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Andreas Huber [Fri, 8 Oct 2010 22:21:08 +0000 (15:21 -0700)]
RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams.
Change-Id: Ie61230cd60dd6c682baf72529100369ad6291189
related-to-bug:
3073955
Andreas Huber [Fri, 8 Oct 2010 19:06:27 +0000 (12:06 -0700)]
Refactor some more h.264 utility code out into avc_utils. Work around a hardware decoder issue by making sure the first access unit submitted to a decoder at startup or after seek is an IDR.
Change-Id: I61936601e55df7e4c23a8c13087579a4f85bd6e6
Andreas Huber [Fri, 8 Oct 2010 18:16:07 +0000 (11:16 -0700)]
Merge "Disable 10secs forward/backward seeking for rtsp as seek is a very expensive operation there. Decouple the 10sec forward/backward button functionality from seekbar functionality." into gingerbread
Andreas Huber [Fri, 8 Oct 2010 17:59:52 +0000 (10:59 -0700)]
Merge "Respect the cropping rectangle when extracting the video dimensions from the sequence parameter set of H.264 streams." into gingerbread
Andreas Huber [Fri, 8 Oct 2010 17:38:39 +0000 (10:38 -0700)]
Respect the cropping rectangle when extracting the video dimensions from the sequence parameter set of H.264 streams.
Change-Id: Id7f21d5ab8533b4e17ceaa8619a39c27f0306150
Andreas Huber [Fri, 8 Oct 2010 17:16:24 +0000 (10:16 -0700)]
Disable 10secs forward/backward seeking for rtsp as seek is a very expensive operation there. Decouple the 10sec forward/backward button functionality from seekbar functionality.
Change-Id: I016e79b688774f8ee91ac53216197b5fb9cb41b2
related-to-bug:
3073955
James Dong [Fri, 8 Oct 2010 17:06:22 +0000 (10:06 -0700)]
Merge "Name the writer threads" into gingerbread
Andreas Huber [Fri, 8 Oct 2010 17:01:37 +0000 (10:01 -0700)]
Merge "Disable the access unit timeout temporarily while a seek operation is in progress." into gingerbread
Andreas Huber [Fri, 8 Oct 2010 16:04:25 +0000 (09:04 -0700)]
Disable the access unit timeout temporarily while a seek operation is in progress.
Change-Id: I116cb76342aae4168f34ebae49ecb2301702a0ea
related-to-bug:
3073955
Eric Laurent [Fri, 8 Oct 2010 15:28:08 +0000 (08:28 -0700)]
Merge "Added getter for session Id to AudioSink" into gingerbread
James Dong [Fri, 8 Oct 2010 03:20:59 +0000 (20:20 -0700)]
Name the writer threads
Change-Id: I51461c3800ac5850e21ff398e80eb20b562264b3
Eric Laurent [Fri, 8 Oct 2010 01:23:03 +0000 (18:23 -0700)]
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
Andreas Huber [Thu, 7 Oct 2010 23:48:50 +0000 (16:48 -0700)]
Increase scratch buffers sizes in mp3 software decoder, this integrates a PV master/opensource patch.
Change-Id: I5a637f1b380e44c94040ec507843d58a1f5a9b61
related-to-bug:
3065605
Andreas Huber [Wed, 8 Sep 2010 21:32:20 +0000 (14:32 -0700)]
Work to support switching transport streams mid-stream and signalling discontinuities to the decoder.
Change-Id: I7150e5e7342e1117c524856b204aadcb763e06ed
related-to-bug:
2368598
Andreas Huber [Wed, 6 Oct 2010 23:43:57 +0000 (16:43 -0700)]
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.
