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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
* A ThreadedSource wraps around an existing MediaSource and reads output buffer...Andreas Huber2010-09-164-0/+289
* Merge "Fix problem in lvm effect bundle wrapper" into gingerbreadEric Laurent2010-09-161-123/+127
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| * Fix problem in lvm effect bundle wrapperEric Laurent2010-09-161-123/+127
* | 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
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* | | 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
* | Fix volume problems with insert revertEric Laurent2010-09-134-57/+152
* | Merge "LVM release 1.09 delivery" into gingerbreadEric Laurent2010-09-104-135/+69
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| * | LVM release 1.09 deliveryEric Laurent2010-09-094-135/+69
* | | 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
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* | | Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer...James Dong2010-09-085-4/+7
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| * | | HW audio encoder expects timestamp via kKeyTime from each input bufferJames Dong2010-09-085-4/+7
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* | | Modify type of some environmental reverb parametersEric Laurent2010-09-082-24/+26
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* | Merge "LVM release 1.08 delivery." into gingerbreadEric Laurent2010-09-035-41/+69
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| * | LVM release 1.08 delivery.Eric Laurent2010-09-035-41/+69
* | | 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 Huber2010-09-034-4/+18
* | | 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 Dong2010-09-033-204/+53
* | | Better file size estimateJames Dong2010-09-022-34/+104
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* | Merge "Calculate audio media drift time from AudioSource" into gingerbreadJames Dong2010-09-028-32/+63
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| * | Calculate audio media drift time from AudioSourceJames Dong2010-09-018-32/+63
* | | Merge "Fix problem in AudioEffect::command() status." into gingerbreadEric Laurent2010-09-021-5/+9
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| * | Fix problem in AudioEffect::command() status.Eric Laurent2010-09-021-5/+9
* | | 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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* | | Audio Effects: fix problems in volume control.Eric Laurent2010-08-313-15/+54
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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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* | Merge "Better detection of connection problems - timeout if no rtcp packets a...Andreas Huber2010-08-312-43/+67
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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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* | Add some encoding parameters for the "record" utilityJames Dong2010-08-311-1/+19
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