Merge commit 'b37fcbfd7f9d65b90b43e0242903030c5a6517b5' into gingerbread-plus-aosp
* commit 'b37fcbfd7f9d65b90b43e0242903030c5a6517b5':
Added getter for session Id to AudioSink
Added a method to expose the audio session id at AudioSink interface
so that the AudioPlayer in stagefright can retrieve it.
Also:
- Fixed audio effect send level not being initialized in mediaplayer.
- Fixed compilation error when LOGV is enabled in mediaplayer JNI
Change-Id: I4bb55454fd63d646e0e677692d737c4843fb05fb
Merge commit '949f7d9066e09768e570686a5695aaba4a1dafd0' into gingerbread-plus-aosp
* commit '949f7d9066e09768e570686a5695aaba4a1dafd0':
Work to support switching transport streams mid-stream and signalling discontinuities to the decoder.
Merge commit '02654f01bc6bd2e581b4a1d2409ecea217294fa2' into gingerbread-plus-aosp
* commit '02654f01bc6bd2e581b4a1d2409ecea217294fa2':
On this particular device the hardware video decoder spits out buffers that don't actually contain our video data, so we cannot use them to restore the video frame after suspend/resume.
Merge commit '17bc4f65324a823598e7671256c815bf32ddcc95' into gingerbread-plus-aosp
* commit '17bc4f65324a823598e7671256c815bf32ddcc95':
Make sure to call AudioTrack::stop() instead of AudioTrack::pause() after submitting all samples to AudioTrack to make sure those remaining samples are actually played out.
The test was failing periodically due to too many files being open.
This change attempts to resolve the problem on the theory that
signaling pipe file descriptors are being leaked due to the large
number of Looper instances created during the test run.
However, it's still possible there are other leaks elsewhere.
Change-Id: I71f9f12d21605c47c9217c72c51e6c768142ce10
Merge commit '6f1c7bda39774fe3a1febf72b03c8ad481c1ea54' into gingerbread-plus-aosp
* commit '6f1c7bda39774fe3a1febf72b03c8ad481c1ea54':
Fixed an issue where the reserved free space in the file writer was larger than intended
The problem was that even though user does not explicitly request the max file size
limit via MediaRecorder.setMaxFileSize(), the file writer sets an implicit file
size limit if 32-bit file offset is used on user's behalf. The reserved free space
is estimated based on the file size, if the file size limit is set by the user.
The fix is to add an extra bool to tell the difference between an
explit requested file size and an implicit file limit and use that
to set the estimated moov box size accordingly.
Change-Id: I731aca6c7833aa764ed7b905edb77721577471b3
Merge commit 'b99f0c7eae365f788a21944fef28de580c7f5f92' into gingerbread-plus-aosp
* commit 'b99f0c7eae365f788a21944fef28de580c7f5f92':
Resilent media time stamp adjustment
Merge commit 'aa1a694dc78e5201fc83fe3d710a6e43eb62831a' into gingerbread-plus-aosp
* commit 'aa1a694dc78e5201fc83fe3d710a6e43eb62831a':
Make sure to finish the preparation phase even EOS occurs before we consider the cache to be completely filled up.
Merge commit '6c609b6a60e533a93c6d0088222bd8da209b9953' into gingerbread-plus-aosp
* commit '6c609b6a60e533a93c6d0088222bd8da209b9953':
Turn off media time adjustment by default
Merge commit 'e619a9da44e4c00f9034917aef67f86da0bc207f' into gingerbread-plus-aosp
* commit 'e619a9da44e4c00f9034917aef67f86da0bc207f':
Start playing live streams from the start, no the middle...
Merge commit 'd1398db35cb2e4d918fc631dda35cacb8540b187' into gingerbread-plus-aosp
* commit 'd1398db35cb2e4d918fc631dda35cacb8540b187':
Squashed commit of the following:
commit 46744c7697f29aec71aed8de3c95ce035c284d97
Author: Andreas Huber <andih@google.com>
Date: Thu Sep 30 16:44:57 2010 -0700
better separation of access units
Change-Id: I5a9e2138aed341f0bcf22cfe368a15ca5ea5a73c
commit d34952ac0feb1ae722ff65824d7353335502219b
Author: Andreas Huber <andih@google.com>
Date: Thu Sep 30 15:35:01 2010 -0700
Support for ES packets that do not start on PES packet boundaries.
Change-Id: I2cf012833948eddfb20b16a1901206cf22ce71e4
related-to-bug: 2368598
Change-Id: Ib9329bd6bb7149b5a6e2483788a96b1b158952fc
This issue showed that when an AudioTrack underruns during a too long period
of time and is therefore disabled by audioflinger mixer, it takes an additional
delay of up to 3 seconds to recover.
This fix adds a simple mechanism to recover immediately when the client application
is ready to write data again in the AudioTrack buffer
Also throttle warnings on record overflows
Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
Merge commit 'd6877fa4971710150de20453bf4ba54dca863429' into gingerbread-plus-aosp
* commit 'd6877fa4971710150de20453bf4ba54dca863429':
AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead.
This allows gsmamr_enc.h and gsmamr_dec.h to be included in the same
file without conflict definition.
Change-Id: I1c8dac235c122735ba14a1af2fda48c0f8d9f87b
Fixed the following issues in LVM effect bundle wrapper:
- memory leaks in EffectCreate() in case effect creation fails at various stages
- Added saturation when accumulating to output buffer
- Fixed problems with enabled effects count when an effect is released while enabled
- Do not allocate temporary buffer for accumulation each time process() is called
Fixed the following issues in effects framework (AudioFlinger)
- Release effect synchronously in the library when deleted from effect chain
- Do not call the effect process function if no tracks are present in the same
audio session
Change-Id: Ifbd80a163415cfb3c0a337c12082853ea45d9c91
Merge commit '88a995edcf3c371845cb32aed8bcddb7509bf875' into gingerbread-plus-aosp
* commit '88a995edcf3c371845cb32aed8bcddb7509bf875':
Properly flush the AudioTrack/AudioSink on a seek request and make sure that both the mp3 decoder and aac software decoders start fresh after a seek without any dependency on previously decoded content.
Merge commit 'be0450619528e65eebfa1d7eab78fde757d094cc' into gingerbread-plus-aosp
* commit 'be0450619528e65eebfa1d7eab78fde757d094cc':
Instead of constantly polling the AudioPlayer to see if it reached EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens.
Merge commit '4769f57948455277d0670ef18f64824ca5d894c1' into gingerbread-plus-aosp
* commit '4769f57948455277d0670ef18f64824ca5d894c1':
Vorbis files may have more samples encoded that should be used, i.e. we have to trim samples at the end of the stream. This is crucial for proper looping of some audio files.