Change-Id: I1b8fe68c1766299844fe84ebbff49cb8b3e4cc7c
related-to-bug:
3070094
Eric Laurent [Wed, 6 Oct 2010 00:06:16 +0000 (17:06 -0700)]
Merge "Fixed AudioFlinger not always pausing tracks" into gingerbread
Eric Laurent [Tue, 5 Oct 2010 21:41:42 +0000 (14:41 -0700)]
Fixed AudioFlinger not always pausing tracks
If the pause request is received before the AudioTrack buffer was
completelly filled and the track ready for mixing, the pause is
not executed: the track just underruns and stays in pausing state.
The fix consists in considering the track ready for mixing immediately
if pausing.
Change-Id: Ia6cb4703fee2126e41011a6400ea8eeb3a3e5456
Andreas Huber [Tue, 5 Oct 2010 17:25:34 +0000 (10:25 -0700)]
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.
Change-Id: Id574a0203efcb5e565f1b0fe77869fc33b9a9d56
Mathias Agopian [Fri, 1 Oct 2010 23:22:41 +0000 (16:22 -0700)]
fix [
2835280] Add support for cancelling buffers to ANativeWindow
There is a new ANativeWindow::cancelBuffer() API that can be used to
cancel any dequeued buffer, BEFORE it's been enqueued. The buffer is
returned to the list of availlable buffers. dequeue and cancel are not
mutually thread safe, they must be called from the same thread or
external synchronization must be used.
Change-Id: I86cc7985bace8b6a93ad2c75d2bef5c3c2cb4d61
James Dong [Mon, 4 Oct 2010 23:41:53 +0000 (16:41 -0700)]
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
James Dong [Mon, 4 Oct 2010 22:35:40 +0000 (15:35 -0700)]
Merge "Resilent media time stamp adjustment" into gingerbread
James Dong [Sun, 3 Oct 2010 17:59:26 +0000 (10:59 -0700)]
Resilent media time stamp adjustment
Change-Id: I13ab87c05f26bb11a3cc9bf8559f98e6ea0752db
Andreas Huber [Mon, 4 Oct 2010 18:36:39 +0000 (11:36 -0700)]
Make sure to finish the preparation phase even EOS occurs before we consider the cache to be completely filled up.
Change-Id: I29143e357fb6ea7b860636100e010f2ea7436798
related-to-bug:
3037389
Andreas Huber [Mon, 4 Oct 2010 18:09:31 +0000 (11:09 -0700)]
Don't retrieve metadata unless necessary for ogg-vorbis ringtone auto-looping.
Change-Id: Iaf5880bb3376f9cbf22aefe198878eaf6f3f08c7
related-to-bug:
3037389
James Dong [Sat, 2 Oct 2010 00:27:51 +0000 (17:27 -0700)]
Merge "Turn off media time adjustment by default" into gingerbread
James Dong [Sat, 2 Oct 2010 00:14:23 +0000 (17:14 -0700)]
Turn off media time adjustment by default
Change-Id: I1f8021d605d0fd896e0639607a84e3f7c459612e
Andreas Huber [Fri, 1 Oct 2010 18:35:17 +0000 (11:35 -0700)]
Merge "Start playing live streams from the start, no the middle..." into gingerbread
Andreas Huber [Fri, 1 Oct 2010 18:28:44 +0000 (11:28 -0700)]
Start playing live streams from the start, no the middle...
Change-Id: Ie01ba1250b51155cb1fb32fc3340189a16c01476
related-to-bug:
2368598
Andreas Huber [Fri, 1 Oct 2010 18:19:26 +0000 (11:19 -0700)]
Remove development-only code.
Change-Id: Ic2ca0efb631eb779ca157fb01b02aa19a1222c06
related-to-bug:
2368598
Andreas Huber [Fri, 1 Oct 2010 17:51:41 +0000 (10:51 -0700)]
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
Eric Laurent [Thu, 30 Sep 2010 23:12:31 +0000 (16:12 -0700)]
Issue
3032913: improve AudioTrack recovery time
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
Chia-chi Yeh [Wed, 29 Sep 2010 04:11:27 +0000 (21:11 -0700)]
Merge "AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead." into gingerbread
Chia-chi Yeh [Wed, 29 Sep 2010 03:41:14 +0000 (11:41 +0800)]
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
Eric Laurent [Tue, 28 Sep 2010 21:09:57 +0000 (14:09 -0700)]
Fix several audio effects problems.
