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* Make sure the message dispatcher stays around until after OMX_FreeHandle is f...Andreas Huber2010-09-172-3/+7
* Register the new OMX components.Andreas Huber2010-09-171-0/+6
* Merge "Make sure the .wav extractor does not read data outside the bounds of ...Andreas Huber2010-09-161-1/+12
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| * Make sure the .wav extractor does not read data outside the bounds of the 'da...Andreas Huber2010-09-161-1/+12
* | Merge "Fixed a bug in the query to the supported profiles and levels" into gi...James Dong2010-09-161-2/+5
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| * | Fixed a bug in the query to the supported profiles and levelsJames Dong2010-09-161-2/+5
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* | Sometimes the avc software decoder will signal that a frame is ready but then...Andreas Huber2010-09-161-4/+11
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* A ThreadedSource wraps around an existing MediaSource and reads output buffer...Andreas Huber2010-09-164-0/+289
* Merge "Upgrade to the latest .webm project code." into gingerbreadAndreas Huber2010-09-151-5/+7
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| * Upgrade to the latest .webm project code.Andreas Huber2010-09-151-5/+7
* | Merge "Add some explicit error log messages" into gingerbreadJames Dong2010-09-153-7/+11
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| * | Add some explicit error log messagesJames Dong2010-09-143-7/+11
* | | Merge "Fix audio input sample timestamp when audio driver loses audio samples...James Dong2010-09-151-1/+4
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| * | Fix audio input sample timestamp when audio driver loses audio samplesJames Dong2010-09-141-1/+4
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* | Merge "Various fixes to improve resilience of the rtsp stack against spurious...Andreas Huber2010-09-153-36/+84
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| * Various fixes to improve resilience of the rtsp stack against spurious errors...Andreas Huber2010-09-153-36/+84
* | TimedEventQueue now explicitly sets its scheduling policy to foreground as it...Andreas Huber2010-09-091-0/+4
* | Merge "Instead of asserting return a runtime error if the maximum sample size...Andreas Huber2010-09-091-1/+5
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| * | Instead of asserting return a runtime error if the maximum sample size cannot...Andreas Huber2010-09-091-1/+5
* | | Merge "When 32-bit offset is used, if the requested max file size is greater ...James Dong2010-09-091-3/+4
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| * | When 32-bit offset is used,James Dong2010-09-081-3/+4
* | | Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3...Andreas Huber2010-09-092-40/+41
* | | HW audio encoder expects timestamp via kKeyTime from each input bufferJames Dong2010-09-084-4/+6
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* | Merge "Not all audio source has the drift time information" into gingerbreadJames Dong2010-09-033-12/+15
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| * | Not all audio source has the drift time informationJames Dong2010-09-033-12/+15
* | | Ogg files can be tagged to be automatically looping, this setting always over...Andreas Huber2010-09-033-4/+16
* | | Merge "Properly buffer a certain amount of data on streaming sources before f...Andreas Huber2010-09-032-88/+119
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| * | Properly buffer a certain amount of data on streaming sources before finishin...Andreas Huber2010-09-032-88/+119
* | | Remove unused/debugging code from MP4 file writerJames Dong2010-09-031-204/+47
* | | Better file size estimateJames Dong2010-09-021-34/+103
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* | Calculate audio media drift time from AudioSourceJames Dong2010-09-015-31/+60
* | Merge "Better support for buffered streaming of rtsp content, if buffer drops...Andreas Huber2010-09-015-2/+77
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| * Better support for buffered streaming of rtsp content, if buffer drops below ...Andreas Huber2010-09-015-2/+77
* | Merge "Make sure that if initialization fails, AudioSource still behaves well...James Dong2010-09-011-4/+17
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| * Make sure that if initialization fails, AudioSource still behaves well.James Dong2010-09-011-4/+17
* | Merge "Keep gtalk video chat specific code consistent with rtsp changes." int...Andreas Huber2010-09-011-0/+8
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| * | Keep gtalk video chat specific code consistent with rtsp changes.Andreas Huber2010-09-011-0/+8
* | | Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer.Andreas Huber2010-09-012-5/+74
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* | Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-h...Andreas Huber2010-08-314-6/+345
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| * | Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.Andreas Huber2010-08-314-6/+345
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* | Better detection of connection problems - timeout if no rtcp packets arrive w...Andreas Huber2010-08-312-43/+67
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* Merge "Recent changes to the rtsp code require every buffer fed to the packet...Andreas Huber2010-08-301-0/+1
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| * Recent changes to the rtsp code require every buffer fed to the packet source...Andreas Huber2010-08-301-0/+1
* | Instead of closing the connection altogether if no UDP packets arrive after a...Andreas Huber2010-08-302-32/+83
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* Finetune some rtsp timeout constants.Andreas Huber2010-08-301-3/+11
* Merge "ALoopers can now be named (useful to distinguish threads)." into ginge...Andreas Huber2010-08-305-4/+13
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| * ALoopers can now be named (useful to distinguish threads).Andreas Huber2010-08-275-4/+13
* | Workaround for a QCOM issue where the output buffer size advertised by the AV...James Dong2010-08-271-0/+13
* | Merge "Suppress the video recording start signal - bug 2950297" into gingerbreadJames Dong2010-08-271-0/+46
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| * Suppress the video recording start signalJames Dong2010-08-261-0/+46
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