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I try to stream live audio using ffmpeg and external usb microphone. I followed this nearly tutorial : http://prupert.co.uk/2010/08/02/stream-live-audio-from-a-microphone-in-near-real-time-in-ubuntu.

I had to adapt some steps but finally, I achieved to receive the stream my laptop using this command :
ffmpeg -f oss -i /dev/dsp1 -acodec libmp3lame -ab 32k -ac 1 -re -f rtp rtp://192.168.28.116:1234

The cpu is near 100%, i have a bad sound during two seconds and after nothing... I see in Wireshark that the board is sending frames continuously.

Does someone have ideas to lower the cpu usage ?

Thanks !

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You may want to look into overclocking and a different memory split. If you have the choice of codec you may be able to achieve some improvement by switching. You may also want to investigate compression of the stream - though this may be a zero some gain - anything you gain in transfer time may be lost in compression. – Steve Robillard Aug 5 '12 at 22:46
Did you try writing to a local file first? I'd try to go step by step: 1) realtime recording to a wave file, 2) realtime recording to an mp3 file 3) streaming stuff over the network – pixelistik Aug 6 '12 at 9:36
not at the moment. seems to be complicated to do all thoses steps manually instead of full automatic ffmpeg ;-) – si2w Aug 6 '12 at 9:51
What are you recording? – Alex Chamberlain Aug 6 '12 at 11:25
I'm trying to make an ethernet babyphone – si2w Aug 6 '12 at 21:45

3 Answers

up vote 3 down vote

ALSA Input

One alternative is to go via ALSA. A similar command to above would be

ffmpeg -ac 1 -f alsa -i hw:0,0 -acodec libmp3lame -ab 32k -ac 1 -re -f rtp rtp://localhost:1234

I'm not sure how this will affect the CPU usage.

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I tried : ffmpeg -ac 1 -f alsa -i hw:1,0 -acodec libmp3lame -ab 32k -ac 1 -re -f rtp rtp://192.168.1.116:1234 But ffmpeg crashes after somes seconds : [alsa @ 0x7a1510] ALSA buffer xrun. Any other ideas ? – si2w Aug 10 '12 at 21:48
I tested ffmpeg -f oss -i /dev/dsp1 -ab 32k -ac 1 -re -f rtp rtp://192.168.28.116:1234 but the sound is very bad... on the lag is around 6 seconds. Any idea to have a better solution ? – si2w Aug 10 '12 at 22:27
@si2w I think 32k for mp3 is a bit of a joke. Either up that, or try a different codec. Maybe one of the G2xx series designed for telephones – Alex Chamberlain Aug 11 '12 at 7:47
up vote 2 down vote

this works without the cpu usage problem.

ffmpeg -f alsa -i default:CARD=U0x46d0x819 -acodec mp2 -ac 1 -re -f rtp rtp://234.5.5.5:1234 2> /tmp/mylog.log &

replace default:CARD=U0x46d0x819 with your microphone id,( obtained from arecord -l ) or use hw:0,0 (or whichever device it is).

I had similar problem, mp3 encoding takes up 90%+ of cpu power and just could not keep up with the audio, so I changed it to mp2 encoding instead. This uses about 15-18% of cpu (top) and streams smoothly to vlc on my lan. It would make a perfect baby monitor, or whatever. there is just a second or so delay, which is the buffering at the vlc end.
note: the ip address is a multicast address ([224-239].x.y.z). you don't have to target it at a particular network device on your lan; and your broadband router will keep the traffic local (by default).

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up vote 1 down vote

To answer your specific question, you can reduce CPU by piping arecord into ffmpeg:

arecord -f cd -D plughw:1,0 | ffmpeg -i - -acodec libmp3lame -ab 32k -ac 1 -re -f rtp rtp://234.5.5.5:1234

You'll need to replace plughw:1,0 with your specific sound card. See arecord -l for info. On my Rasp Pi it went from ~95% CPU to ~35%.

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