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
Andreas Huber [Tue, 28 Sep 2010 20:23:26 +0000 (13:23 -0700)]
Merge "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." into gingerbread
Andreas Huber [Tue, 28 Sep 2010 20:13:38 +0000 (13:13 -0700)]
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.
Change-Id: I4909fdf19518dbabb6c340e2a31b50dfe6c5b067
related-to-bug:
3029947
Andreas Huber [Tue, 28 Sep 2010 19:02:35 +0000 (12:02 -0700)]
Merge "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." into gingerbread
Andreas Huber [Tue, 28 Sep 2010 18:56:39 +0000 (11:56 -0700)]
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.
Change-Id: I43875b6adaf96d4e982ef3dfc3d6c8f7034ac51d
related-to-bug:
3036592
Andreas Huber [Tue, 28 Sep 2010 18:46:41 +0000 (11:46 -0700)]
Merge "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." into gingerbread
Andreas Huber [Tue, 28 Sep 2010 18:27:33 +0000 (11:27 -0700)]
Merge "Squashed commit of the following:" into gingerbread
Andreas Huber [Tue, 28 Sep 2010 16:16:21 +0000 (09:16 -0700)]
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.
related-to-bug:
3036592
Change-Id: Ib142b171c829ed74156c0281d9d4543fcc96c802
Dave Sparks [Mon, 27 Sep 2010 23:55:18 +0000 (16:55 -0700)]
Fix media.player dumpsys to output open/mapped files correctly. Bug
2866669.
The dump function whitelists several directories as filters to the output.
The mount point changed for SD card in Froyo, and we started filtering files
that were open on the SD card. This fix changes the filter for the SD card,
and adds the directory for data files as well.
Change-Id: I61b67c3d11d93dbd530e8b3566000e79bc037137
Andreas Huber [Mon, 27 Sep 2010 19:04:43 +0000 (12:04 -0700)]
Squashed commit of the following:
commit
29a4d3effb05a2e074cb0693316ab1977baeb0b6
Author: Andreas Huber <andih@google.com>
Date: Mon Sep 27 12:01:32 2010 -0700
Fully working implementation of MPEG2TSWriter (for AAC and AVC sources).
Change-Id: I8a32a47565b647bf6c078c520e39565e08ea0d84
commit
f4dec4c3899f3be393508e180d6c07e249d3335e
Author: Andreas Huber <andih@google.com>
Date: Mon Sep 27 10:36:31 2010 -0700
More reliable identification of MPEG2 transport streams. Don't keep scanning forever in case the stream does not have both audio and video tracks.
Change-Id: Icc5b4e8be145b2805e8776559546a6818342aea7
commit
4fe3cc942f9b3d3cf54138b828c41214aa916dd2
Author: Andreas Huber <andih@google.com>
Date: Mon Sep 27 08:23:39 2010 -0700
test code
Change-Id: I16560a17661407d06497f99ff88230724bb898af
commit
64d988b24f49f179a90fa677be11c823959e734b
Author: Andreas Huber <andih@google.com>
Date: Thu Sep 23 14:42:52 2010 -0700
First shot at supporting writing to an MPEG2 transport stream.
Change-Id: Ie537939a99fa3ddc0c7661c47c18277584817c74
Change-Id: If78fd034af8f6e8ceac8dbeff96d5ecb3f6b96dc
Wu-cheng Li [Fri, 24 Sep 2010 00:17:43 +0000 (17:17 -0700)]
The old overlay should be destroyed if orientation changes.
Previously the orientation was wrong after suspend and resume.
When the camera app is resumed behide the lock screen, it
orinteation is portrait. When users slide to unlock the screen,
surfaceChanged is called and the orientation is landscape.
