I am trying to record 55 seconds of radiostation.
ffmpeg -t 55 -i http://19233.live.streamtheworld.com/BLZE_1.mp3 toofastrecord.mp3 .
FFMPEG do this nearly for 10 seconds. How does it possible to have speed speed=5.67x instead 1 for live recording?
Below is FFMPEG output.
ffmpeg -t 55 -i http://19233.live.streamtheworld.com/BLZE_1.mp3 toofastrecord.mp3
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.2.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 12.100 / 55. 12.100
libavcodec 57. 21.100 / 57. 21.100
libavformat 57. 21.101 / 57. 21.101
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 23.100 / 6. 23.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
[mp3 @ 0000017b0ad9aa00] Skipping 0 bytes of junk at 0.
Input #0, mp3, from 'http://19233.live.streamtheworld.com/BLZE_1.mp3':
Metadata:
icy-br : 64
icy-description :
icy-genre : Talk
icy-name :
icy-url :
Duration: N/A, start: 0.000000, bitrate: 64 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, mono, s16p, 64 kb/s
Output #0, mp3, to 'toofastrecord.mp3':
Metadata:
icy-br : 64
icy-description :
icy-genre : Talk
icy-name :
icy-url :
TSSE : Lavf57.21.101
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, mono, s16p
Metadata:
encoder : Lavc57.21.100 libmp3lame
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
size= 430kB time=00:00:55.01 bitrate= 64.1kbits/s speed=5.67x
video:0kB audio:430kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.077898%
Your source is definitely live. You can prove this simply by listening to it. When you reload it, it's all new audio.
What's happening is that the server has a relatively large buffer, where it buffers data from the source. When new clients connect, they require a few frames of audio data before decoding can begin. And, most players have a playback buffer of at least two seconds. Therefore, the server has this large buffer and can flush the whole thing upon connect, meaning a listener doesn't have to wait two seconds to buffer two seconds of audio. The playback can begin immediately.
For this particular stream, it looks to be about a 384KB buffer. This stream has a low bitrate of 64kbps. That's 8KB per second, meaning that initial buffer has a whole 48 seconds of buffered audio in it.
You're only recording for 55 seconds, so this can run as fast as the buffer can be flushed to you. If you were to record for a few more minutes, you would see that the speed drops dramatically once the first 64 seconds are downloaded.