mirror of https://github.com/sipwise/sems.git
master
mr13.5.1
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pristine-tar
upstream
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ngcp/1%1.4.3-1
${ noResults }
4 Commits (6fb4e0eaaa163465e5737c052d23add32a30ffd4)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Raphael Coeffic
|
6fb4e0eaaa |
- fixes a buffer overflow occuring if an RTP packet...:
- ... is too big. - ... has a wrong header length. Thanks to Andrei for that year-old-yet-not-applied patch! git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@221 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
|
6d8f8d9645 |
enabled Id svn keyword property
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@145 8eb893ce-cfd4-0310-b710-fb5ebe64c474 |
19 years ago |
|
Raphael Coeffic
|
37abd5379e |
initial checkin new open source version.
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@5 8eb893ce-cfd4-0310-b710-fb5ebe64c474 |
20 years ago |