Unable to convert MP3 file to PCM with ffmpeg - ffmpeg

I have compiled ffmpeg to convert mp3 file with this config, as the ffmpeg output size is matter to me, I have disabled everything in ffmpeg:
#!/bin/bash
. abi_settings.sh $1 $2 $3
pushd ffmpeg
case $1 in
armeabi-v7a | armeabi-v7a-neon)
CPU='cortex-a8'
;;
x86)
CPU='i686'
;;
esac
make clean
./configure \
--target-os="$TARGET_OS" \
--cross-prefix="$CROSS_PREFIX" \
--arch="$NDK_ABI" \
--cpu="$CPU" \
--enable-runtime-cpudetect \
--sysroot="$NDK_SYSROOT" \
--disable-all \
--disable-network \
--enable-avcodec \
--enable-avformat \
--enable-swresample \
--enable-avfilter \
--enable-parser=mpegaudio \
--enable-demuxer=mp3,wav \
--enable-muxer=mp3,wav \
--enable-decoder=pcm*,mp3*\
--enable-encoder=pcm*,libmp3lame \
--enable-filter=aresample \
--enable-protocol=file \
--enable-ffmpeg \
--enable-libmp3lame \
--enable-pthreads \
--disable-debug \
--disable-ffserver \
--enable-version3 \
--enable-hardcoded-tables \
--disable-ffplay \
--disable-ffprobe \
--enable-gpl \
--enable-yasm \
--disable-doc \
--disable-shared \
--enable-static \
--enable-small \
--enable-nonfree \
--pkg-config="${2}/ffmpeg-pkg-config" \
--prefix="${2}/build/${1}" \
--extra-cflags="-I${TOOLCHAIN_PREFIX}/include $CFLAGS" \
--extra-ldflags="-L${TOOLCHAIN_PREFIX}/lib $LDFLAGS" \
--extra-libs="-lm" \
--extra-cxxflags="$CXX_FLAGS" || exit 1
make -j${NUMBER_OF_CORES} && make install || exit 1
popd
but when I want to convert mp3 to pcm, I get this error:
configuration: --target-os=linux --cross-prefix=/root/ffmpeg2/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi- --arch=arm --cpu=cortex-a8 --enable-runtime-cpudetect --sysroot=/root/ffmpeg2/ffmpeg-android/toolchain-android/sysroot --disable-all --disable-network --enable-avcodec --enable-avformat --enable-swresample --enable-avfilter --enable-parser=mpegaudio --enable-demuxer='mp3,wav' --enable-muxer='mp3,wav' --enable-decoder='pcm*,mp3*--enable-encoder=pcm*,libmp3lame' --enable-filter=aresample --enable-protocol=file --enable-ffmpeg --enable-libmp3lame --enable-pthreads --disable-debug --disable-ffserver --enable-version3 --enable-hardcoded-tables --disable-ffplay --disable-ffprobe --enable-gpl --enable-yasm --disable-doc --disable-shared --enable-static --enable-small --enable-nonfree --pkg-config=/root/ffmpeg2/ffmpeg-android/ffmpeg-pkg-config --prefix=/root/ffmpeg2/ffmpeg-android/build/armeabi-v7a --extra-cflags='-I/root/ffmpeg2/ffmpeg-android/toolchain-android/include -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all' --extra-ldflags='-L/root/ffmpeg2/ffmpeg-android/toolchain-android/lib -Wl,-z,relro -Wl,-z,now -pie' --extra-libs=-lm --extra-cxxflags=
libavutil 55. 17.103 / 55. 17.103
libavcodec 57. 24.102 / 57. 24.102
libavformat 57. 25.100 / 57. 25.100
libavfilter 6. 31.100 / 6. 31.100
libswresample 2. 0.101 / 2. 0.101
[mp3 # 0xb72041d0] Skipping 0 bytes of junk at 0.
[mp3 # 0xb72041d0] Estimating duration from bitrate, this may be inaccurate
[mp3 # 0xb72041d0] Could not find codec parameters for stream 0 (Audio: mp3, 44100 Hz, 1 channels, 256 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
/storage/emulated/0/Android/data/com.edspace.alefba/files/alefba/page/voices/109/cource_1.mp3: could not find codec parameters
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, mp3, from '/storage/emulated/0/Android/data/com.edspace.alefba/files/alefba/page/voices/109/cource_1.mp3':
Duration: 00:00:08.67, start: 0.000000, bitrate: 256 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, 1 channels, 256 kb/s
[NULL # 0xb7242370] Requested output format 's16le' is not a suitable output format
/storage/emulated/0/Android/data/com.edspace.alefba/files/alefba/page/voices/109/cource_1.pcm: Invalid
what should I change in configure parameters to make it work?
EDIT:
it seem that the problem is with output format:
Requested output format 's16le' is not a suitable output format , but I do not know how can I fix that

