[MPlayer-users] stream with AAC plus version 2 codec - how to recognise it ?
Goga777
goga777 at bk.ru
Wed Apr 10 15:24:21 CEST 2013
hi
is it possible to recognize in mplayer logs that stream is using AACplus v2 ?
for example - does it stream use aacplus v2 ?
mplayer http://92.241.191.130/stream/2118/ -v
MPlayer 1.1-4.7 (C) 2000-2012 MPlayer Team
CPU vendor name: GenuineIntel max cpuid level: 5
CPU: Intel(R) Pentium(R) 4 CPU 3.00GHz (Family: 15, Model: 3, Stepping: 4)
extended cpuid-level: 8
extended cache-info: 67133504
Detected cache-line size is 64 bytes
Testing OS support for SSE... yes.
Tests of OS support for SSE passed.
CPUflags: MMX: 1 MMX2: 1 3DNow: 0 3DNowExt: 0 SSE: 1 SSE2: 1 SSSE3: 0
Compiled with runtime CPU detection.
get_path('codecs.conf') -> '/home/goga/.mplayer/codecs.conf'
Reading optional codecs config file /home/goga/.mplayer/codecs.conf: No such file or directory
Reading optional codecs config file /etc/mplayer/codecs.conf: No such file or directory
Using built-in default codecs.conf.
init_freetype
Using MMX (with tiny bit MMX2) Optimized OnScreenDisplay
get_path('fonts') -> '/home/goga/.mplayer/fonts'
Configuration: --prefix=/usr --confdir=/etc/mplayer --datadir=/usr/share/mplayer --enable-xmga --enable-mga --enable-joystick --disable-tremor-internal --enable-libopencore_amrnb --enable-libopencore_amrwb --disable-openal --disable-decoder=amrnb --language=all --enable-menu --disable-libdvdcss-internal --disable-dvdread-internal --enable-radio --enable-radio-capture --extra-libs=-ldl -lvorbisenc -lvorbis --enable-xvmc --with-xvmclib=XvMCW --enable-win32dll --enable-tdfxfb --enable-s3fb --disable-gui --enable-runtime-cpudetection
CommandLine: 'http://92.241.191.130/stream/2118/' '-v'
Using nanosleep() timing
get_path('input.conf') -> '/home/goga/.mplayer/input.conf'
Reading optional input config file /home/goga/.mplayer/input.conf: No such file or directory
Parsing input config file /etc/mplayer/input.conf
Input config file /etc/mplayer/input.conf parsed: 92 binds
get_path('.conf') -> '/home/goga/.mplayer/.conf'
Playing http://92.241.191.130/stream/2118/.
get_path('sub/') -> '/home/goga/.mplayer/sub/'
Filename for url is now http://92.241.191.130/stream/2118/
Filename for url is now http://92.241.191.130/stream/2118/
STREAM_HTTP(1), URL: http://92.241.191.130/stream/2118/
Connecting to server 92.241.191.130[92.241.191.130]: 80...
--- HTTP DEBUG HEADER --- START ---
protocol: [ICY]
http minor version: [0]
uri: [(null)]
method: [(null)]
status code: [200]
reason phrase: [OK]
body size: [0]
Fields:
0 - icy-notice1:<BR>This stream requires <a href="http://www.winamp.com/">Winamp</a><BR>
1 - icy-notice2:Moskva audio streamer
2 - icy-name:Moskva audio streamer
3 - content-type:audio/aacp
4 - icy-pub:1
5 - icy-br:32
--- HTTP DEBUG HEADER --- END ---
Name : Moskva audio streamer
Public : yes
Bitrate: 32kbit/s
Cache size set to 0 KBytes
STREAM: [null] http://92.241.191.130/stream/2118/
STREAM: Description: http streaming
STREAM: Author: Bertrand, Albeau, Reimar Doeffinger, Arpi?
STREAM: Comment: plain http
Stream not seekable!
Stream not seekable!
demux_aac_probe, INIT: 0, PROBED: 1617, cnt: 8
AAC file format detected.
==> Found audio stream: 0
get_path('sub/') -> '/home/goga/.mplayer/sub/'
==========================================================================
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
dec_audio: Allocating 1536000 + 131072 = 1667072 bytes for output buffer.
FFmpeg's libavcodec audio codec
libavcodec version 54.23.100 (internal)
Configuration: --enable-gpl --enable-postproc
INFO: libavcodec "aac" init OK!
AUDIO: 44100 Hz, 2 ch, s16le, 0.0 kbit/0.00% (ratio: 0->176400)
Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio))
==========================================================================
Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??...
[libaf] Adding filter dummy
[dummy] Was reinitialized: 44100Hz/2ch/s16le
[dummy] Was reinitialized: 44100Hz/2ch/s16le
Trying preferred audio driver 'alsa', options '[none]'
alsa-init: requested format: 44100 Hz, 2 channels, 9
alsa-init: using ALSA 1.0.25
alsa-init: setup for 1/2 channel(s)
alsa-init: using device default
alsa-init: opening device in blocking mode
alsa-init: device reopened in blocking mode
alsa-init: got buffersize=32768
alsa-init: got period size 1024
alsa: 48000 Hz/2 channels/4 bpf/32768 bytes buffer/Signed 16 bit Little Endian
AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample)
AO: Description: ALSA-0.9.x-1.x audio output
AO: Author: Alex Beregszaszi, Zsolt Barat <joy at streamminister.de>
AO: Comment: under development
Building audio filter chain for 44100Hz/2ch/s16le -> 48000Hz/2ch/s16le...
[dummy] Was reinitialized: 44100Hz/2ch/s16le
[libaf] Adding filter lavcresample
[dummy] Was reinitialized: 48000Hz/2ch/s16le
[dummy] Was reinitialized: 48000Hz/2ch/s16le
Video: no video
Freeing 0 unused video chunks.
Starting playback...
[libaf] Reallocating memory in module lavcresample, old len = 0, new len = 34974
Increasing filtered audio buffer size from 0 to 34932
A: 0.1 (00.0) of 0.0 (unknown) ??,?%
[aac @ 0x8c0a120]Reserved SBR extensions are not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x8c0a120]If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
A: 0.1 (00.1) of 0.0 (unknown) ??,?%
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