Add .gitignore files
Fix compilation error when CONFIG_SEC_DEBUG is disabled
Revert "android: logger: Allow a UID to read it's own log entries"

This reverts commit 1c8beb6c4e2b7382b6a79989d81e1b47b08efc62.
gpu: ion: Use correct type for variables

Use correct type for variables to prevent overflow of signed
type when assiging from an unsigned type.

Change-Id: I0055d35c7310d111de590db3798950fd98f1b298
Signed-off-by: Olav Haugan <ohaugan@codeaurora.org>
gpu: ion: enable the kmalloc heap

There are some Ion clients who would like to use the kmalloc
heap. Enable it.

Change-Id: Ie818be9e1211d52d489ad2cb63d94e2fddfc0f01
Signed-off-by: Mitchel Humpherys <mitchelh@codeaurora.org>
gpu: ion: Do fallback when allocating large sizes

The preferred allocation of memory for kernel data structures in
ION is to use kmalloc. This is mainly for performance reasons. However,
we are not guaranteed to be able to allocate large sizes. So to keep
the code performing well and at the same time allow for larger allocations,
implement a fallback mechanism for allocation of memory for ION kernel data
structures.

CRs-Fixed: 488521
Change-Id: I8c6a13ad74cce69f590920d28c01a58f7a540fd2
Signed-off-by: Olav Haugan <ohaugan@codeaurora.org>
[cpandya@codeaurora.org: resolved merge conflict]
Signed-off-by: Chintan Pandya <cpandya@codeaurora.org>
msm: kgsl: Squashed update to CAF's jb_2.5

* As of up to: msm: kgsl: Fix the extra draw packet on resume for A305
ASoC: Avoid putting stream state to STOP when stream paused

When multiple Front-end are using the same Back-end, putting
the stream state for a Front-end to STOP will make the the Backend
stream to be released, when the front-end is paused. This will avoid
the Backend to be released (when another active front-end stream is
present).
Instead, put the stream state to PAUSED, which will keep the Back-end
ON.

Change-Id: Ic31bf90a09b605b91f6b9fe3c09361b1145581fb
CRs-fixed: 428825
Signed-off-by: Banajit Goswami <bgoswa@codeaurora.org>
asoc: msm: Add flexible period count to pcm record driver

 In pcm recording, if buffers are not requested from dsp at real
 time then dsp drops the buffers and sends partial buffers. This
 is causing record failures. Pcm record platform driver has two
 periods. If system is loaded, read request will not be at real
 time. Add flexible period count to pcm record platform driver
 so client can configure the period count.

CRs-Fixed: 467746
Change-Id: Iadd47979f7b29a01fb8e6479383094a2ed1d93c5
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
ASoC: msm: Add support for external EC reference point

Add mux to sound card topology with output of mux routed
to CS voice uplink front-end as way to indicate external
EC reference point. Currently, MI2S_TX is the only
AFE input port defined for this MUX.

Change-Id: I0c81c579cc01bf92e743eda0e8dee15be2a8c01f
Signed-off-by: Venkat Sudhir <vsudhir@codeaurora.org>
ALSA: include: Add new ioctl for metadata mode

Add a new ioctl to support metadata mode in lpa driver. This ioctl
allows lpa driver to work in default alsa compliant mode and also
metadata mode.

CRs-Fixed: 458904
Change-Id: I8111e8652bbfb6dd93bdbda69bd16f53b45cb756
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
ASoC: msm: Add metadata mode changes to LPA driver

Playback hangs for three seconds on seeking to EOS. This is
because the lpa driver works on period size. Partial buffers
are not handled. Add metadata mode changes to LPA Driver to
support partial frames.

