11 Commits (e5dfd2811fdbeb7bf92c54ed9b751a398ddd026c)

Author	SHA1	Message	Date
Stefan Sayer	9fe41686d8	PLC buffer changed to simpler interface (makes testing tool for playout buffer possible) ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@326 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Stefan Sayer	c7f1345d28	puzzle: what happens after 3.106891851851851852 days? ... - initializes correctly r_ts and w_ts (would lead to no audio input after a while) git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@309 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Stefan Sayer	027de94734	fixes rtp ts handling bug in receiving introduced in r281. ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@306 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Stefan Sayer	7c964b9b03	unifies indentation level in all source files to 2 (-i2) ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@261 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Stefan Sayer	f4d64a3d76	applies patch from Andriy which makes adaptive jitter buffer and adaptive playout buffer selectable at run time ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@254 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Maxim Sobolev	1bebb118d9	Improve jitter bufering a bit: ... 1. Use more aggressive resyncronization algorithm. 2. Respect the marker bit in RTP packets and do immediate resync on such packets without increasing the jitter delay. This makes the jitter buffer work well with UAs that use silence suppression. Tested with Cisco and X-Lite. 3. The AmPlayoutBuffer has been modified to allow the audio data be put directly in the audio buffer by real timestamp from RTP. This change can possibly break the adaptive playout buffer so it has been put in "#ifdef USE_ADAPTIVE_JB" blocks. The last change is required due to the fact that internal audio packet size (in timestamp units) can differ from RTP packet size and so the internal audio packet can be overlapped by several RTP packets or vice versa. While the original playout buffer is generally ok but the problem is in the loss concealments that can break the buffer as the concealment algorithm can refuse to return any data and so breaking the timestamp monotony and thus causing the voice corruption. Sponsored by: Digifonica Canada Limited Work done by: Sippy Software, Inc. git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@234 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Stefan Sayer	89ba248551	re-added adaptive playout buffer with the packet size patches from rc1. ... Adaptive jitter buffer can be compiled in instead by making with WITH_ADAPTIVE_JITTERBUFFER=yes (e.g. WITH_ADAPTIVE_JITTERBUFFER=yes make all) Echo application switches with the star key the adaptive playout on and off. git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@225 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Maxim Sobolev	58f33c8556	Second, much improved revision of adaptive jitter buffering patch. ... Changes ------- 1. Make the AmJitterBuffer work with variable size RTP packets. Packet size can be changed even during session (ex. Cisco in fax passthrough mode). Also several improvements and fixes have been made to resyncronization logic. 2. Fix made to the AmPlayoutBuffer class to avoid reading chunks of size larger than requested. This is required in cases when RTP packets contain more or less data than internal frame size. 3. Small fix to AmRtpPacket class - replace the pointer to internal buffer with offset in the buffer. This eliminates the nesessity to reparse the packet each time the packet has been copied. 4. Replace the sample size field in amci_codec_t structure with two functions - sampes2bytes and bytes2samples as that field did not allow to specify sample size for LBR codecs (iLBC, gsm). This also brings ability for codecs to determine the sample size at runtime (ex. iLBC). 5. Remove the sample size from amci_file_desc_t structure as it was used as internal attribute of WAV files only and doesn't make much sense for other file formats. Use the codec's ability to calculate sample size instead. 6. Parameter list for amci_inoutfmt_t.on_close() has been changed to give ability to determine sample size in this file handler (WAV write_header procedure requires this). 7. Fix gsm, ilbc, wav plugins and AmPlugin.c to reflect changes to amci. Add corresponding samples2bytes and bytes2samples functions. Caveats -------- 1. AmAdaptivePlayout class needs additional checking with RTP streams with packets containing number of samples different from internal frame size (for example 240 samples per packet in G711). Adaptive playout class potentially may produce big packets of audio and therefore make the Conference application work badly. The adaptive playout is used in Conference application only and the application is working fine now with jitter buffer and without adaptive playout. So I turned the adaptive playout off in the Conference app as a workaround. Developed by: Sippy Software, Inc. Sponsored by: Digifonica Canada Limited git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@185 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Stefan Sayer	809a2727d7	added doxygen class documentation ... beginning of the sems documentation initiative... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@83 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Raphael Coeffic	2e28833e88	* fixes timestamp overflow problem after about 7 days. ... this would have cause SEMS to stop send any packets after about 7 days. * adds a simple playout buffer for application which do not need adaptive playout (voicemail, annoucement, ...). git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@51 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Raphael Coeffic	75a8aa318a	- added adaptive playout buffer with order statistic based delay estimation ... and adaptive WSOLA playout delay modification. - fixed serious bug in the RTP receiver. git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@41 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago