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Hi Ross,<br>
<br>
I've noticed that bit rate is constant 64kb/s when streaming a-law<br>
from file, but 57-61 kb/s when streaming from live source. <br>
<br>
Felix.<br>
<br>
<br>
On 02/20/2013 06:10 PM, Felix Radensky wrote:
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Hi Ross,<br>
<br>
Thanks a lot for answering.<br>
<br>
On 02/20/2013 02:51 AM, Ross Finlayson wrote:
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cite="mid:E9FF4552-371A-4BA4-BBA2-40FA81A67446@live555.com"
type="cite">
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<div bgcolor="#ffffff" text="#000000">I have problems
streaming live PCM audio. Audio comes either directly<br>
from microphone (16-bit LE) or from hardware encoder
(A-LAW).</div>
</blockquote>
<div><br>
</div>
The problem is that a-law audio is *not* PCM, and therefore
has a different RTP payload format (if a-law audio is what
you're sending).</div>
<div><br>
</div>
<div>Specifically, if you're streaming a-law audio, then when
you create your "SimpleRTPSink" object, the
"rtpPayloadFormatName" parameter should be "PCMA" (and, of
course, the "sdpMediaTypeString" parameter will be "audio").
Because a-law audio is 8-bits per sample, you don't do any
byte swapping.</div>
<div><br>
</div>
<div>If, on the other hand, you are converting the audio from
(8-bit) a-law to (16-bit) PCM before streaming it, then you
need to (1) make sure that the 16-bit audio is in big-endian
order, and (2) use "L16" as the "rtpPayloadFormatName"
parameter when you create your "SimpleRTPSink".</div>
</blockquote>
<br>
When streaming a-law I do set mime type to "PCMA",
"sdpMediaTypeString" parameter to "audio" and payloadFormatCode to
8, as <br>
it's done in testWAVAudioStreamer. VLC detects the stream as PCM
ALAW, so I don't think that is my problem. However there's a
strong<br>
echo and noise in my stream, which do not exist when streaming
from file.<br>
<blockquote
cite="mid:E9FF4552-371A-4BA4-BBA2-40FA81A67446@live555.com"
type="cite">
<div><br>
</div>
<div>
<blockquote type="cite">
<div bgcolor="#ffffff" text="#000000">I know that audio is
correct, because if I save it to file instead of sending
to<br>
RTPSink, convert to WAV and stream via
testWAVAudioStreamer, VLC reproduces<br>
it correctly.</div>
</blockquote>
<div><br>
</div>
That works because (presumably) there's an appropriate header
in the WAV file that tells "testWAVAudioStreamer" what kind of
audio this is. If you run "testRTSPClient" on the stream, and
look at the SDP description, you'll see the proper RTP payload
format name name ("PCMA" or "L16") for this audio.</div>
</blockquote>
<br>
After saving a-law samples to file I use the following command to
convert it to WAV:<br>
<br>
sox -b 8 -e a-law -r 8000 -c 1 test.raw test.wav<br>
<br>
Maybe something is wrong with the way I bundle samples for RTP
sink, or the way I time-stamp them ?<br>
WAVAudioFileStreamer class tries to read several samples from the
file, up to 20 ms, but I already get<br>
a buffer of 160 bytes, which is exactly 20ms for a-law. I set
fPresentationTime on that buffer. <br>
Should I play with OutPacketBuffer::maxSize ?<br>
<br>
Thanks a lot for your help so far.<br>
<br>
Felix.<br>
<br>
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</blockquote>
<br>
<div class="moz-signature">-- <br>
Felix.
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