The camera app stops the preview, sets the display orientation,
and starts the preview. Overlay should be destroyed if the
orientation has changed.
bug:
3031640
Change-Id: I38b527f9ea78c91b538463292152c023383b4695
James Dong [Mon, 27 Sep 2010 17:08:52 +0000 (10:08 -0700)]
Merge "Fix track duration calculation if the start timestamp is non-zero" into gingerbread
James Dong [Fri, 24 Sep 2010 17:01:29 +0000 (10:01 -0700)]
Fix track duration calculation if the start timestamp is non-zero
o Updated the comments in the patch
o Added some additional checks on the timestamp
Change-Id: I8ad81eb6bfe358c1db5245cbb52efc905cdc234c
Andreas Huber [Mon, 27 Sep 2010 16:20:58 +0000 (09:20 -0700)]
Merge "Support other kinds of HTTP redirect in NuHTTPDataSource" into gingerbread
Andreas Huber [Mon, 27 Sep 2010 15:17:40 +0000 (08:17 -0700)]
Support other kinds of HTTP redirect in NuHTTPDataSource
Change-Id: I3268a94cfb48cee9caa8914804d7907ff91d8a8e
related-to-bug:
3037686
Eric Laurent [Fri, 24 Sep 2010 18:52:04 +0000 (11:52 -0700)]
Fix error in AudioEffect command status reporting.
AudioEffect::command() should not interpret the status
returned in pReplyData unless the command is ENABLE or
DISABLE.
Also fixed:
- bug in Visualizer::getWAveForm() status reporting.
- wrong initialization value for capture buffer in visualizer engine.
Change-Id: I60fb0011f9dba96ee74661dec1e8ef441edcaf35
Andreas Huber [Fri, 24 Sep 2010 17:44:01 +0000 (10:44 -0700)]
Merge "Instead of just writing one track to the .mp4 file, write all of them (at most 1 video and 1 audio track). Also support httplive URIs in the stagefright commandline tool." into gingerbread
Andreas Huber [Fri, 24 Sep 2010 16:35:49 +0000 (09:35 -0700)]
Instead of just writing one track to the .mp4 file, write all of them (at most 1 video and 1 audio track). Also support httplive URIs in the stagefright commandline tool.
Change-Id: I67b287cf76546191f398939aabe4480d9e79a948
Eric Laurent [Fri, 24 Sep 2010 16:43:24 +0000 (09:43 -0700)]
Merge "Fix issue
3007862" into gingerbread
Eric Laurent [Thu, 23 Sep 2010 23:10:16 +0000 (16:10 -0700)]
Fix issue
3007862
Removed a cross deadlock condition between audioflinger and audio policy
service mutexes.
Audioflinger::createEffect() locks audioflinger mutex and then calls
AudioSystem::getOutputForEffect() which ends up in
AudioPolicyService::getOutputForEffect() which locks audio policy service
mutex. If at the same time, the command thread in audio policy service is
processing a command(set volume, set route...), the mutex is locked and the
command will call one audioflinger method which in turn will attempt to
lock audioflinger mutex.
The fix consists in releasing audioflinger mutex before calling
getOutputForEffect().
Change-Id: Id44e7feb36e0a295731f6aa97cf32d022edd34d0
Andreas Huber [Thu, 23 Sep 2010 22:01:30 +0000 (15:01 -0700)]
Proper sync-frame detection for sources that don't already provide it.
Change-Id: Iafeee847b015c5cf6bec6dd5fac22a8d3e8cb78e
Andreas Huber [Thu, 23 Sep 2010 21:16:48 +0000 (14:16 -0700)]
Merge "Depending on our preference to write 2-byte or 4-byte NALs, patch the codec specific data 'avcC' accordingly." into gingerbread
Andreas Huber [Thu, 23 Sep 2010 21:08:32 +0000 (14:08 -0700)]
Depending on our preference to write 2-byte or 4-byte NALs, patch the codec specific data 'avcC' accordingly.
