Accessing buffer pointer before initialization results in device crash.
To memset the buffer using physical address also results in a device crash.
Fixed this by initializing the buffer pointer before it is accessed and
memset the buffer using the virtual address.
Change-Id: I3b03f56cf988c9471c7988665bcec3c467e60bfc
Signed-off-by: Amal Paul <amal@codeaurora.org>
- When paused and press next button to play next song,
sometimes CMD_EOS fails to get Ack from LPASS and
is wait timeout for 5sec causing delay for next
playback start.
- In the failure case, even before trigger start
of driver is done,LPA driver receives pcm_close.
In this case, though EOS is issued, it is not
getting honored from LPASS.
- If trigger start not happen in LPA driver,
avoid CMD_EOS to LPASS as it will not be handled.
CRs-Fixed: 368519, 366926
Change-Id: Ib5ff21925bb44849b27ed4709b72efcccf412b5d
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
The current US/EURO headset detection algorithm is overwriting detected
unsupported headset detection with invalid headset detection.
Don't make it to overwrite to report unsupported headset correctly.
CRs-fixed: 359290, 368319
Change-Id: If2d02c0d68be1c6f3f2eb1aa89c7a08ffe166446
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
WCD9310 requires the charge pump to be enable for lineout (speaker)
as well. Fix to add charge pump in the routing for lineout
CRs-fixed: 369639
Change-Id: Ia6f699d1e659c68062d599820768a495d1f8d05a
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Updated dai links to use dpcm trigger posts based on
the latest alsa sound soc framework.
Change-Id: I80819c87e7307ed24874e3ea237ff9d09770818f
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
- Add all codec driver common functionality in common files.
- Add separate files for callback function definitions.
- Add header files according to the platform.
Change-Id: I906811043d4bb33571f719f79988fbdb89f5c385
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>
-RT proxy driver is using DMA alloc. Migrate them to
use ION memory to save memory and avoid stability
issues.
Change-Id: Idc2296bc5ecf76d6b846f204478c33f58423bb72
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
There are use cases that 16kHz sample rate AUX PCM needed
besides 8kHz. Add the amix control so that the AUX PCM
sample rate can be configured by application.
Change-Id: Ie2eaa5e844057065af4cedd64aa2c040f392721b
Signed-off-by: Shiv Maliyappanahalli <smaliyap@codeaurora.org>
CPU, platform and codec drivers does not support bespoke trigger.
update the front end dai links trigger option from bespoke to
dpcm trigger post.
Change-Id: I15bbb3b3fb1611bc3f77f729830fc8a6ba72b40a
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
The WCD9320 Codec Driver is an ALSA-compliant codec driver. This driver
constructs the internal codec audio paths with DAPM widgets and
controls and provides the controls to the upper layers to enable and
configure audio paths.
Change-Id: Iee29359bebfc838cd200732b7191a2eb6a2087ee
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
wcd9310's MUX for calibration is connected only to micbias 2.
Therefore if MBHC micbias is other than micbias 2, it's required to route
override to micbias 2 during calibration.
Set cfilt which is associated with micbias 2 as fast mode during
calibration for the same reason.
CRs-fixed: 369684
Change-Id: I910a8d6747ad013fbd7a006662a650f048ffc545
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Both audio path and MBHC accessory detection logic need master clock
enabled along with bandgap and clock block of CODEC. Clock control
is managed by the machine driver. Calls to clock control function from
audio and MBHC detection paths can be nested. As a result, reference
counter of master clock is incremen/decrement out of order and cause
master clock not enabled when audio path is enabled. Without master clock
CODEC will not consume data and this leads to SLIMBUS overflow error.
CRs-fixed: 370335
Change-Id: Id8d3b98496c95bb7ada9ca102fb867f52c0e500c
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Add 4 FE to support multiple playback use case as mentioned
4 parallel pcm and 4 parallel compressed playback
1) system tone
2) local file playback
3) picture in picture channel 1
4) picture in picture channel 2
Change-Id: Ibab38f18146aeb20273546005586846a59d1cd46
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
This patch ads core.c, the file which implements the ioctls and
registers the devices
Change-Id: I30a1d8b561ecd15b6a862d6948368394edd49665
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
ASoC core will scan for path, get the list of widgets
when playback and capture is started.When a mixer
command is issued it needs to scan only the path
to find if back end or front end dai needs to be shutdown.