Change to
--enable-muxer=mp3,pcm_s16le,wav \

Related

Compiled FFmpeg not accepting -c:v and -c:a

I compiled FFmpeg with libsrt, with the online compile guide. https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu & how to compile ffmpeg with enabling libsrt
It seems to compile correctly.
ffmpeg version N-96575-g843c24a Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ubuntu/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libsrt --enable-nonfree
libavutil 56. 38.100 / 56. 38.100
libavcodec 58. 67.100 / 58. 67.100
libavformat 58. 37.100 / 58. 37.100
libavdevice 58. 9.103 / 58. 9.103
libavfilter 7. 72.100 / 7. 72.100
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
But when running this command to convert a incoming SRT stream to HLS, it doesn't know the -c:a command. When switching the order, it runs that it doesn't know about the -c:v command.
ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316&mode=listener -c:a copy -c:v copy -strict -f hls -hls_time 4 -hls_playlist_type event stream.m3u8
~$ ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316&mode=listener -c:a copy -c:v copy -strict -f hls -hls_time 4 -hls_playlist_type event stream.m3u8
[2] 9930
ffmpeg version N-96575-g843c24a Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ubuntu/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ubuntu/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libsrt --enable-nonfree
libavutil 56. 38.100 / 56. 38.100
libavcodec 58. 67.100 / 58. 67.100
libavformat 58. 37.100 / 58. 37.100
libavdevice 58. 9.103 / 58. 9.103
libavfilter 7. 72.100 / 7. 72.100
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
-c:a: command not found
[2]+ Stopped ffmpeg -re -i srt://0.0.0.0:25000?pkt_size=1316
I have searched the issue, but I could not find anything similar.
Does someone know what I have missed in the setup?
Everything is manually compiled through the guide, this was the final command I run to compile FFmpeg:
cd ~/ffmpeg_sources && \
wget -O ffmpeg-snapshot.tar.bz2 https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 && \
tar xjvf ffmpeg-snapshot.tar.bz2 && \
cd ffmpeg && \
PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
--prefix="$HOME/ffmpeg_build" \
--pkg-config-flags="--static" \
--extra-cflags="-I$HOME/ffmpeg_build/include" \
--extra-ldflags="-L$HOME/ffmpeg_build/lib" \
--extra-libs="-lpthread -lm" \
--bindir="$HOME/bin" \
--enable-gpl \
--enable-libaom \
--enable-libass \
--enable-libfdk-aac \
--enable-libfreetype \
--enable-libmp3lame \
--enable-libopus \
--enable-libvorbis \
--enable-libvpx \
--enable-libx264 \
--enable-libx265 \
--enable-libsrt \
--enable-nonfree && \
PATH="$HOME/bin:$PATH" make && \
make install && \
hash -r
Nothing to do with ffmpeg. The shell treats 1316 as the end of the command and & as the background operator.
Enclose the full URL in quotes to avoid this.
"srt://0.0.0.0:25000?pkt_size=1316&mode=listener"

