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* Respect the cropping rectangle when extracting the video dimensions from the ↵Andreas Huber2010-10-081-1/+38
| | | | | | sequence parameter set of H.264 streams. Change-Id: Id7f21d5ab8533b4e17ceaa8619a39c27f0306150
* Increase scratch buffers sizes in mp3 software decoder, this integrates a PV ↵Andreas Huber2010-10-073-4/+4
| | | | | | | master/opensource patch. Change-Id: I5a637f1b380e44c94040ec507843d58a1f5a9b61 related-to-bug: 3065605
* Work to support switching transport streams mid-stream and signalling ↵Andreas Huber2010-10-0714-110/+459
| | | | | | | discontinuities to the decoder. Change-Id: I7150e5e7342e1117c524856b204aadcb763e06ed related-to-bug: 2368598
* On this particular device the hardware video decoder spits out buffers that ↵Andreas Huber2010-10-073-16/+51
| | | | | | | 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
* Make sure to call AudioTrack::stop() instead of AudioTrack::pause() after ↵Andreas Huber2010-10-053-11/+26
| | | | | | submitting all samples to AudioTrack to make sure those remaining samples are actually played out. Change-Id: Id574a0203efcb5e565f1b0fe77869fc33b9a9d56
* Fixed an issue where the reserved free space in the file writer was larger ↵James Dong2010-10-041-2/+14
| | | | | | | | | | | | | | | 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 "Resilent media time stamp adjustment" into gingerbreadJames Dong2010-10-041-24/+181
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| * Resilent media time stamp adjustmentJames Dong2010-10-041-24/+181
| | | | | | | | Change-Id: I13ab87c05f26bb11a3cc9bf8559f98e6ea0752db
* | Make sure to finish the preparation phase even EOS occurs before we consider ↵Andreas Huber2010-10-041-0/+4
| | | | | | | | | | | | | | the cache to be completely filled up. Change-Id: I29143e357fb6ea7b860636100e010f2ea7436798 related-to-bug: 3037389
* | Don't retrieve metadata unless necessary for ogg-vorbis ringtone auto-looping.Andreas Huber2010-10-041-5/+11
|/ | | | | Change-Id: Iaf5880bb3376f9cbf22aefe198878eaf6f3f08c7 related-to-bug: 3037389
* Merge "Turn off media time adjustment by default" into gingerbreadJames Dong2010-10-011-1/+1
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| * Turn off media time adjustment by defaultJames Dong2010-10-011-1/+1
| | | | | | | | Change-Id: I1f8021d605d0fd896e0639607a84e3f7c459612e
* | Merge "Start playing live streams from the start, no the middle..." into ↵Andreas Huber2010-10-011-1/+1
|\ \ | | | | | | | | | gingerbread
| * | Start playing live streams from the start, no the middle...Andreas Huber2010-10-011-1/+1
| | | | | | | | | | | | | | | Change-Id: Ie01ba1250b51155cb1fb32fc3340189a16c01476 related-to-bug: 2368598
* | | Remove development-only code.Andreas Huber2010-10-011-15/+0
|/ / | | | | | | | | Change-Id: Ic2ca0efb631eb779ca157fb01b02aa19a1222c06 related-to-bug: 2368598
* | Squashed commit of the following:Andreas Huber2010-10-019-357/+597
|/ | | | | | | | | | | | | | | | | | | | | 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
* Issue 3032913: improve AudioTrack recovery timeEric Laurent2010-09-301-0/+8
| | | | | | | | | | | | 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 "AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead." ↵Chia-chi Yeh2010-09-283-55/+19
|\ | | | | | | into gingerbread
| * AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead.Chia-chi Yeh2010-09-293-55/+19
| | | | | | | | | | | | | | This allows gsmamr_enc.h and gsmamr_dec.h to be included in the same file without conflict definition. Change-Id: I1c8dac235c122735ba14a1af2fda48c0f8d9f87b
* | Fix several audio effects problems.Eric Laurent2010-09-282-126/+211
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 "Properly flush the AudioTrack/AudioSink on a seek request and make ↵Andreas Huber2010-09-283-0/+14
|\ \ | | | | | | | | | sure that both the mp3 decoder and aac software decoders start fresh after a seek without any dependency on previously decoded content." into gingerbread
| * | Properly flush the AudioTrack/AudioSink on a seek request and make sure that ↵Andreas Huber2010-09-283-0/+14
| | | | | | | | | | | | | | | | | | | | | 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
* | | Merge "Instead of constantly polling the AudioPlayer to see if it reached ↵Andreas Huber2010-09-283-8/+27
|\ \ \ | |/ / | | | | | | EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens." into gingerbread
| * | Instead of constantly polling the AudioPlayer to see if it reached EOS or ↵Andreas Huber2010-09-283-8/+27
| | | | | | | | | | | | | | | | | | | | | finished seeking, initiate the notification from the AudioPlayer when the event happens. Change-Id: I43875b6adaf96d4e982ef3dfc3d6c8f7034ac51d related-to-bug: 3036592
