Date: Mon, 19 Sep 2011 13:12:42 +0700 From: Victor Sudakov <vas@mpeks.tomsk.su> To: freebsd-multimedia@freebsd.org Subject: Re: /dev/dsp to RTP Message-ID: <20110919061242.GA50407@admin.sibptus.tomsk.ru> In-Reply-To: <20110916045027.GA95062@admin.sibptus.tomsk.ru> References: <20110912074323.GA81311@admin.sibptus.tomsk.ru> <20110913193919.GQ3098@funkthat.com> <20110916045027.GA95062@admin.sibptus.tomsk.ru>
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Victor Sudakov wrote: > > Multicasting with ffmpeg works fine. The command line > ffmpeg -i file.mp3 -acodec copy -f rtp rtp://239.8.8.8:5000 -re > > does send a multicast stream which can be listened to with VLC (but > not mplayer for some reason) on multiple hosts. > > Now I need to figure out how to stream live sound from /dev/dsp. All > my attemps to record sound from a USB audio interface, as simple as > > ffmpeg -f oss -i /dev/dsp1 out.wav > > have resulted so far in a severely distorted growl instead of normal > voice. Do you know how to figure out the sampling rate and other > parameters of the sound card? "cat /dev/sndstat" does not output > anything really useful. > > The audio interface is not to blame because I use it all the time with > linphone for SIP calls. I have tried with a different soundcard and the following command line: ffmpeg -f oss -i /dev/dsp -acodec mp2 -f rtp rtp://239.8.8.8:5000 -re seems to work fine. However, the delay of voice is about 2-3 seconds. If I use the libmp3lame codec instead of mp2, the voice quality degrades. I don't know what the problem with the first audio interface is, so that linphone works fine but ffmpeg records distorted sounds. -- Victor Sudakov, VAS4-RIPE, VAS47-RIPN sip:sudakov@sibptus.tomsk.ru
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