Change-Id: I1423923fafbcac6911218277e2bbfa068e49dcc6
James Dong [Thu, 23 Sep 2010 00:37:42 +0000 (17:37 -0700)]
Ignore errors from correction parameter query and config for M4v and H263 encoders
This feature is not essential for camcorder application
Change-Id: Id560632207dcf24fe361280ebe8178ea7332f22d
James Dong [Thu, 23 Sep 2010 00:42:09 +0000 (17:42 -0700)]
Use the advertised profile and level from M4V and H263 video encoders
This is already the case for AVC video encoder
Change-Id: I9cc97a03ab457b4bd0b21832c09f6d0aa7f5fc39
Eric Laurent [Wed, 22 Sep 2010 21:17:38 +0000 (14:17 -0700)]
Request permission for global audio effects.
Applications creating an audio effect on the output mix must
have the MODIFY_AUDIO_SETTINGS permission.
Change-Id: I57d88533f91ad0d33680107d79abcec28f7263b5
Andreas Huber [Wed, 22 Sep 2010 17:12:14 +0000 (10:12 -0700)]
Raise the amount of memory set aside for omx buffer allocations in the test harness to accomodate the new requirements of some codecs.
Change-Id: I73974a336852974e2fb95a9626c1a96f296449eb
Andreas Huber [Wed, 22 Sep 2010 18:47:00 +0000 (11:47 -0700)]
Make sure we drain the avc software decoder's output queue once we run out of input data.
Change-Id: I3d92ebae4951b618a6dfcf42936cc4c63d4de539
Eric Laurent [Tue, 21 Sep 2010 22:55:17 +0000 (15:55 -0700)]
Merge "Fix issue
2913071." into gingerbread
Andreas Huber [Tue, 21 Sep 2010 22:17:04 +0000 (15:17 -0700)]
Merge "This log message is codec specific." into gingerbread
Andreas Huber [Tue, 21 Sep 2010 22:14:43 +0000 (15:14 -0700)]
Merge "Remove stagefright foundation's incompatible logging interface and update callsites." into gingerbread
Andreas Huber [Tue, 21 Sep 2010 20:13:15 +0000 (13:13 -0700)]
Remove stagefright foundation's incompatible logging interface and update callsites.
Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
Eric Laurent [Tue, 21 Sep 2010 21:52:01 +0000 (14:52 -0700)]
Fix issue
2913071.
Scale audio signal during capture according to peak level so that
returned values on 8 bits contain enough information even for weak
signals.
Also do not reject requests to enable/disable the visualizer if we are
already in the requested state.
Change-Id: I07a705619764350834e61f82d161761eab688747
Andreas Huber [Tue, 21 Sep 2010 22:08:52 +0000 (15:08 -0700)]
This log message is codec specific.
Change-Id: I603ae7bd13b186a6df0a26ea9abe073e89276caf
James Dong [Tue, 21 Sep 2010 18:34:26 +0000 (11:34 -0700)]
Merge "Allow record to set input color format as a command line option" into gingerbread
James Dong [Tue, 21 Sep 2010 17:19:22 +0000 (10:19 -0700)]
Merge "Another attempt for fixing AAC+/eAAC+ related issue" into gingerbread
Wu-cheng Li [Mon, 20 Sep 2010 23:15:32 +0000 (16:15 -0700)]
Rename FOCUS_MODE_CONTINUOUS to FOCUS_MODE_CONTINUOUS_VIDEO.
This constant is not public yet. Continuous autofocus should
behave differently in still camera and camcorder. In camcorder,
lens movement may be more smooth. And the triggers to start a
new focus search may be different. If there is a need,
FOCUS_MODE_CONTINUOUS_PHOTO can be added in the future.