* | | Merge "Vorbis files may have more samples encoded that should be used, i.e. ↵Andreas Huber2010-09-283-0/+96
|\ \ \ | |/ / | | | | | | we have to trim samples at the end of the stream. This is crucial for proper looping of some audio files." into gingerbread
| * | Vorbis files may have more samples encoded that should be used, i.e. we have ↵Andreas Huber2010-09-283-0/+96
| | | | | | | | | | | | | | | | | | | | | 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
* | | Merge "Squashed commit of the following:" into gingerbreadAndreas Huber2010-09-284-42/+820
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| * | Squashed commit of the following:Andreas Huber2010-09-274-42/+820
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | | Fix media.player dumpsys to output open/mapped files correctly. Bug 2866669.Dave Sparks2010-09-271-2/+4
| |/ |/| | | | | | | | | | | | | | | 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
* | Merge "Fix track duration calculation if the start timestamp is non-zero" ↵James Dong2010-09-271-3/+36
|\ \ | |/ |/| | | into gingerbread
| * Fix track duration calculation if the start timestamp is non-zeroJames Dong2010-09-271-3/+36
| | | | | | | | | | | | | | o Updated the comments in the patch o Added some additional checks on the timestamp Change-Id: I8ad81eb6bfe358c1db5245cbb52efc905cdc234c
* | Merge "Support other kinds of HTTP redirect in NuHTTPDataSource" into ↵Andreas Huber2010-09-271-1/+6
|\ \ | | | | | | | | | gingerbread
| * | Support other kinds of HTTP redirect in NuHTTPDataSourceAndreas Huber2010-09-271-1/+6
| |/ | | | | | | | | Change-Id: I3268a94cfb48cee9caa8914804d7907ff91d8a8e related-to-bug: 3037686
* | Fix error in AudioEffect command status reporting.Eric Laurent2010-09-243-13/+23
|/ | | | | | | | | | | | 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
* Merge "Depending on our preference to write 2-byte or 4-byte NALs, patch the ↵Andreas Huber2010-09-231-1/+9
|\ | | | | | | codec specific data 'avcC' accordingly." into gingerbread
| * Depending on our preference to write 2-byte or 4-byte NALs, patch the codec ↵Andreas Huber2010-09-231-1/+9
| | | | | | | | | | | | specific data 'avcC' accordingly. Change-Id: I1423923fafbcac6911218277e2bbfa068e49dcc6
* | Ignore errors from correction parameter query and config for M4v and H263 ↵James Dong2010-09-231-2/+9
| | | | | | | | | | | | | | | | encoders This feature is not essential for camcorder application Change-Id: Id560632207dcf24fe361280ebe8178ea7332f22d
* | Use the advertised profile and level from M4V and H263 video encodersJames Dong2010-09-221-4/+4
|/ | | | | | This is already the case for AVC video encoder Change-Id: I9cc97a03ab457b4bd0b21832c09f6d0aa7f5fc39
* Raise the amount of memory set aside for omx buffer allocations in the test ↵Andreas Huber2010-09-221-1/+1
| | | | | | harness to accomodate the new requirements of some codecs. Change-Id: I73974a336852974e2fb95a9626c1a96f296449eb
* Make sure we drain the avc software decoder's output queue once we run out ↵Andreas Huber2010-09-222-43/+55
| | | | | | of input data. Change-Id: I3d92ebae4951b618a6dfcf42936cc4c63d4de539
* Merge "Fix issue 2913071." into gingerbreadEric Laurent2010-09-211-2/+20
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| * Fix issue 2913071.Eric Laurent2010-09-211-2/+20
| | | | | | | | | | | | | | | | | | | | | | 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
* | Merge "This log message is codec specific." into gingerbreadAndreas Huber2010-09-211-1/+1
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| * | This log message is codec specific.Andreas Huber2010-09-211-1/+1
| |/ | | | | | | Change-Id: I603ae7bd13b186a6df0a26ea9abe073e89276caf
* | Remove stagefright foundation's incompatible logging interface and update ↵Andreas Huber2010-09-2121-435/+298
|/ | | | | | callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
* Another attempt for fixing AAC+/eAAC+ related issueJames Dong2010-09-202-12/+59
| | | | | | | | | 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
* Add a check to track a problem the monkey script has been triggering.Marco Nelissen2010-09-171-0/+3
| | | | Change-Id: If3896946d993cbab2442e17740f18efd7e753a3b
* Make sure the message dispatcher stays around until after OMX_FreeHandle is ↵Andreas Huber2010-09-172-3/+7
| | | | | | finished in case it posts some more messages during shutdown. Clear the source as soon as possible in OMXCodec's destructor. Change-Id: I9c896cf07dea0c3201b6f074dbaf27e6d85cd784
* Register the new OMX components.Andreas Huber2010-09-171-0/+6
| | | | Change-Id: I795287c63073ca86ad0f97d6afcfc85c401dc984
* Merge "Make sure the .wav extractor does not read data outside the bounds of ↵Andreas Huber2010-09-161-1/+12
|\ | | | | | | the 'data' box." into gingerbread
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