From: Ken L. <ken...@us...> - 2007-08-24 16:42:32
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Actually I was hoping for a simple one-file example that uses a Processor. Perhaps the best way for me to try this is to try transmitting an mp3 with RTP myself, using FMJ studio. Ken Damian Minkov wrote: > Hi, > > I'm testing it with sip-communicator. I call a sip number in asterisk > playing an mp3 and sending it to ulaw. > playing it on the speakers and the rate conversation is after its > captured from the microphone. > Do you want to send you the changes I have made in sip-communicator so > you can run this on your machine ? > > damencho > > > Ken Larson wrote: > >> Damian - >> >> Do you have a sample program, and a sample media file i can use to >> test this? >> >> Thanks, >> >> Ken >> >> Damian Minkov wrote: >> >>> Hi, >>> >>> I am trying to use fmj RateConverter but it dosn't work. >>> I remove sun and ibm ones so they are no more available as plugins >>> and add the fmj one. but the captured media is scrambled. (I've >>> attached some part of the raw captured media). >>> >>> and here is the log I've put in the process method for the conversation. >>> >>> [java] 18:32:55.380 INFO: fmj.process() fmj numChanged 2 >>> from: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed, >>> 88200.0 frame rate, FrameSize=32 bits >>> to: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed, 8000.0 >>> frame rate, FrameSize=16 bits >>> >>> thanks :) >>> damencho >>> >>> >>> Ken Larson wrote: >>> >>>> See what formats are being returned from >>>> >>>> net.sf.fmj.media.protocol.javasound.DataSource.querySupportedFormats >>>> >>>> Ken >>>> >>>> Damian Minkov wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm trying to force capture with the same samplerate as the >>>>> incoming stream (LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, >>>>> Signed) so the resampling to be skipped, but with no luck :( >>>>> >>>>> damencho >>>>> >>>>> Ken Larson wrote: >>>>> >>>>>> Damian - >>>>>> >>>>>> The FMJ one can actually make multiple conversions. It used to >>>>>> only do one at a time when I was testing it. >>>>>> (ONLY_CHANGE_1_PARAMETER false). >>>>>> >>>>>> However, the main drawback to the FMJ one is that it doesn't do >>>>>> any interpolation or anything to optimize audio quality when >>>>>> resampling. >>>>>> >>>>>> Ken >>>>>> >>>>>> >>>>>> Damian Minkov wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> thanks. I found that actually is used >>>>>>> com.ibm.media.codec.audio.rc.RCModule. >>>>>>> I've tried to use the fmj one (as compiling it under ibm package >>>>>>> with the same name) but I saw that >>>>>>> fmj.media.codec.audio.RateConverter can make one conversation at >>>>>>> a time which may be the reason for not using it then >>>>>>> jmf switches to com.sun.media.codec.audio.rc.RateCnvt and the >>>>>>> sound quality is much much better but still has the problem with >>>>>>> the time differs if you put both raw streams captured from mic >>>>>>> and speaker in audacity you can see in the graph that after 20 >>>>>>> seconds the stream has difference in about 20 ms. Which is much >>>>>>> better then when the ibm's one is used. >>>>>>> >>>>>>> damencho >>>>>>> >>>>>>> Ken Larson wrote: >>>>>>> >>>>>>>> Damian - >>>>>>>> >>>>>>>> This would normally be done by a rate conversion codec. >>>>>>>> >>>>>>>> In FMJ, this would be net.sf.fmj.media.codec.audio.RateConverter >>>>>>>> In JMF, this would be com.sun.media.codec.audio.rc.RateCnvt >>>>>>>> >>>>>>>> Ken >>>>>>>> >>>>>>>> Damian Minkov wrote: >>>>>>>> >>>>>>>>> Hi Ken, >>>>>>>>> >>>>>>>>> I have successfully run FMJ JavasoundDatasource and the problem >>>>>>>>> in my tests is the same as the jmf one. Now I think that the >>>>>>>>> real problem is the resampling the media, as I recorded the >>>>>>>>> media coming from the mic directly from the datasource (44100) >>>>>>>>> and the media is ok. but the media captured in the Effect(8000) >>>>>>>>> has problems. I saw a mail on this case in jmf mailinglist but >>>>>>>>> no answers there. >>>>>>>>> http://archives.java.sun.com/cgi-bin/wa?A2=ind0106&L=jmf-interest&P=9775 >>>>>>>>> >>>>>>>>> >>>>>>>>> Do you know where, in which class is done the resampling? Thanks >>>>>>>>> >>>>>>>>> damencho >>>>>>>>> >>>>>>>>> Ken Larson wrote: >>>>>>>>> >>>>>>>>>> Damian - >>>>>>>>>> >>>>>>>>>> I honestly hadn't tested the javasound datasource much, so I >>>>>>>>>> had a closer look and fixed a number of problems, the main one >>>>>>>>>> being that it was not calling the buffer transfer handler, as >>>>>>>>>> you observed. >>>>>>>>>> >>>>>>>>>> See if it works better now, >>>>>>>>>> >>>>>>>>>> Ken >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>> >>>>> >>> > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > Fmj-devel mailing list > Fmj...@li... > https://lists.sourceforge.net/lists/listinfo/fmj-devel > > |