- Current implementation supports only fixed buffer size
of 320 bytes only for audio recording. This results in
performance overhead for recording at higher sample rate.
- Added support for flexible period size so that user can
use larger buffers for recording at higher sample rates.
Signed-off-by: SathishKumar Mani<smani@codeaurora.org>
Signed-off-by: Iliyan Malchev <malchev@google.com>
With the recent change in tabla shutdown, turning off the clocks
were being taken care after all the slimbus ports are closed.
But, there are instants where tabla startup is being called
during bootup, which keeps the runtime PM votes running,
and as there are no ports open, clocks are on all the time,
due to one of the votes. Bringing back tabla shutdown, but
turn off the clocks only if there are no slimbus ports
open at the time of shutdown, else it will taken care when
the slimbus ports are getting closed
Change-Id: Iaa9378b171d7c169a0f3306d55698e18d28dd111
CRs-fixed: 390003
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
Signed-off-by: Ajay Dudani <adudani@codeaurora.org>
- Increase default buffersize to 4k from 2k for audio playback
when HAL is configured with deep buffer output.
bug-id: 7129131
Signed-off-by: SathishKumar Mani <smani@codeaurora.org>
- Small buffersize is resulting in scheduling issues.
- Increase max buffer size so that user space can
configure large buffer size to avoid scheduling
issues.
bug-id: 6865729
Signed-off-by: SathishKumar Mani <smani@codeaurora.org>
- Speaker warm-up time is too long(over 100 ms) due to big delay
(16*2ms for mono and 16*4ms for stereo)after lineout PA enabled.
- Measured by "adb shell dumpsys media.audio_flinger |grep measuredWarmup",
speaker warm-up time is reduced to ~70-80ms
Signed-off-by: SathishKumar Mani <smani@codeaurora.org>
bug-id: 7022794
Update platform driver open function call to access
audio_client pointer only after allocation to avoid crash.
Signed-off-by: SathishKumar Mani <smani@codeaurora.org>
ALSA framework in kernel 3.4 requires all CPU DAIs to be routed to
the repective back-end input or output. Add the routing for STUB_1,
SLIMBUS_1, SLIMBUS_3, and SLIMBUS_4 CPU DAIs.
It is from QCT
Change-Id: Ia7b76ce04b4e19f2f0e9acf9886361e3d113cef6
- Add lowlatency pcm driver for Playback and Recording.
- Add support in target board files
- Add Recording Path to Multimedia5 FE DAI
- Add support in routing, platform, machine drivers
- Add low latency interfaces support in ASM and ADM drivers.
Change-Id: I1beb11db9010534e5aa91179ac6040a41622185d
Signed-off-by: Jayasena Sangaraboina <jsanga@codeaurora.org>
The source of input signal(ADC or DMIC) needs to configured correctly
in CDC_TXn_MUX_CTL registers for correct operation of Decimators.
Currently for DMIC's, this configuration is done in DMIC DAPM widget
power up/down call back creating dependency between DMIC number and
Decimator number (with current code, DMIC1 can send signal to only
DEC1, DMIC2 can send signal to only DEC2). Remove this dependency by
setting type of input signal when Decimator MUX input is set.
CRs-Fixed: 384279
Change-Id: Ic084bb892d663dea51ca5a5a95c6bdba30453744
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
The token field is for identification to determine response packet's
source. Fill this field correctly to address afe_loopback timeout.
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
If the vote for pm runtime is done while codec shutdown, it is possible
that the runtime pm vote occurs even before the slimbus port for tx/rx
audio channel is disconnected. This can cause problem in audio playback/
record. Fix by moving the vote for runtime pm after slimbus port has
been disconnected
Change-Id: I711bc5cfee5b832575ea0b91cf68e826f1a3c0f5
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Headphone path can either be in CLASS G mode or CLASS AB mode.
The class G mode in the codec hardware is used to adjust the
supply voltage of the PA's according to the signal level. This
makes the system as efficient as possible. Fix the register
sequence to enable headphone in class G mode
CRs-fixed: 380598
Change-Id: I110c4e0ea958ef55c0b407c566deb7da58f4d99a
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Upgrade the board file to change the microphone bias used by MBHC to
2.7 Volts. The button voltage ranges need to be updated accordingly
Change-Id: Ia2f91271864e0f8fe25b866bff8006af0dd6f20b
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
The debugfs TRRS entry is for overriding headset button polling.
Fix regression not to start button polling when this flag is set.
CRs-fixed: 386038
Change-Id: I469c366bc111f37ecbb46708d2800200dd3d7584
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Some of accessories on the headset jack takes longer to settle down
voltage. Make adjust time to be configurable so those accessories can
be detected correctly.
Change-Id: I3c2f68c8a4bb1a8f94669bd910728f014ee39874
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Read the codec specific data from device tree instead of board file.
Change-Id: Iad382b89692903d2434b63d34c7121fe0b4b9dda
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
Resetting this channel actual flag without
checking whether the set of channels are already in
use, could cause failures in disconnecting those
set of ports associated with that channels.
