[Solved]ffmpeg - default audio doesn't exist, /dev/dsp error

EDIT: See my third post. Finally got a command that works

---------- original post ----------------------------------------------
I've been searching around for a way to use a Pogoplug as a live audio streamer for a baby monitor. Basically, either using a C-media USB audio adapter or by using the microphone on a USB webcam.

I came across a post about using ffmpeg to stream audio via rtp using this command:

$this->bbcode_second_pass_quote('', 'f')fmpeg -f oss -i /dev/dsp -acodec libmp3lame -ab 32k -ac 1 -re -f rtp rtp://localhost:1234

[permitivity]

Oh, when I do this, this is the output:

$this->bbcode_second_pass_quote('', '[')root@alarm ~]# cat /proc/asound/pcm
00-00: USB Audio : USB Audio : playback 1 : capture 1

[moonman]

With the command above you are trying to use oss driver (old and is not really used anymore) and /dev/dsp which doesn't exist. Use alsa and the something like hw:0.

So it would be -f alsa -i hw:0

EDIT 2: FINALLY GOT A COMMAND THAT WORKS. SEE LATER POST.


Ok...

I've tried these lines with these results:

$this->bbcode_second_pass_quote('', '
')ffmpeg -f alsa -i hw:0 -acodec libmp3lame -ab 32k -ac 1 -re -f rtp rtp://localhost:1234

hw:0: Input/output error

[moonman]

Have you tried 0,1 ? Device 0, subdevice 1.

EDIT. NVM, it should be 0,0. Check ffmpeg manual on how to capture

[permitivity]

When I run "aplay -l", it tells me it's "device 0".

So, randomly pasting ffmpeg commands I see on google, I finally run into something that seems to be doing something!!!

$this->bbcode_second_pass_quote('', 'f')fmpeg -re -f alsa -i plughw:0 -ar 8000 -f mulaw -f rtp rtp://224.168.2.105:1234

[permitivity]

Ok, got a command that works:

$this->bbcode_second_pass_quote('', 'f')fmpeg -re -f alsa -i plughw:0 -acodec libmp3lame -ab 32k -ac 1 -f rtp rtp://224.168.2.105:1234

[moonman]

bitrate 32 is what makes it sound so terrible. Reduce the sampling rate from 44.1 to say 11 or increase bitrate

Yup, you're right. Here's two commands using different codecs. Increased bitrate and sampling rate. On my system, it takes about 26kBps instead of the 6kBps of the old command above. I thought mp3 would need less bandwidth, but they seem about the same.

I also took out the Syba usb audio card. I was using a 3.5mm microphone on the mic jack. Lots of noise. Using the mic from a USB webcam works much better.

$this->bbcode_second_pass_quote('', '
')ffmpeg -re -f alsa -i plughw:0 -acodec mp2 -ab 192000 -ar 44100 -ac 1 -f rtp rtp://224.168.2.105:1234
ffmpeg -re -f alsa -i plughw:0 -acodec libmp3lame -ab 192000 -ar 44100 -ac 1 -f rtp rtp://224.168.2.105:1234

[moonman]

You actuallly need to lower sampling rate when you use low bitrate. I would also try 48 khz sampling rate since this maybe native to the hardware. Also 64 should be enough since the mic input is mono (equivalent to 128 stereo, unless your mic is stereo). How's the cpu load? Its doing realtime encoding afterall.