2334 Commits

Author SHA1 Message Date
Andreas Huber
bb70837397 am 949f7d90: Merge "Work to support switching transport streams mid-stream and signalling discontinuities to the decoder." into gingerbread
Merge commit '949f7d9066e09768e570686a5695aaba4a1dafd0' into gingerbread-plus-aosp

* commit '949f7d9066e09768e570686a5695aaba4a1dafd0':
  Work to support switching transport streams mid-stream and signalling discontinuities to the decoder.
2010-10-07 14:02:26 -07:00
Andreas Huber
45bd1159fa am 02654f01: Merge "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." into gingerbread
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.
2010-10-07 14:02:01 -07:00
Andreas Huber
4c19bf9833 Work to support switching transport streams mid-stream and signalling discontinuities to the decoder.
Change-Id: I7150e5e7342e1117c524856b204aadcb763e06ed
related-to-bug: 2368598
2010-10-07 11:41:43 -07:00
Andreas Huber
1e19416383 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
2010-10-07 09:19:25 -07:00
Andreas Huber
56ee1080f0 am 17bc4f65: Merge "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." into gingerbread
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.
2010-10-05 14:13:16 -07:00
Andreas Huber
c743f4506f 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
2010-10-05 13:53:39 -07:00
Jeff Brown
f358cecb9c am 79a3981e: Fix Looper leaks in MediaRecorderStressTest.
Merge commit '79a3981e3885b9144bb3d458682141eed7365939' into gingerbread-plus-aosp

* commit '79a3981e3885b9144bb3d458682141eed7365939':
  Fix Looper leaks in MediaRecorderStressTest.
2010-10-04 21:05:02 -07:00
Jeff Brown
79a3981e38 Fix Looper leaks in MediaRecorderStressTest.
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
2010-10-04 21:01:29 -07:00
James Dong
a86a6c4e32 am 6f1c7bda: Merge "Fixed an issue where the reserved free space in the file writer was larger than intended" into gingerbread
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
2010-10-04 18:39:52 -07:00
James Dong
6a9e39ac55 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
2010-10-04 16:54:59 -07:00
James Dong
6a02ba0100 am b99f0c7e: Merge "Resilent media time stamp adjustment" into gingerbread
Merge commit 'b99f0c7eae365f788a21944fef28de580c7f5f92' into gingerbread-plus-aosp

* commit 'b99f0c7eae365f788a21944fef28de580c7f5f92':
  Resilent media time stamp adjustment
2010-10-04 15:48:59 -07:00
James Dong
b99f0c7eae Merge "Resilent media time stamp adjustment" into gingerbread 2010-10-04 15:35:40 -07:00
James Dong
acee8e7131 Resilent media time stamp adjustment
Change-Id: I13ab87c05f26bb11a3cc9bf8559f98e6ea0752db
2010-10-04 15:11:19 -07:00
Andreas Huber
dfd03fe993 am aa1a694d: Merge "Make sure to finish the preparation phase even EOS occurs before we consider the cache to be completely filled up." into gingerbread
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.
2010-10-04 11:47:52 -07:00
Andreas Huber
05f6787b88 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
2010-10-04 11:36:39 -07:00
Andreas Huber
bb506dab1f am 7fa69374: Merge "Don\'t retrieve metadata unless necessary for ogg-vorbis ringtone auto-looping." into gingerbread
Merge commit '7fa693740756123fa9d05e62fb47aae5d703c71d' into gingerbread-plus-aosp

* commit '7fa693740756123fa9d05e62fb47aae5d703c71d':
  Don't retrieve metadata unless necessary for ogg-vorbis ringtone auto-looping.
2010-10-04 11:22:22 -07:00
Andreas Huber
1913c1aeab Don't retrieve metadata unless necessary for ogg-vorbis ringtone auto-looping.
Change-Id: Iaf5880bb3376f9cbf22aefe198878eaf6f3f08c7
related-to-bug: 3037389
2010-10-04 11:09:31 -07:00
James Dong
2bf74b8f0a am 6c609b6a: Merge "Turn off media time adjustment by default" into gingerbread
Merge commit '6c609b6a60e533a93c6d0088222bd8da209b9953' into gingerbread-plus-aosp

* commit '6c609b6a60e533a93c6d0088222bd8da209b9953':
  Turn off media time adjustment by default
2010-10-01 17:32:16 -07:00
James Dong
6c609b6a60 Merge "Turn off media time adjustment by default" into gingerbread 2010-10-01 17:27:51 -07:00
James Dong
9160e4aa2a Turn off media time adjustment by default
Change-Id: I1f8021d605d0fd896e0639607a84e3f7c459612e
2010-10-01 17:14:23 -07:00
Andreas Huber
57853559c3 am e619a9da: Merge "Start playing live streams from the start, no the middle..." into gingerbread
Merge commit 'e619a9da44e4c00f9034917aef67f86da0bc207f' into gingerbread-plus-aosp

* commit 'e619a9da44e4c00f9034917aef67f86da0bc207f':
  Start playing live streams from the start, no the middle...
2010-10-01 11:37:44 -07:00
Andreas Huber
e619a9da44 Merge "Start playing live streams from the start, no the middle..." into gingerbread 2010-10-01 11:35:17 -07:00
Andreas Huber
a424f7c628 Start playing live streams from the start, no the middle...
Change-Id: Ie01ba1250b51155cb1fb32fc3340189a16c01476
related-to-bug: 2368598
2010-10-01 11:28:44 -07:00
Andreas Huber
d6d5cfb914 am 469b8033: Merge "Remove development-only code." into gingerbread
Merge commit '469b80336368bef3742e97c15e9017c1f2d404ae' into gingerbread-plus-aosp