CRs-Fixed: 458904
Change-Id: Ieb89b59dbca4b40ec0915a286d53c4617c23c471
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
ASoC: audio: Fix BE not disconnected even codec teardown

Scenario:
When FE1(LowLatency) comes in which opens BE1 and puts it in state
SND_SOC_DPCM_STATE_START. Similarly FE2(LPA) comes in with
connecting to same BE. Now FE2 is in SND_SOC_DPCM_STATE_PREPARE
state and waits for TRIGGER_START. Meanwhile FE1 is closed which
doesn't affect BE1 and FE2 which are connected.
Also, for FE2 pcm_close is called before TRIGGER_START itself.
In such a case, BE1 is not getting disconnected (afe_close is
not called) while Codec is teared down resulting in Slimbus
overflow.

Fix:
Whenever BE1 is found to be in SND_SOC_DPCM_STATE_START state
and FE1 being connected is not started yet, and close sequence
happens for this FE, then allow hw_free to happen for BE1 if
this is the only FE connected to this BE1. This allows BE to be
disconnected which calls afe_close then followed by Codec teardown
not resulting in any Slimbus overflows.

CRs-Fixed: 468118
Change-Id: I03a80a12c1fd0cf09bd2e3dfe1f1eb56589016d3
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
ASoC: msm: flush if prior and current backends rate not matching

It is found that during device switch from one backend
with one sample rate to another backend with another sample rate,
the command to QDSP6 ADM which maps audio stream session to a
particular backend would not get carried out until pending
data of audio stream session from previous backend is either
read out or flushed. This scenario occurs when application
stops providing more buffers to retrieve captured data.
Remedy is to flush upon detection of rate mismatching

Change-Id: I2c01c036d9bb71f938a6795337f08948bd986553
CRs-fixed: 422205
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
ASoC: msm: Support independent left-right channel volume control

The current volume sent to DSP is averaged between left and
right channels. As a result with volume level as zero on one
channel and non zero on the other channel results to playback on
both channels. Add support to send left and right volumes to DSP

Change-Id: Icc5f182d138c841607ea356ec37a7f78b4ba770d
CRs-Fixed: 449284
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Signed-off-by: Jay Wang <jaywang@codeaurora.org>
ASoc: msm: Fix for pcm_read getting struck in alsa core.

Advance hardware pointer by period size even if the packet
coming from qdsp6 is of less size. QDSP6 sometimes
(for rapid device switches) gives packets worth less
than period size and this results in hw_ptr appl_ptr mismatch.

CRs-Fixed: 486276
CRs-Fixed: 488449
Change-Id: I893f9a52e4cd507fd72e4a7aba3c5fa76527d1bf
Signed-off-by: Sidipotu Ashok <sashok@codeaurora.org>
msm: audio: qdsp6v2: Add size safety check to ACDB driver

Check that the size sent by userspace is not larger
then the internal amount allowed. This protects
against overflowing the stack due to an invalid size.

Change-Id: I8230fdb00a7b57d398929e8ab0eb6587476f3db1
CRs-fixed: 470222
Signed-off-by: Ben Romberger <bromberg@codeaurora.org>
msm: audio: qdsp6v2: Fix command sequence issue in Non-tunnel player

The CMD_CLOSE and CMD_UNMAP needs to be sequentially sent to dsp.
Avoid sending CMD_UNMAP if CMD_CLOSE is not sent.

CRs-Fixed: 485051
Change-Id: Ic6dd9775b0aea218874161b8a44dc0286608616c
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
msm: audio: qdsp6v2: Fix for kernel crash during stability

- Userspace calling AYSN_READ during flush, READ_DONE ack for
  this is coming after FLUSH_DONE which is hitting BUG_ON as
  buffers are removed from in_queue/out_queue during flush.
- Fix is to avoid sending READ requests to DSP until flush
  is complete.

CRs-Fixed: 485288
Change-Id: Ia2a64471a3507b3f5565b695c40cc6dd9081930d
Signed-off-by: Deepa Madiregama <dmadireg@codeaurora.org>
msm: usbaudio: Fix memory allocation for USB audio module init

Fix memory allocation issue in snd_usb_audio_init()
use sizeof(struct) insead of sizeof(pointer)

Change-Id: I270bbd7111e97dd4e0e92378f61d5025e16f5fca
Signed-off-by: Alex Fradkin <afradkin@codeaurora.org>