[FFmpeg-user] decoding raw RTP?

[Amit Dor-Shifer]

On Wed, Nov 2, 2011 at 10:33 AM, Amit Dor-Shifer
<amit.dor.shifer at gmail.com>wrote:

> Attached is a sample of mulaw-encoded audio, packeted as RTP. Is there a
> way for ffmpeg to decode this?
>
> $ ffmpeg -f rtp -acodec pcm_mulaw -i testing-123.mulaw.rtp.sample
> /tmp/out.wav
> ffmpeg version N-34304-gc0dbab9, Copyright (c) 2000-2011 the FFmpeg
> developers
>   built on Oct 31 2011 17:54:58 with gcc 4.4.5
>   configuration: --disable-vaapi --enable-libtheora --enable-libvorbis
> --enable-nonfree --enable-postproc --enable-version3 --enable-libspeex
>   libavutil    51. 22. 0 / 51. 22. 0
>   libavcodec   53. 26. 0 / 53. 26. 0
>   libavformat  53. 18. 0 / 53. 18. 0
>   libavdevice  53.  4. 0 / 53.  4. 0
>   libavfilter   2. 45. 2 /  2. 45. 2
>   libswscale    2.  1. 0 /  2.  1. 0
> [rtp @ 0x1a2c8c0] Guessing on RTP content - if not received properly you
> need an SDP file describing it
> [rtp @ 0x1a2c8c0] Estimating duration from bitrate, this may be inaccurate
> Input #0, rtp, from 'testing-123.mulaw.rtp.sample':
>   Duration: N/A, bitrate: N/A
>     Stream #0:0: Audio: pcm_mulaw, 8000 Hz, 1 channels, s16, 64 kb/s
> Output #0, wav, to '/tmp/out.wav':
>   Metadata:
>     encoder         : Lavf53.18.0
>     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1
> channels, s16, 128 kb/s
> Stream mapping:
>   Stream #0.0 -> #0.0 (pcm_mulaw -> pcm_s16le)
> Press [q] to stop, [?] for help
> size=       0kB time=00:00:00.00 bitrate=   0.0kbits/s
> video:0kB audio:0kB global headers:0kB muxing overhead inf%
>
> ffmpeg -f rtp -acodec pcm_mulaw -i testing-123.mulaw.rtp.sample
>

re-posting. Anyone?