* commit '469b80336368bef3742e97c15e9017c1f2d404ae':
  Remove development-only code.
2010-10-01 11:25:57 -07:00
Andreas Huber
21d28a2a13 Remove development-only code.
Change-Id: Ic2ca0efb631eb779ca157fb01b02aa19a1222c06
related-to-bug: 2368598
2010-10-01 11:19:26 -07:00
Andreas Huber
14401bf7bc am d1398db3: Merge "Squashed commit of the following:" into gingerbread
Merge commit 'd1398db35cb2e4d918fc631dda35cacb8540b187' into gingerbread-plus-aosp

* commit 'd1398db35cb2e4d918fc631dda35cacb8540b187':
  Squashed commit of the following:
2010-10-01 11:03:47 -07:00
Andreas Huber
b72c7e36a2 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
2010-10-01 10:51:41 -07:00
Eric Laurent
de12c3cf56 am 220ab887: Merge "Issue 3032913: improve AudioTrack recovery time" into gingerbread
Merge commit '220ab8877b234e6807b7f6d9028ba55d23220301' into gingerbread-plus-aosp

* commit '220ab8877b234e6807b7f6d9028ba55d23220301':
  Issue 3032913: improve AudioTrack recovery time
2010-09-30 17:47:07 -07:00
Eric Laurent
4712baab81 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
2010-09-30 17:21:23 -07:00
Chia-chi Yeh
10b15c08bb am d6877fa4: Merge "AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead." into gingerbread
Merge commit 'd6877fa4971710150de20453bf4ba54dca863429' into gingerbread-plus-aosp

* commit 'd6877fa4971710150de20453bf4ba54dca863429':
  AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead.
2010-09-28 21:13:54 -07:00
Chia-chi Yeh
d6877fa497 Merge "AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead." into gingerbread 2010-09-28 21:11:27 -07:00
Chia-chi Yeh
081833d791 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
2010-09-29 12:00:18 +08:00
Eric Laurent
b047e3cdf2 am aeb2c62e: Merge "Fix several audio effects problems." into gingerbread
Merge commit 'aeb2c62e7669f004512c42ad8572d1fdd2c25f68' into gingerbread-plus-aosp

* commit 'aeb2c62e7669f004512c42ad8572d1fdd2c25f68':
  Fix several audio effects problems.
2010-09-28 16:42:14 -07:00
Eric Laurent
4fd3ecc1f0 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
2010-09-28 14:23:39 -07:00
Andreas Huber
387bdcdadc am 88a995ed: 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
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.
2010-09-28 13:26:26 -07:00
Andreas Huber
88a995edcf 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 2010-09-28 13:23:26 -07:00
Andreas Huber
ad3fcfe845 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
2010-09-28 13:13:38 -07:00
Andreas Huber
d6c30e8c15 am be045061: 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
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.
2010-09-28 12:05:08 -07:00
Andreas Huber
be04506195 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 2010-09-28 12:02:35 -07:00
Andreas Huber
2b359ed5b5 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
2010-09-28 11:56:39 -07:00
Andreas Huber
c889bbfa96 am 4769f579: 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
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.
2010-09-28 11:48:44 -07:00
Andreas Huber
4769f57948 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 2010-09-28 11:46:41 -07:00
Andreas Huber
9d59186068 am 3a9cc8ce: Merge "Squashed commit of the following:" into gingerbread
Merge commit '3a9cc8cef49165d6b1a1eb37b15a0ae21b0042f7' into gingerbread-plus-aosp

* commit '3a9cc8cef49165d6b1a1eb37b15a0ae21b0042f7':
  Squashed commit of the following:
2010-09-28 11:35:22 -07:00
Andreas Huber
3a9cc8cef4 Merge "Squashed commit of the following:" into gingerbread 2010-09-28 11:27:33 -07:00
Andreas Huber
38ae220bb7 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
2010-09-28 11:26:03 -07:00
Dave Sparks
7dff9dfd61 am 269dea1c: Merge "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 t
Merge commit '269dea1c48640ec2bf5a4e9de851cac2dbfbafd1' into gingerbread-plus-aosp

* commit '269dea1c48640ec2bf5a4e9de851cac2dbfbafd1':
  Fix media.player dumpsys to output open/mapped files correctly. Bug 2866669.
2010-09-28 08:52:23 -07:00
Dave Sparks
124a346c7a 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
2010-09-27 17:04:05 -07:00
Andreas Huber
c751eccad3 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
2010-09-27 12:04:43 -07:00
James Dong
a9f17c8b42 am bf47092e: Merge "Fix track duration calculation if the start timestamp is non-zero" into gingerbread
Merge commit 'bf47092e1e76224474745b6919febd3aa9df5bcf' into gingerbread-plus-aosp

* commit 'bf47092e1e76224474745b6919febd3aa9df5bcf':
  Fix track duration calculation if the start timestamp is non-zero
2010-09-27 10:13:04 -07:00
James Dong
bf47092e1e Merge "Fix track duration calculation if the start timestamp is non-zero" into gingerbread 2010-09-27 10:08:52 -07:00