> Audio is getting on to the network but in a bad state with many pops ticks
> and glitches, and a lot of variable latency.
This suggests that you're probably not packing PCM audio samples into outgoing
RTP packets properly. Make sure that your source's "doGetNextFrame()"
implementation is pac
> Probably increasing of the send buffer (SO_SNDBUF) can help
Yes, you can call "increaseSendBufferTo()" in your application. However,
you'll still get data loss if your stream's bitrate exceeds the capacity of
your TCP connection (which is *not* the same as the nominal bitrate of your
network
Hi All -
I have a project being built on Live555 to stream PCM audio from an ASIO
device to an RTP stream in as close to real-time as possible. Latency is
biggest concern, next-up would be quality of the audio reaching the
network.
The project so far is built on Windows / VS2012 using the RtAudio
--- Begin Message ---
Thank you very much for detailed explanation.
Probably increasing of the send buffer (SO_SNDBUF) can help, it's value depends
on estimatedBitrate from RTPSink:
if (rtpSink != NULL && rtpSink->estimatedBitrate() > 0) streamBitrate =
rtpSink->estimatedBitrate();
if (rtpGroup
> I tried to fix the problem by this way:
>
> - sendResult = send(socketNum, (char const*)(&data[numBytesSentSoFar]),
> numBytesRemainingToSend, 0/*flags*/);
> +
> + do {
> + sendResult = send(socketNum, (char
> const*)(&data[numBytesSentSoFar]), numBytesRemainingToSend
