Change AudioResampler's source quality.

Original commit.
commit ac6020508acedd316391dee42329040bf45f8d90
change ID ac602050
Some part of commit messages 'Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1, and uses low quality for all other use cases.'

But there are no codes for enabling 'Very high quality'.
So, I changed DEFAULT_QUALITY to VERY_HIGH_QUALITY to follow the intent of the author.

Change-Id: Ia0d081c96ff22b2a027452f969f6fa88402c0440
screenrecord: Raise time limit to 30 minutes
Merge branch '4.4.1' into HEAD

Conflicts:
	cmds/screenrecord/screenrecord.cpp
	services/audioflinger/AudioMixer.cpp

Change-Id: I791fb9021030c9c55073d13014c77e259da5adf0
Merge remote-tracking branch 'cm/cm-11.0' into 4.4-rb
FLAC coded file does not play (single metadata block files)

FLAC files which has the last metadata block flag set in
STREAMINFO were not playing. Sniff of FLAC is modified to
accomodate this set flag.

Change-Id: I76f216cc2cde6e10958324afbc704fc3e5a0e2bf
Author: Gurudatta Bhakte <gurudattax.bhakte@intel.com>
Signed-off-by: Gurudatta Bhakte <gurudattax.bhakte@intel.com>
Signed-off-by: Shuo Gao <shuo.gao@intel.com>
Signed-off-by: Bruce Beare <bruce.j.beare@intel.com>
Signed-off-by: Jack Ren <jack.ren@intel.com>
Author-tracking-BZ: 32421
Increase kFastTrackMultiplier from 1 to 2

Change-Id: I158f147295eebcea96e4047d7618069bc48bdd7d
Revert "frameworks/av: rendering statistics for Stagefright."

This reverts commit 1a65f74eebe15e1533ebde98268a454a8b14f815.

Change-Id: I755e293705a72d90a4a855343439b625226c050b

Revert "libstagefright: adjust AV sync late margin based on property"

This reverts commit 14b0bb438a8b3291edd10fb068656e707a31cfb5.

Change-Id: Ibcc3573283282b3f52a813c789f8586965bbc088

Revert "libstagefright: post video events dynamically - calculate amount of time to delay video event after renderusing the formula: delay = (us between frames)-(us taken by event)-(lateness)"

This reverts commit 1ac2a4f83f7d14b3a4d9c4691a5d6bb62fcc5522.

Change-Id: I06b8023d45a2ea1497a87d5adf7bfa7e74dc0710
stagefright: add an extra buffer to ANW's BUfferQueue

Add an extra buffer to ensure video player has a free buffer to
dequeue rather than wait for the queued buffer to be displayed.
Enable logs to indicate number of buffers allocated.

NOTE: This is not needed for 30fps content on 60fps panel, But
making this change regardless for all framerates till root-cause
is identified in SF

Change-Id: Id1048d7a104b2a75613dda8b9cf166e59bc4bb04
libstagefright: Filewrite in seperate thread for single video track

For files with single video track, the filewrite() and OMXRead()
operations are happening in the samethread. It impacts the
recording performance.

Hence moving filewrite to seperate thread for files with single
video track.

CRs-Fixed: 574807

Change-Id: I3a9bf7869e197973a85783510254cf66b8d1d91b
stagefright: add rendering statistics

 - adds extra fps statistics
 - report via dumpsys
 - prints statistics before reset at the end of playback onto
   logcat
 - adds resume latency, from play_l to render
 - adds seek latency, from seek_l to render
 - enables using persist.debug.sf.statistics 1
 - prints all statistics when seek or resume is issued

Change-Id: I6fa34f3d0180c1b8bbad40da42bd241fb89e6866
libstagefright: print seek statistics even when paused

 - if seek is issued during paused state, SEEK_PREVIEW flag is set
   and statistics needs to be printed even if that happens.

Change-Id: Ida5040d2961bff4dfb8a53b50be01609aaac9976
CRs-Fixed: 512801
libstagefright: Fix intermittent pause during tunnel A-V playback

-Media time is used in onVideoEvent for A-V sync. In offload
 playback we only use realtime(time from DSP).
-Media time should be same as real time. If media time is not
 updated with the real time value A-V sync issues are seen.
-Update media time to real time value to avoid intermittent pause
 during playback.

CRs-Fixed: 572060
Change-Id: Ib1123b7554f3508230ca67cd20a68fa32c753057
stagefright: Fix "add an extra buffer to ANW's BUfferQueue"

OMAP devices, such as maguro, don't work properly with an extra buffer.
This fixes the commit 2bce3e06f60a8d60c81261b0438d0cc6f2bbe7d1.

Change-Id: Id78442831e95b6021ea8b6d3648678db094f4565
Revert extra buffer hack.

 * Breaks too many things, and this is an awful hack anyway.

Revert "stagefright: Fix "add an extra buffer to ANW's BUfferQueue""

This reverts commit dbb2bb90acd47a53d84809b1aa150bac6093f752.

Revert "stagefright: add an extra buffer to ANW's BUfferQueue"

This reverts commit 2bce3e06f60a8d60c81261b0438d0cc6f2bbe7d1.

Change-Id: I5209858d09eb40581d740df500dee016b80b531f
audio: Modified API for compress voip formats

 - Modified API for supported compress voip formats.

Change-Id: I0165fc48bbe631d3febf2104e8a28708bf85ef23
audio: do not invalidate stream used by voip

During the process of invalidation of voip streams,
actual driver is not closed so when it tries to open again
getoutput fails as only one instance of voip can be created.
so do not invalidate the tracks used by voip driver.
CRs-Fixed: 494093

Change-Id: I42a962991ce1d586d87f4aecba19c0e5dde6b9dd
audio: Fix for OMXCodec::read() getting blocked

- Sometimes notification sound is not heard if it is played
  and stopped multiple times in a short duration.
- If OMXCodec::read() gets blocked for filled buffers while
  in pause, it never returns as mFilledBuffers is always empty.
  Due to this AudioTrack destructor never exits as it waits for
  the AudioCallback thread to unblock it.
- Ensure OMXCodec::read() returns even if timeout occurs while
  in pause and handle the same in AudioPlayer by returning zero
  bytes in AudioCallback.
CRs-Fixed: 478768

Change-Id: I2ca40a673fdf95bccc3069a18b694670612a9895
 audio: Fix framecount calculation in AudioFlinger

 -Sharedblock implementation expects the framecount to
  be round up to the next highest power of 2. This is
  not done in TrackBase for non-voip case which leads
  to shared block getting corrupted.
 -Fix is to round up the framecount in TrackBase for
  all the audio formats.

  CRs-Fixed:  571279, 573735

Change-Id: I0186bb0e04a080100b364aead447ff2fce495ca2
Audio: Tunnel Exception for AAC main profile format

- Audio is not decoding properly for some mp4 content

- ADSP audio decoders don't support aac main profile

- Added tunnel exception for AAC main profile format

Change-Id: I8470457492e32d94ab36eea7b97f4e4ec8776bb4
NuPlayer: Check if video input format is NULL

Adds null check on querying for video input format metadata.
CRs-Fixed: 582533

Change-Id: I3832de56f67bf8e6b4abb426c6e539dd66009fa5