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Author SHA1 Message Date
Andreas Huber
4b3d32bb1b TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. 
Change-Id: I630c9fb51686d87a4075f01a6d7f6f9139ddcb4b
related-to-bug: 2944452
 2010-09-09 16:14:02 -07:00
Andreas Huber
f3de053c0a Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread 2010-09-09 10:13:26 -07:00
James Dong
5c43a7af7b 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 2010-09-09 10:13:08 -07:00
Andreas Huber
d7f2225e74 Instead of asserting return a runtime error if the maximum sample size cannot be determined. 
Change-Id: Icf17ed04323f5415e0f9f1e4fd9f19ca60ce15ac
related-to-bug: 2602446
 2010-09-09 10:10:15 -07:00
Andreas Huber
3e0f2be7d6 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. 
Change-Id: I26db4524c5306bf2346438d2bd359c5cfb95cead
related-to-bug: 2900419
 2010-09-09 09:48:41 -07:00
James Dong
a4fb816bd5 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.

Change-Id: Ie74cbed98469721d4280a0b87491e888948f0046
 2010-09-08 17:56:11 -07:00
James Dong
d353c840ad Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread 2010-09-08 17:51:59 -07:00
James Dong
d015ccf62b HW audio encoder expects timestamp via kKeyTime from each input buffer 
- This fixes media server crashes on droid

Change-Id: I7191cadc5275107425ec3ee3d437b2c5295858dc
 2010-09-08 17:28:57 -07:00
Eric Laurent
95d5de0681 Modify type of some environmental reverb parameters 
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
 2010-09-08 16:06:18 -07:00
Eric Laurent
7e427934e6 Merge "LVM release 1.08 delivery." into gingerbread 2010-09-03 16:35:54 -07:00
James Dong
9077f8ec93 Merge "Not all audio source has the drift time information" into gingerbread 2010-09-03 15:42:09 -07:00
Eric Laurent
5fa6df6ebf LVM release 1.08 delivery. 
- Changed bundle SamplesToExit to 0.1 secs
- Added SamplesToExit to Revreb
- Removed mixer from Core reverb

Change-Id: I675ec22889f20ef35a0ac427600c2654111c397e
 2010-09-03 15:22:18 -07:00
Andreas Huber
9fee0b2a02 Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer's setLooping setting. 
Change-Id: Ifb564c6cdf6137eac14869f9ca7d471f05a5556a
related-to-bug: 2974691
 2010-09-03 14:31:50 -07:00
Andreas Huber
cc4a38c60f Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread 2010-09-03 13:46:02 -07:00
Andreas Huber
87ab9cdd0f Properly buffer a certain amount of data on streaming sources before finishing prepare(). 
Change-Id: I39bf3c6dafcbe003b51dea4795742dcd8548f207
related-to-bug: 2875110
 2010-09-03 13:44:42 -07:00
James Dong
3caa71483f Not all audio source has the drift time information 
Change-Id: I74e502376348ca4a6ffaa7492bed35c1355e7e62
 2010-09-03 12:01:55 -07:00
James Dong
7755cdd696 Remove unused/debugging code from MP4 file writer 
o also makes nal length in the recorded file modifiable at runtime

Change-Id: I731b4dde7070d8d9628b36b523a5b2c011c7c2cf
 2010-09-03 10:13:19 -07:00
James Dong
cb7e65c6cb Better file size estimate 
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
 2010-09-02 20:10:00 -07:00
James Dong
4c23815c39 Calculate audio media drift time from AudioSource 
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
 2010-09-01 20:45:39 -07:00
Andreas Huber
a2511da9d6 Merge "Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data." into gingerbread 2010-09-01 15:44:46 -07:00
James Dong
d3c1bae4eb Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread 2010-09-01 15:41:10 -07:00
Andreas Huber
4d8f66bce3 Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data. 
Change-Id: Ice8564e902e48c89c9c00f6651c5504b3c41fcad
related-to-bug: 2556656
 2010-09-01 15:05:28 -07:00
James Dong
a87544b35f Make sure that if initialization fails, AudioSource still behaves well. 
Change-Id: I16dfc90bcb8a324d6ee9a38a5a1a31cc094c820a
 2010-09-01 14:11:28 -07:00
Andreas Huber
6c33904ad9 Merge "Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds." into gingerbread 2010-09-01 13:54:12 -07:00
Andreas Huber
412fc7cdb6 Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread 2010-09-01 13:39:27 -07:00
Andreas Huber
8d7d413959 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. 
Change-Id: Ia9748691ba60d3c4b5fcaf319ed0b4493d69abc6
related-to-bug: 2963846
 2010-09-01 13:27:14 -07:00
Andreas Huber
4dcc6a1020 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. 
Change-Id: I15e21eae50beb6057024ea42a7e9bf3b8d8a0603
related-to-bug: 2368598
 2010-09-01 12:25:36 -07:00
Andreas Huber
27b9c8ec16 Keep gtalk video chat specific code consistent with rtsp changes. 
Change-Id: I5f3f46c2150e16b26674432e427f79c04a69cd8e
 2010-09-01 09:27:47 -07:00
Eric Laurent
a92ebfa1cd Audio Effects: fix problems in volume control. 
- Fixed click when re-enabling effect during the turn off phase:
make sure the effect states where effect is processed are the same
where volume control is delegated to effect.
- Fixed click when effect is deleted while still active: do not apply
volume ramp if an effect having volume control was just removed from the
effect chain.

