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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Hi Ross<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I know you won’t be able to debug my code but can you say if I’m going in the right direction or if there is anything obviously missing from this description?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I have written PCMSource.cpp, based on DeviceSource.cpp, in which createNew() creates a socket and connects to the remote endpoint, and doGetNextFrame() does
the recv() and then calls deliverFrame() to do the memmove(fTo, receivedData).<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">PCMStreamServerMediaSubsession.cpp is based on WAVAudioFileServerMediaSubsession.cpp but the createNewStreamSource() can hardcode the bits/sample, sampling
frequency and number of channels information that would normally come from the .wav file header.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">These new files compile on CentOS 5.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Adding an extra section to testOnDemandRTSPServer.cpp to create a stream name allows me to enter the url and reproduce the speech from the original mp3 podcast
I’m using to test this with VLC from a Win 7 workstation. Nice; the raw samples are being converted to a recognisable RTP stream. Unfortunately, I am only getting the first second of a minute’s worth of audio data.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">openRTSP -Q rtsp://134.32.45.50:8554/pcmAudioTest shows that a variable number of packets are being lost in a sequence of tests.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I haven’t done anything with the TaskScheduler code. I have changed MediaSink.cpp to set OutPacketBuffer::maxSize appropriately for the amount of data I am
sending.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Thanks<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Martin<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> live-devel-bounces@ns.live555.com [mailto:live-devel-bounces@ns.live555.com]
<b>On Behalf Of </b>Ross Finlayson<br>
<b>Sent:</b> 25 April 2012 15:45<br>
<b>To:</b> LIVE555 Streaming Media - development & use<br>
<b>Subject:</b> Re: [Live-devel] DeviceSource.cpp socket implementation<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I’m wondering if anybody has a simple socket reading version of the testOnDemandRTSPServer program.</span><o:p></o:p></p>
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<p class="MsoNormal">What you really mean to ask for is "a subclass of the 'OnDemandServerMediaSubsession' class that reads from a socket". There is such a class that you can use as a model: "MPEG2TransportUDPServerMediaSubsession". Note also how this is
used in "testOnDemandRTSPServer".<o:p></o:p></p>
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<p class="MsoNormal">If, however, you are asking about reading from a *stream* (i.e., TCP) socket, rather than from a datagram socket, then you should be able to convert the socket into a "FILE*" (assuming that you're not using Windoze), and instead use "WAVAudioFileServerMediaSubsession"
as a model for your new "OnDemandServerMediaSubsession" subclass.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">(Either way, you will need to write your own "OnDemandServerMediaSubsession" subclass.)<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal"><span class="apple-style-span"><span style="font-size:13.5pt;font-family:"Helvetica","sans-serif";color:black">Ross Finlayson</span></span><span style="font-size:13.5pt;font-family:"Helvetica","sans-serif";color:black"><br>
<span class="apple-style-span">Live Networks, Inc.</span><br>
<span class="apple-style-span"><a href="http://www.live555.com/">http://www.live555.com/</a></span></span>
<o:p></o:p></p>
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