[Live-devel] Darwin Streaming Server and Audio Problem

[Keith]

Well, I think I have found at least one problem and I am now writing a good
.wav file.  It was operator error and that I was only writing half the
samples (160 bytes) instead of (320).  Some logic in my part was assuming 8
bit output.

- Keith

> -----Original Message-----
> From: live-devel-bounces at ns.live.com 
> [mailto:live-devel-bounces at ns.live.com] On Behalf Of Keith
> Sent: Friday, February 18, 2005 8:02 AM
> To: 'LIVE.COM Streaming Media - development & use'
> Subject: RE: [Live-devel] Darwin Streaming Server and Audio Problem
> 
> 
> 
> > How, specifically, are you trying to use the "LIVE.COM
> > Streaming Media" 
> > software?  Are you running "openRTSP", to record a RTSP/RTP 
> > audio stream to 
> > a file?
> 
> I started with the openRTSP example.  I have modified it so 
> that instead of recording the streams to a file I get the 
> streams in the function afterGettingFrame callback.  When I 
> get an audio packet data, I decode it and write it to a wav 
> file.  I know that it is 8000 khz, mono and 16 bit because I 
> control that on the encoding side so I create the correct wav 
> header. The packet is encoded with Speex Narrowband and I 
> just decode each packet as it comes in and write it to the 
> wav file.  When I stop the encode, I update the sizes in the 
> wav file.  If you listen to the sample I pointed to in the 
> previous email, you can see that I am creating a good .wav 
> file with the decoded data I am receiving.  However, the 
> audio does not sound correct as is it does when I listen to 
> it with the QuickTimePlayer and my Speex decoder there.  One 
> thing I am assumming at this time is that the packets are 
> coming in order and none are dropped when I get the call in 
> afterGettingFrame.  Could this be my problem? I just thought 
> that by listening to the audio, someone would have a clue to 
> what I need to do.
> 
> > Note that 16-bit samples in PCM audio RTP streams are
> > *big-endian*, by 
> > definition, whereas in WAV files, they're little endian.  
> If you use 
> > "openRTSP" to record a PCM RTSP/RTP audio stream into a file, 
> > and then just 
> > add a WAV file header on the front, the audio won't be 
> > correct.  Instead, 
> > you'll need to byte-swap the audio samples before writing 
> > them to a WAV file.
> 
> I am encoding the audio stream with my QT Speex component and 
> broadcasting with QT to a DSS.  I did try swapping the buffer 
> after the decode, but all I got was garbage sound then.  I 
> think I have the correct endian as the audio in the sample 
> wav file sounds very close to what it should be.  Maybe I am 
> wrong there too.
> 
> - Keith
> 
> 
> 
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