missed
freeing params.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add check on of_property_read to return error when
DT required property is not defined.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the SSP port settings are being clobbered as part of the DSP
RTD3 restore logic. make sure we save the correct params and restore them
at resume. The FW sadly does not save SSP settings as part of the PM
context.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M3800 is very minor workstation variant of the XPS 15 which has
already been patched for this issue. I figured it's probably more
important for this version of the laptop to be patched than the
regular XPS as Dell sells is pre-configured with Ubuntu to be used as
a Linux workstation. I have tested the patch on my the hardware on
Linux 4.2.0.
Signed-off-by: Niranjan Sivakumar <ns253@cornell.edu>
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the machine has a bass speaker, so take a correct
fixup entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the bug report, FSC Amilo laptops with ALC880 can detect
the headphone jack but currently the driver disables it. It's partly
intentionally, as non-working jack detect was reported in the past.
Let's enable now.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