Change-Id: I05df9e491aca37829be3df92a73b952f26c86a4a
James Dong [Sat, 18 Sep 2010 03:50:07 +0000 (20:50 -0700)]
Another attempt for fixing AAC+/eAAC+ related issue
The decoder has to decode the very first two frames (decoder specific data + first data
frame) in order to be really sure whether a stream is AAC or AAC+/eAAC+.
bug -
2966336
Change-Id: Ib12a07285933df0d6c346370a12f70e99e811881
James Dong [Sat, 18 Sep 2010 23:13:57 +0000 (16:13 -0700)]
Allow record to set input color format as a command line option
o Only YUV420SP and YUV420P are supported
Change-Id: I425e8b29d9dca8c05066c1cd777f8bdede52964a
Marco Nelissen [Fri, 17 Sep 2010 22:04:01 +0000 (15:04 -0700)]
Add a check to track a problem the monkey script has been triggering.
Change-Id: If3896946d993cbab2442e17740f18efd7e753a3b
Andreas Huber [Fri, 17 Sep 2010 18:49:39 +0000 (11:49 -0700)]
Make sure the message dispatcher stays around until after OMX_FreeHandle is finished in case it posts some more messages during shutdown. Clear the source as soon as possible in OMXCodec's destructor.
Change-Id: I9c896cf07dea0c3201b6f074dbaf27e6d85cd784
Andreas Huber [Thu, 16 Sep 2010 18:23:09 +0000 (11:23 -0700)]
Register the new OMX components.
Change-Id: I795287c63073ca86ad0f97d6afcfc85c401dc984
Andreas Huber [Thu, 16 Sep 2010 22:20:33 +0000 (15:20 -0700)]
Merge "Make sure the .wav extractor does not read data outside the bounds of the 'data' box." into gingerbread
Andreas Huber [Thu, 16 Sep 2010 22:20:21 +0000 (15:20 -0700)]
Merge "Make sure stagefright -o terminates even if we're using a raw audio source (such as .wav pcm)" into gingerbread
Andreas Huber [Thu, 16 Sep 2010 21:55:48 +0000 (14:55 -0700)]
Make sure stagefright -o terminates even if we're using a raw audio source (such as .wav pcm)
Change-Id: Iec809df89e6b75706cf5b1ce21006c2e217e018f
Andreas Huber [Thu, 16 Sep 2010 21:54:10 +0000 (14:54 -0700)]
Make sure the .wav extractor does not read data outside the bounds of the 'data' box.
Change-Id: Icf18f9224d97e6a78328dd429ebc3a3433e5cecd
related-to-bug:
3007790
James Dong [Thu, 16 Sep 2010 18:45:01 +0000 (11:45 -0700)]
Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread
James Dong [Thu, 16 Sep 2010 04:07:52 +0000 (21:07 -0700)]
Fixed a bug in the query to the supported profiles and levels
According to OMX spec, the levels returned is the max level settings.
In fact, we could not enum all the levels.
Change-Id: Ib1cba74100512800a5761c7567894c7ea5b5a452
Andreas Huber [Thu, 16 Sep 2010 17:25:34 +0000 (10:25 -0700)]
Sometimes the avc software decoder will signal that a frame is ready but then unexpectedly fail to return the frame... stop asserting on that and return an error instead.
Change-Id: I6c0782b51b57e9be0fd410bd34078dc3c2b088d6
related-to-bug:
3005156
Andreas Huber [Wed, 15 Sep 2010 23:20:42 +0000 (16:20 -0700)]
A ThreadedSource wraps around an existing MediaSource and reads output buffers on a separate thread. It's now used for the vpx decoder to decode frames ahead of time to improve playback performance.
Change-Id: I57a798b00adeb2c8056e85aab29a2b57aef00b63
Eric Laurent [Thu, 16 Sep 2010 16:04:37 +0000 (09:04 -0700)]
Merge "Fix problem in lvm effect bundle wrapper" into gingerbread
Eric Laurent [Thu, 16 Sep 2010 01:29:49 +0000 (18:29 -0700)]
Fix problem in lvm effect bundle wrapper
When an effect was deleted while enabled, the count of enabled effects
in the bundle was not decremented. Resulted a state where if another effect
was still enabled, its process function was called only every other time.