Change-Id: If0b917023c8f6d11d6b5cd92708715e10a1408ab
CRs-fixed: 384055
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
Update the number of supported voice networks
in the ACDB driver and change the voice driver
to support concurrent calibration for VOLTE,
VOIP & voice.
Change-Id: I945fa40cbb4ac079a79fa0cb829971f1aa9d07f6
Signed-off-by: Ben Romberger <bromberg@codeaurora.org>
When SRS Tru media topology is enabled the mixer controls set from
userspace are not sent to driver as the validation fails with
the missing field initialization in the structure. Update the same
to receive the SRS parameters from userspace
Change-Id: Ia2bf800a540adf3bcc2a99fc6e0d9b043c826272
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
This change sets rate on pcm0 root clock (pcm0_src_clk)
and pcm out enable (PCM_OE) clock for AUX PCM CPU DAI.
Once the rate is set on the source clocks, corresponding
branch clocks get enabled.
Change-Id: I59bc339d69fe836f613c1bebf7561184dced2dcf
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
After recent movement of afe_start to prepare the interrupts start
coming earlier from proxy port.
This results in proxy port driver not ready for interrupts. Start the
timer later to ensure processing of packet once trigger start happens.
Change-Id: Ie82d982b6c147afdef46a8c7bfe7234dd14486f1
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
If interrupt handler is not quicker than voltage ramp up by button
release, driver requested measurement won't see stable voltage.
Enhance button press detection performance by comparing voltage
measurements from codec hardware only.
CRs-fixed: 376825
Change-Id: I294239df02fb5afeb3527dda85924c06ab15e54c
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Enhance button release detection performance by adopting dynamic release
threshold adjusting logic.
Button release is now detected more quickly.
Change-Id: I3a2379e10663cf91df671e8a3894b8805d1ccf9c
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
copy_to_user and copy_from_user can block(go to sleep). These two functions
should be removed from spinlock protection.
Change-Id: I0c10796ece4ac218600ef3b214c88a06004a0a0d
CRs-fixed: 381839
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
If Mic Bias is high Z by default and when the LDOH is enabled,
the Mic Bias output can pull-up to a non-zero voltage if there
is no loading or if the load leakage is very small. Pulling
down the Mic Bias output so it is not in high Z to eliminate
the floating voltage.
Change-Id: I38e76a8c03107879727564f177b2713c9dfa4631
CRs-Fixed: 368898
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
The value for the amix command with the numeric strings is
the actual value not the order of the string in the array
of text so the put function has to be updated to handle it.
For example, the following amix comand:
amix 'Internal BTSCO SampleRatee' 16000'
The value passed from the alsa-lib is not 2, i.e., the 2nd
paramater in the command string:
static const char *btsco_rate_text[] = {"8000", "16000"};
Instead, it is 16000.
CRs-fixed: 364832
Change-Id: Ie7c83a460900b54e2b317e1c77a064efc22e6bcd
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
-Voice call recording is not working because of missing
backends configuration for incall record Tx/Rx.
Change-Id: Ie69112e2929ed98a5bc19164cf9a9c66d73cc8dc
Signed-off-by: Vidyakumar Athota <vathota@codeaurora.org>
Update DAI link to connect slimbus RX/TX port 0 to CPU.
Change-Id: Id8aa2ca92ea9b1e0d51b6d3ccf113860a9147c44
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Add audio OCMEM driver support to exercise On-Chip
Memory (OCMEM) for low power audio and voice. The
driver is implemented as standalone and it gets
exercised based on the usecase. Also, this design
reduces the latency associated with OCMEM handshaking
protocol. The audio OCMEM driver is enabled by default
with a Kconfig option using select. Add device tree node
for audio OCMEM to retrieve platform data.
Change-Id: Iba46ce675fc03843d88cd7cf2aa9bc92fe70a955
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Register slimbus CPU dai link to support slimbus data path.
Change-Id: I3584306ac1e0ad6561a19cecfe71f2a63aadafa9
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
During rapid open and close of slimbus ports, it may happen that when
port open is performed, the port is not disconnected from the previous
open. This will cause inconsistent state and may result in failure to
playback audio. Fix by waiting for the slimbus port disconnect to happen
before opening the port.
CRs-fixed: 375689
Change-Id: Id1303deae296eb6842074837183ab231aa2b4dad
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
In AFE-PCM RX and TX dai links of MPQ8064 machine driver there
was a conflict in codec dai name and codec name, with this ALSA
Framework is not creating the node in /dev/snd/
CRs-Fixed: 377509
Change-Id: I5337b216a3d0a2cdc36292ccdafe3e144e7f1d41
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
The wcd9310 codec driver already can detect and report unsupported headset
plugging. Create headset jack with unsupported headset mask to be able to
report via sound core.
Change-Id: I0119d01c039362cc7b185f9f3407d78c958bc49a
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