Also fixed a crash when PCM dump is enabled in effect bundle wrapper.

Change-Id: Ib562f5cf75c69af75df0e862536262e2514493e4
 2010-08-31 15:26:23 -07:00
Andreas Huber
48ac68e1b1 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread 2010-08-31 14:54:37 -07:00
Andreas Huber
e536f800c6 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. 
Change-Id: Ied92ea8c2448a2cb1a732c72c21c69da1913dbc8
related-to-bug: 2556656
 2010-08-31 14:25:36 -07:00
Andreas Huber
3a48d4d726 Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer) 
Change-Id: I3c1ae79bb9342770e959ebdcdc6b748549b76330
related-to-bug: 2556656
 2010-08-31 11:13:51 -07:00
Chia-chi Yeh
12006013cc fixedfft: Only includes cpu-features.h when __arm__ is defined. 
Change-Id: Ifb6c03b38eff3c94a507ceb5043fcc48b364c25c
 2010-08-31 12:56:01 +08:00
Andreas Huber
68ae91cbd2 Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread 2010-08-30 16:12:46 -07:00
Andreas Huber
0ddf8c09f9 Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder. 
Change-Id: Ib8615ce5a89a9a846ee2f9f96cdfb23462f72c7a
 2010-08-30 16:08:03 -07:00
Andreas Huber
f88ca7a033 Instead of closing the connection altogether if no UDP packets arrive after a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection. 
Change-Id: Ie8d6a3865a0477e28d4b76bb9038e468451287b1
related-to-bug: 2556656
 2010-08-30 15:25:35 -07:00
Andreas Huber
681c5ff208 Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore." into gingerbread 2010-08-30 13:04:21 -07:00
Andreas Huber
30cfa20dfc Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. 
Change-Id: I1ca6bd8faba0185f9694f9dc04d2b3e6a7ab5ac3
related-to-bug: 2370115
 2010-08-30 12:46:12 -07:00
Eric Laurent
858bb4f66e Merge "LVM release 1.07 delivery." into gingerbread 2010-08-30 11:39:34 -07:00
Andreas Huber
f6639c46e8 Finetune some rtsp timeout constants. 
Change-Id: Ice731c5097c2a2dee8a7f0cd45b547cd34f532c6
 2010-08-30 10:35:56 -07:00
Andreas Huber
df992ac9cc Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread 2010-08-30 10:28:24 -07:00
Andreas Huber
c4e0b70a21 ALoopers can now be named (useful to distinguish threads). 
Change-Id: Ieabaddb2e3a9e3a7a5bc36e55cd0721b60dbd50e
 2010-08-27 15:21:07 -07:00
James Dong
90862e2a8b Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder 
is occasionally too small.

bug - 2882917

Change-Id: Id59d8529084c5689a26f272e0cd3b1e955fd8a30
 2010-08-27 13:59:26 -07:00
James Dong
b86365ad74 Merge "Suppress the video recording start signal - bug 2950297" into gingerbread 2010-08-27 13:47:06 -07:00
Andreas Huber
eeb97d91b9 Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long. 
Change-Id: Id491541a6ae501604cda815f8e961a3bfe26db7d
related-to-bug: 2556656
 2010-08-27 13:29:08 -07:00
Eric Laurent
adecf1c1a9 LVM release 1.07 delivery. 
- Virtualizer now uses the correct control parameter, instead of reverberation
- Volume smoothing for first frame has been added
- Equalizer_setParameter now returns correct error code
- Correcting Non-Linear compressor gain step noise during transitions and effect level changes
- Removed SVN header blocks
- Memory and MIPS values have been added to the API
- Reverb uses a more efficient malloc for input PCM
- Reverb DecayHFRatio now ranges up to 2000
- Logging has been removed for most volume functions

Change-Id: Ib59e7e331263c3811559231b4ae90c82e34a8421
 2010-08-27 11:54:39 -07:00
Andreas Huber
d6a4004741 We accidentally always aborted after 10 secs, even if the connection was fine. 
Change-Id: I3f2ae2f46ae62b84b1e253658d7182c04ee3dfae
 2010-08-27 10:11:04 -07:00
James Dong
d7f1c3d692 Suppress the video recording start signal 
- bug 2950297

Change-Id: I0044d07178691feb904cf81e87c1b6d4b714dc1a
 2010-08-26 16:56:49 -07:00
Andreas Huber
17a765a139 Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread 2010-08-26 13:47:16 -07:00
Andreas Huber
0416da73a0 Support for RTP packets arriving interleaved with RTSP responses. 
Change-Id: Ib32fba257da32a199134cf8943117cf3eaa07a25
 2010-08-26 11:19:08 -07:00