[FFmpeg-user] Streaming audio input in a lossless format

[Alex Becerra]

Of what configuration?

I attach the output using pcm_s16le. VLC gives me problem with m=audio 1234
RTP/AVP 97. It says an error in SDP with that number 97.

root at server:~# ffmpeg -f alsa -i hw:0 -acodec pcm_s16le -f rtp rtp://
234.5.5.5:1234
ffmpeg version 0.8.3-6:0.8.3-7, Copyright (c) 2000-2012 the Libav developers
  built on Aug 25 2012 13:25:50 with gcc 4.7.1
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a
future release. Please use avconv instead.
[alsa @ 0x9701bc0] Estimating duration from bitrate, this may be
inaccurate
Input #0, alsa, from
'hw:0':

  Duration: N/A, start: 123733.999989, bitrate: N/A
    Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
[rtp @ 0x97022e0] Cannot respect max delay: frame size = 0
Output #0, rtp, to 'rtp://234.5.5.5:1234':

  Metadata:
    encoder         : Lavf53.21.0
    Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 234.5.5.5
t=0 0
a=tool:libavformat 53.21.0
m=audio 1234 RTP/AVP 97
b=AS:200

Press ctrl-c to stop encoding



2012/10/2 Roger Pack <rogerdpack2 at gmail.com>

> > There's an standard process (or series of steps) to do an streaming with
> a
> > lossless format using the line in input of the sound card?
> > MP3 coding works perfectly but I need the lossless data. The command used
> > is "ffmpeg -f alsa -i hw:0 -acodec libmp3lame -ar 11025 -ac 1 -f rtp
> rtp://
> > 234.5.5.5:1234" and I opened it without problems using VLC player.
> > I've tried to many configurations changing the codec to pcm_s16le or flac
> > but none of them worked, also I've tried with udp protocol without good
> > results. Some of them showed an error in VLC (SDP related) and many other
> > works (the streaming starts) but VLC didn't reproduce anything (only
> > silence)
>
> what's your full output?
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