What about G.722 delivering a 16 khz audio stream, and not 8 khz as is hardcoded in rtp_player.c's device opening call to PortAudio currently?
Where is an explanation about the string tags used in register_codec_module? Can I use something like "g726_32"? > From: [email protected] > To: [email protected] > Date: Thu, 27 Jan 2011 10:02:37 +0100 > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark > > Hi, > > I would recommend to use codec table (see epan/codecs.h) for new codecs > instead of including them hardcoded into the rtp_player. > It should not require any changes in the rtp_player for any new codec. > Only changes for dynamic payload handling should be implemented but it is > common for all codecs. > > The codec table is used just for codec plugins now but using for internal > Wireshak codec would be good too and does not need many changes (I hope ;-). > > Example of G.722 implemented inside plugin (using external library) is here: > http://anonsvn.wireshark.org/viewvc/trunk/plugins/easy_codec/easy_codec_plugin.c?view=markup > http://anonsvn.wireshark.org/viewvc/trunk/plugins/easy_codec/codec-g722.c?view=markup > > Wireshak internal implementation using spandsp library could be very similar. > > Best regards, > Tomas > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Jaap Keuter > Sent: Wednesday, January 26, 2011 10:22 PM > To: Developer support list for Wireshark > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark > > Hi, > > Found it, I'll have a look. > > Thanks, > Jaap > > On 01/26/2011 04:59 PM, Dietfrid Mali wrote: > > G.722 and G.726 (-32) codec integration using spandsp: > > https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=5619 > > > > > Date: Wed, 26 Jan 2011 00:09:50 +0100 > > > From: [email protected] > > > To: [email protected] > > > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark > > > > > > On 01/25/2011 05:48 PM, Dietfrid Mali wrote: > > > > The problem with e.g. G.726 is that Wireshark gives those packets RTP > > > > type 102 which afaik is an error code ("unknown encoding"). > > > > > > No, that's your RTP endpoint configured to label these as such. RFC > > 3550 says: > > > "A profile MAY specify a default static mapping of payload type codes > > to payload > > > formats. Additional payload type codes MAY be defined dynamically > > through > > > non-RTP means (see Section 3)." > > > RFC 1890/RFC 3551 defines the "RTP Profile for Audio and Video > > Conferences with > > > Minimal Control", which lists several static payload types. The old > > RFC lists > > > G.721 (aka G.726-32), while the new one dropped that one and added > > references to > > > G.726 at various bit rate, with a dynamic payload type. > > > RFC 3550 says in Section 3: "Non-RTP means: Protocols and mechanisms > > that may be > > > needed in addition to RTP to provide a usable service. In particular, > > ..., and > > > define dynamic mappings between RTP payload type values and the > > payload formats > > > they represent for formats that do not have a predefined payload type > > value." > > > with reference to Session Description Protocol (SDP) > > > > > > So, payload type 102 is a dynamic payload type which has to be given > > meaning > > > (through SDP for instance) within the session. In your case Wireshark > > didn't > > > pick that up from the trace, hence cannot give you the proper > > interpretation of > > > that payload type within that session. > > > > > > > I would need to know where and how Wireshark maps dynamic payload types > > > > (negotiated via SDP) to internal static ones. Above that RFC3551 > > notes that > > > > static G.726 payload types are obsolete, and afaik there aren't even > > > > (obsolete) static payload types for all G.726 variants, so Wireshark > > > > would need to > > > > take care of that by using some (more or less arbitrary) internal > > static > > > > type numbers. > > > > > > Yep, that is done by the SDP dissector. It tries to interpret the SDP > > offer > > > (should be the answer, but that a whole other story) and create > > conversations > > > (see doc/README.developer, section 2.2) for the RTP dissector, > > feeding it > > > dynamic payload type information it has learned from the media > > attributes. > > > > > > The RTP dissector does then the heavy lifting on the RTP packets, > > based on the > > > information feed in by the SDP dissector. > > > > > > > I will do my best to provide a patch once I have fully integrated all > > > > codecs (currently only G.726-32 has been implemented as proof of > > > > concept, but since > > > > this is working adding more is no big deal). > > > > > > Just one to get started is fine. Does it integrate into codecs/ > > directory > > > besides G711a and G711u (and G729 and G723, if you have them)? > > > > > > > Getting G.726 to work was a bit of a pain btw because of the weird > > frame > > > > sync calculation in rtp_player.c::play_channels() as this function > > seems to > > > > assume 1:1 relationships of decoder input and output stream sizes and > > > > thus simply halves the decoder output batch sizes to determine whether > > > > frames > > > > are properly sync'd. This doesn't work for compressed audio. To > > > > compensate, my decodeG726_32() function doubles the number of bytes > > returned > > > > (as it has a 1:2 relationship of input and output buffer sizes). Before > > > > it did that, lots of silence frames were inserted and half of the audio > > > > data was > > > > dropped by the player. > > > > > > I'm no sure if I understand you correctly. Working with these decode > > functions > > > there is an input buffer with its length as input, and two output > > parameters, > > > being the output buffer and it a pointer to store its size. This size > > of the > > > output buffer has to be set, by the decoder, to the number of samples > > in output > > > buffer. That should be enough, see for instance > > rtp_player.c:decode_rtp_packet() > > > the handling of G.279 and G.723. > > > Be aware that you have to store 16 bit linear samples in the output > > buffer, > > > maybe that's your factor 2? > > > > > > Thanks, > > > Jaap > > > > > > > > > > > > > > Dietfrid > > > > > > > > > From: [email protected] > > > > > Date: Tue, 25 Jan 2011 16:54:25 +0100 > > > > > To: [email protected] > > > > > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark > > > > > > > > > > Hi, > > > > > > > > > > That would be interesting. Can you put the code in a patch on > > bugzilla? > > > > > > > > > > Can't work on it right now, but would be nice to have. > > > > > > > > > > btw: their are already static RTP types assigned for both codecs. The > > > > dynamic types should come in through protocols like SDP, or a dissector > > > > preference. > > > > > > > > > > Thanks, > > > > > Jaap > > > > > > > > > > Send from my iPhone > > > > > > > > > > On 25 jan. 2011, at 16:07, Dietfrid Mali <[email protected]> > > wrote: > > > > > > > > > > > Hi, > > > > > > > > > > > > using spandsp, I have added G.722 and G.726 decoders to Wireshark. > > > > > > > > > > > > Currently this is a bit of a hack job, particularly regarding > > > > inclusion of the spandsp lib, and I could need a bit help to properly > > > > integrate it into Wireshark's automake hell (configure.in). > > > > > > > > > > > > There also isn't a proper Wireshark signature for that RTP type (I > > > > am simply reacting to RTP type 102, which actually is an error > > code), so > > > > some help getting this straight and introducing proper codec types > > would > > > > be appreciated, too. > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > Sent via: Wireshark-dev mailing list <[email protected]> > Archives: http://www.wireshark.org/lists/wireshark-dev > Unsubscribe: https://wireshark.org/mailman/options/wireshark-dev > mailto:[email protected]?subject=unsubscribe > ___________________________________________________________________________ > Sent via: Wireshark-dev mailing list <[email protected]> > Archives: http://www.wireshark.org/lists/wireshark-dev > Unsubscribe: https://wireshark.org/mailman/options/wireshark-dev > mailto:[email protected]?subject=unsubscribe
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