Hi,
I need some help with using avconv/avplay for simple audio streaming
over wireless network.
I have an Arietta G25 with ARM9@400Mhz CPU with linux and I would like
to use this device for streaming audio over wifi network.
On this embedded linux I use avconv to encode the audio into mpeg stream
and send it over rtp to multicast address:
# /usr/bin/avconv -f oss -i /dev/dsp -ar 22050 -ab 64k -ac 1
-filter_complex 'volume=50' -f mpeg rtp://224.0.0.1:1234/
avconv version 11.4-6:11.4-1~deb8u1, Copyright (c) 2000-2014 the Libav
developers
built on Jun 4 2015 20:16:59 with gcc 4.9.2 (Debian 4.9.2-10)
[oss @ 0x11d1e0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, oss, from '/dev/dsp':
Duration: N/A, start: 1167613970.360718, bitrate: N/A
Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Output #0, mpeg, to 'rtp://224.0.0.1:1234/':
Metadata:
encoder : Lavf56.1.0
Stream #0.0: Audio: mp2, 22050 Hz, mono, s16, 64 kb/s
Metadata:
encoder : Lavc56.1.0 mp2
Stream mapping:
Stream #0:0 (pcm_s16le) -> volume
volume -> Stream #0:0 (mp2)
Press ctrl-c to stop encoding
Non-monotonous DTS in output stream 0:0; previous: 1225224, current:
1224159; changing to 1225225. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 2886678, current:
2886580; changing to 2886679. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 2886820, current:
2886302; changing to 2886821. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 4550980, current:
4550020; changing to 4550981. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 5541698, current:
5540641; changing to 5541699. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 5541699, current:
5541588; changing to 5541700. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6023020, current:
6020335; changing to 6023021. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6023021, current:
6022424; changing to 6023022. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6506967, current:
6506029; changing to 6506968. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6984910, current:
6984229; changing to 6984911. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6984911, current:
6984253; changing to 6984912. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6984912, current:
6983143; changing to 6984913. This may result in incorrect timestamps in
the output file.
Non-monotonous DTS in output stream 0:0; previous: 6984913, current:
6984543; changing to 6984914. This may result in incorrect timestamps in
the output file.
size= 656kB time=83.02 bitrate= 64.7kbits/s
On the laptop I use avplay to listen:
zim@laptop ~ $ avplay -f mpeg -i rtp://224.0.0.1:1234
avplay version 9.18-6:9.18-0ubuntu0.14.04.1, Copyright (c) 2003-2014 the
Libav developers
built on Mar 16 2015 13:19:10 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
[mpeg @ 0x7efef8005be0] max_analyze_duration reached
[mpeg @ 0x7efef8005be0] Estimating duration from bitrate, this may be
inaccurate
Input #0, mpeg, from 'rtp://224.0.0.1:1234':
Duration: N/A, start: 1.000000, bitrate: 64 kb/s
Stream #0.0[0x1c0]: Audio: mp2, 22050 Hz, 2 channels, s16p, 64 kb/s
[mp2 @ 0x7efef8004b40] Header missingvq= 0KB sq= 0B f=0/0
103.42 A-V: 0.000 s:0.0 aq= 0KB vq= 0KB sq= 0B f=0/0
Unfortunately, after ~10-20 seconds audio queue (aq) comes to zero kB
and audio starts to interrupt constantly.
I think it's because the embedded system is overloaded, but as far as I
can see, the avconv is using 100% cpu and the load is higher than 1.
I tried to use ac3 codec, but it didn't help.
Also changing the bitrate or audio sampling rate didn't help. avconv
still eats almost 100% cpu power, no matter how I use it.
Do you have any idea what could be changed?
This device I am using is not so powerful, but it should easily handle
this kind of tasks.
Thank you.
R.
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