[Live-devel] Live G711 audio source

[Ross Finlayson]

>Here's how I implemented the subsession methods:
>
>     RTPSink * LiveG711MediaSubsession::createNewRTPSink(Groupsock 
>*rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource 
>*inputSource)
>     {
>         char const* mimeType = "PCMU";
>         unsigned char payloadFormatCode = 0;
>         int sampleFrequency = 8000;
>         unsigned int numChannels = 1;
>
>         return SimpleRTPSink::createNew(envir(), rtpGroupsock,
>                     payloadFormatCode, sampleFrequency,
>                     "audio", mimeType, numChannels);
>     }

If your input data is *already* u-law audio (i.e., 8-bits-per 
sample), then this should work.  (If, instead, it's 16-bit-per-sample 
PCM audio, then you need to insert a filter to convert it to u-law.)


>The "LiveG711AudioStreamFramer is also something I wrote, and 
>"m_mediaSource" is the thing that supplies my audio.  For the values 
>I supply to SimpleRTPSink, I basically copied what 
>WAVAudioFileServerMediaSubsession was delivering for PCMU audio 
>(8000 hz, 1 channel, etc.). 
>
>Still not sure what problem I'm having--I see VLC is receiving the 
>audio stream, but it's dropping all of the data claiming the "PTS is 
>out of range."

Ahh!  The problem here is probably that you're not setting 
"fPresentationTime" properly in your 'framer' class.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/