Changed type of decay time, reverb delay and reflections delay parameters
from signed to unsigned int to match OpenSL ES interface definition.
Also fixed some type casts in lvm reverb wrapper.
Change-Id: I5ca5e76a87c2590f01f031f3168355586ef22556
When the recorded file becomes large, the metadata size can
no longer be ignored. This makes it possible to save the
recorded file when the storage becomes almost full at the
end of the recording session.
Change-Id: Ief038080f825c9946ce550949c03e914aec1e31a
The problem was that the time to receive an output buffer
from an audio encoder is different because the encoder does not
need to read from the source for all output buffers. This leads
to large fluctuation in terms of wall clock duration between two
neighboring audio sample outputs from the audio encoder. As a
result, the media time for the video track after adjustment using
the drifting changes wildly sometimes.
This patch addresses this issue by only updating the media drift
time when an audio source input buffer is read. the wall clock
for the audio track is also calculated at the same time when
the input audio buffer is read at AudioSource.
bug - 2959800
Change-Id: I3174aa182f744784b540f0a7198524d4eee8bd7b
Modified lvm reverb wrapper code to expose a preset reverb interface.
Also removed debug log from bundle and reverb wrapper.
Change-Id: If9b95d91e25a6ff834decdfdda34b17df9b46967
o Make the API consistent with SF framework, which the MediaSource
provides a return status for stop
o Also, helps to convey errors that occurred right when a
premature stop() is called, leading to a potentially
mal-formed output file.
Change-Id: I52a932345f38570fdf8ea04d67d73dd94ccd30ef
This implementation uses fixed points instead of floating points. It
is slightly inaccurate compared to the old one but still perfect for
visualization purpose. It runs 40% faster on passion, 5 times faster
on sholes, and of course 14 times faster on sapphire.
Change-Id: I1e868417bcffda091becf106a7b941d02813faec
o Only do this for realtime applications
o Adjust other track clock based on audio clock
o Assume other track uses wall clock as the media clock
o Use some heuristics to reduce the size of stts box by 2/3.
- also
o Remove one unused key from MetaData.h
Change-Id: Ib9432842627b61795b533508158c25258a527332
- If the target color format is unavailable, the default
platform-dependent color format will be used.
- Also add some logic to prevent looping forever if the
omx component is buggy supporting color format enumeration.
Change-Id: I119a78f0d6201b4c3621235cca2f523ec14e24e3
+ This reduces the file I/O block time for audio/video track processing
- Since the file writer is buffering some output samples, the memory
usage would go up, depending on how many output samples are buffered.
Change-Id: I780cc5b26f4b53a5efbd643fcf9505dfc19cd4cd
commit 4abf16bb04dc9695fedf4007a84f903074312ccd
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 20 09:21:17 2010 -0700
Support a single format change at the beginning of audio playback. This way the AAC+ decoder may change its output format from what is originally encoded in the audio stream and we'll still play it back correctly.
Change-Id: Icc790122744745e9a88099788d4818ca1e265a82
related-to-bug: 2826841
commit 09c74da63e6ad5cb5dafb70f62696d75d2978967
Author: James Dong <jdong@google.com>
Date: Sun Jul 18 17:57:01 2010 -0700
Fix MPEG4Extractor to extract sampling frequency correctly when SBR is enabled.
Change-Id: I883c81dad3ea465e71cb5590e89d763671a90ff8
commit f672bf2a782dc7d5fb6325d611a7fe17045dfe9a
Author: James Dong <jdong@google.com>
Date: Thu Jul 8 20:56:13 2010 -0700
Enable the support for decoding audio with AAC+ and eAAC+ features
bug - 282684
Change-Id: I73c8377af3cc4edd3ee7cea86dc3b1c369fbd78b
Change-Id: I012f1179e933b6d1345d2368f357576c722485f7
The type of the cmd, cmdSize and *pReplySize parameters of the effect control interface command()
function have been modified from int to uint32_t. This is more consistent with their role.
Change-Id: I84d289fc262d6753747910f06f485597dfee6591
a single call.
Addional changes:
- Fixed simulator build
- Use effect interface UUIDs from OpenSL ES includes when available
- Added cleanspec rules to remove now obsolete test effect libraries
- Fixed bug in AudioEffect JNI setParameter function.
Change-Id: Ic25ddb135e2cec5a68c181d727321f5ac7a1ab6b
This is 1st part of the work to allow audio and video resync if
we found out that audio and video are out of sync during authoring
- also fixed a problem in AACEncoder::read() where the buffer acquired
from the buffer group does not release when error out at
reading from source.
Change-Id: I8a2740097fcfdf85e6178869afeb9f3687a99118
Added methods to AudioTrack and MediaPlayer java classes to enable use of
auxiliary audio effects. The effect can be attached and detached by specifying its
ID and the send level controlled.
Change-Id: Ie74ff54a453096a742688476f612ce355543b6f3
Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.
Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.
Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
- Fixed crash when deleting an effect chained before an effect having volume control
- Changed EFFECT_FLAG_VOLUME_CTRL to implicitely include EFFECT_FLAG_VOLUME_IND
(not need to set both in effect descriptor).
- Volume control changes from one effect to another if needed according to effect enable state
- EFFECT_CMD_SET_VOLUME is only sent when their is an actual change in volume
Change-Id: Ieebaf09157e2627366023569d95516646e03e26c
- Separate the updating of effect engine state from the process call in EffectModule so that the state
of all effects in the same effect chain is updated simultaneusly before all process functions are called.
- Added a mechanism for the effect engine to continue being called for processing after receiving the disable
commands untils it considers that the framework can stop calling the process function without causing
a glitch or loosing some effect tail.
- Updated test reverb and equalizer to support this new feature
Change-Id: Icb56ae2c84c076d4dbad6cf733b1a62f823febe7