Change-Id: I59a0c5b7f50f416a9ecadae297b42912e56ddd51
Andreas Huber [Wed, 15 Sep 2010 23:23:06 +0000 (16:23 -0700)]
Merge "Upgrade to the latest .webm project code." into gingerbread
Andreas Huber [Wed, 15 Sep 2010 22:13:16 +0000 (15:13 -0700)]
Upgrade to the latest .webm project code.
Change-Id: I645eff9b7c10bf33144977e6093663407af402e2
James Dong [Wed, 15 Sep 2010 21:51:57 +0000 (14:51 -0700)]
Merge "Add some explicit error log messages" into gingerbread
James Dong [Wed, 15 Sep 2010 21:51:49 +0000 (14:51 -0700)]
Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread
Andreas Huber [Wed, 15 Sep 2010 18:25:40 +0000 (11:25 -0700)]
Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread
Andreas Huber [Wed, 15 Sep 2010 18:18:13 +0000 (11:18 -0700)]
Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting.
Change-Id: Idbec5996ed0675c70e911b9c0514961fea099fb4
James Dong [Mon, 13 Sep 2010 23:30:51 +0000 (16:30 -0700)]
Add some explicit error log messages
Change-Id: I8a69157b75a67f0d3c19f05775a8cb0629232299
James Dong [Tue, 14 Sep 2010 18:48:11 +0000 (11:48 -0700)]
Fix audio input sample timestamp when audio driver loses audio samples
Change-Id: Ic0f1489f710929af50e7714867ae5153b3242dd8
Eric Laurent [Sat, 11 Sep 2010 00:44:44 +0000 (17:44 -0700)]
Fix volume problems with insert revert
- Use a constant input level to the reverb engine and implement volume control in the
insert reverb. This avoids the volume spikes when an effect that was inserted after
the reverb is disabled or removed.
- Fix clicks (one silent buffer) at the end of the reverb disable period.
- Modified volume management in audioflinger so that the volume ramp is also done by
the insert effect if present when the track is paused (avoids clicks).
- Increased room level for all presets.
Also fixed problems with output stage session (-1):
- effect bundle wrapper was not designed to support session -1
- the permission check in audioflinger for using session -1 failed due to a wrong usage of
getCallingPid()
Change-Id: Id1ff51327263364bf71d3f2668fa5cde4311d84f
Eric Laurent [Fri, 10 Sep 2010 17:24:02 +0000 (10:24 -0700)]
Merge "LVM release 1.09 delivery" into gingerbread
Andreas Huber [Thu, 9 Sep 2010 23:12:31 +0000 (16:12 -0700)]
TimedEventQueue now explicitly sets its scheduling policy to foreground as it should.
Change-Id: I630c9fb51686d87a4075f01a6d7f6f9139ddcb4b
related-to-bug:
2944452
Eric Laurent [Thu, 9 Sep 2010 19:01:11 +0000 (12:01 -0700)]
LVM release 1.09 delivery
- Reverb:
- 1 channel mixed has gain zero
- core componenet now only outputs STEREO samples
- wrapper now handles only STEREO from the Reverb Core
- Bass Boost:
- increase scratch memory
Change-Id: I00ae31051cc40a7006fc993420facfeb4d30dc52
Andreas Huber [Thu, 9 Sep 2010 17:13:26 +0000 (10:13 -0700)]
Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread
James Dong [Thu, 9 Sep 2010 17:13:08 +0000 (10:13 -0700)]
Merge "When 32-bit offset is used, if the requested max file size is greater than the 32-bit offset limit, set the limit to the max 32-bit offset limit." into gingerbread
Andreas Huber [Thu, 9 Sep 2010 17:10:15 +0000 (10:10 -0700)]
Instead of asserting return a runtime error if the maximum sample size cannot be determined.
Change-Id: Icf17ed04323f5415e0f9f1e4fd9f19ca60ce15ac
related-to-bug:
2602446