Change ehances asoc core path finding functionality
to provide support for different usecases.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Change the wigdet power up and power down sequnce
in the ASOC as per msm platform requirements.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
When hardware params are set for the session alsa
core will query for the RX/TX channels for the
current back end dai.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Change adds debugfs support which gives information
on params, state of active pcm instances.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Hostless PCM nodes will not exchange data with
the userspace clients.Control paths will
be setup by userspace clients.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Change creates a PCM stream for ASOC backend which will only be used
internally by kernel drivers.It provides existing ASoC components
drivers with a substream and access to any private data.
- During bootup, create a file system storing the
information about usb device state and name, and
broadcast the message to userspace whenever the
device is plugged or unplugged.
- Register the device at USB sound card initialization
for creating file system and toggle the state in that
filesystem when USB is plugged/unplugged. This will
also send a message to user space informing it about
state change.
Change-Id: Icd78273bb765a26bbc70725afebe8955de8f7315
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>
Use SPECIAL format for AMR. Add mixer control to
input mode and rate.
Change-Id: I8746d86ce323744995575a22b6128b39daaa3d13
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
[sboyd: Drop soc/msm parts]
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
A few last-minute regression fixes for 3.4 final kernel.
All trivial, and Cc'ed to stable kernel.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few last-minute regression fixes for 3.4 final kernel. All trivial,
and Cc'ed to stable kernel."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Fix AIF2ADC power down
ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops
ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute fixes
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
BIOS on some HP laptops don't set the speaker-pins as fixed but expose
as jacks, and this confuses the driver as if these pins are
jack-detectable. As a result, the machine doesn't get sounds from
speakers because the driver prepares the power-map update via jack
unsol events which never come up in reality. The bug was introduced
in some time in 3.2 for enabling the power-mapping feature.
This patch fixes the problem by replacing the check of the persistent
power-map bits with a proper is_jack_detectable() call.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43240
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Slightly more than expected as rc7, but all are reasonablly small fixes.
A few additions of HD-audio fixup entries, a couple of other regression
fixes including a revert, and a few other trivial oneliners.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Slightly more than expected as rc7, but all are reasonablly small
fixes. A few additions of HD-audio fixup entries, a couple of other
regression fixes including a revert, and a few other trivial
oneliners."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: sh: fix migor.c compilation
ALSA: HDA: Lessen CPU usage when waiting for chip to respond
Revert "ALSA: hda - Set codec to D3 forcibly even if not used"
ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup
ALSA: hdsp - Provide ioctl_compat
ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo
ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G
ALSA: echoaudio: Remove incorrect part of assertion
An API update which wasn't sufficiently thorough in updating the tree...
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Build fix for SH in 3.4
An API update which wasn't sufficiently thorough in updating the tree...
Fix a recent compilation breakage, caused by a change in SH clock API.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When an IRQ for some reason gets lost, we wait up to a second using
udelay, which is CPU intensive. This patch improves the situation by
waiting about 30 ms in the CPU intensive mode, then stepping down to
using msleep(2) instead. In essence, we trade some granularity in
exchange for less CPU consumption when the waiting time is a bit longer.
As a result, PulseAudio should no longer be killed by the kernel
for taking up to much RT-prio CPU time. At least not for *this* reason.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 785f857d1c.
The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.
Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence. But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.
In short, the safest option now is to revert this affecting commit.
Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.
Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hdsp uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels.
Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Reviewed-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This assertion seems to imply that chip->dsp_code_to_load is a pointer.
It's actually an integer handle on the actual firmware, and 0 has no
special meaning.
The assertion prevents initialisation of a Darla20 card, but would also
affect other models. It seems it was introduced in commit dd7b254d.
ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting...
ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000
ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware...
ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20
------------[ cut here ]------------
WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]()
Hardware name: Dell DM051
BUG? (!chip->dsp_code_to_load || !chip->comm_page)
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
