[FFmpeg-user] Suggestions for making timeshifted audio less bandwidth intensive

Hereward Cooper coops at fawk.eu
Fri Aug 21 05:53:00 CEST 2015

I've a ffserver setup that's buffering and streaming radio, so that it can
be listened to timeshifted.

When I use the timeshift feature, it correctly starts the stream at the
requested point in history.

However I've just noticed that on the ffserver stats page the "bytes
transferred" shows a huge amount. What seems to be happening is that when a
client connects, they are downloading the entire stream at once (~8 hours
of audio), from the timeshifted point right through to the current point in

This is resulting in about 400MB being downloaded in the first minute or so
that a client connects.

Is there a way to make this stream less bandwidth intensive, i.e. to make
clients on download chunks of the stream as they need it?

Currently I'm using a mp2/libmp3lame stream. Would switching the stream to
another format achieve what I want?

<Feed radio.ffm>
    File /tmp/radio.ffm
    FileMaxSize 1G
    Launch ./ffmpeg -f lavfi -i nullsrc -i http://internet/radiostream
    ACL allow localhost

<Stream radio.mp3>
    Feed radio.ffm
    Format mp2
    AudioCodec libmp3lame
    AudioBitRate 64
    AudioChannels 1
    AudioSampleRate 44100




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