ffmpeg and streaming to youtube displaying logotype and webcamera

➜ tmp ffmpeg -version
ffmpeg version 3.1.7 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 6.3.1 (GCC) 20161221 (Red Hat 6.3.1-1)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro -specs=/usr/lib/rpm/redhat/redhat-hardened-ld' --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libcdio --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --extra-cflags=-I/usr/include/nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect
libavutil 55. 28.100 / 55. 28.100
libavcodec 57. 48.101 / 57. 48.101
libavformat 57. 41.100 / 57. 41.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 47.100 / 6. 47.100
libavresample 3. 0. 0 / 3. 0. 0
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
I'm streaming to youtube with the following settings
ffmpeg \
-f x11grab -video_size 3840x2160 -framerate 30 -i :1.0 \
-f v4l2 -video_size 320x240 -framerate 30 -i /dev/video0 \
-f alsa -ac 2 -i pulse -i logo.png -filter_complex \
"[0:v]scale=1920:-1,setpts=PTS-STARTPTS[bg]; \
[1:v]scale=120:-1,setpts=PTS-STARTPTS[fg]; \
[bg][fg]overlay=W-w-10:10[bg2]; \
[bg2][3:v]overlay=W-w-10:H-h-10,format=yuv420p[out]" \
-map "[out]" -map 2:a -c:v libx264 -preset veryfast \
-maxrate 9000k -bufsize 12000k -c:a aac -b:a 160k -ar 44100 -b:a 128k \
-f flv rtmp://a.rtmp.youtube.com/live2/TOKEN
I'm getting that huge black screen that cuts a lot of visible screen canvas. How do I get rid of it keeping webcamera preview on top and logo at the bottom?
This is what I see in console
https://gist.github.com/dmitry-saritasa/cc85ba6650d9bd50b28a61624711b96a
as you can see I only get 6-9fps while I expect 24+ (per settings?)
frame= 101 fps=6.4 q=23.0 size= 1669kB time=00:00:11.37 bitrate=1202.2kbits/s dup=0 drop=4 speed=0.718x
frame= 105 fps=6.4 q=27.0 size= 1866kB time=00:00:12.09 bitrate=1263.7kbits/s dup=0 drop=4 speed=0.74x
frame= 107 fps=6.4 q=28.0 size= 1871kB time=00:00:12.32 bitrate=1243.0kbits/s dup=0 drop=4 speed=0.732x
frame= 111 fps=6.4 q=26.0 size= 1879kB time=00:00:12.79 bitrate=1203.4kbits/s dup=0 drop=4 speed=0.737x
Questions
How to get rid of the top black area? So that I can screen share my full display?
How to increase fps?

ffmpeg: Encoder (codec amr_nb) not found for output stream #0:1

I'm using this command to convert flv to 3gp
ffmpeg -y -i in.flv -ar 8000 -b:a 12.20k -ac 1 -s 176x144 out.3gp
And it's giving this error
Encoder (codec amr_nb) not found for output stream #0:1
I have searched for amr_nb packages on yum but didn't able to find it.
I'm on CentOS 7
ffmpeg -version
ffmpeg version 2.6.8 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --enable-libfaac --enable-nonfree --enable-libfdk-aac --enable-nonfree --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
libavutil 54. 20.100 / 54. 20.100
libavcodec 56. 26.100 / 56. 26.100
libavformat 56. 25.101 / 56. 25.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 11.102 / 5. 11.102
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
I'm not sure what's missing.
Edit:
I followed this guide https://s3bubble.com/installing-ffmpeg-on-centos-6-6-in-usrlocalbin/ + alijandro's advice.
The ffmpeg installed in your system didn't support encoder amr-nb which is the default encoder for 3gp format.
See if you can find in the encoders list
$ ffmpeg -encoders |grep amr_nb
You can try
use h263/aac as the output encoder, add the option -vcodec h263 -acodec aac
or build ffmpeg with amr-nb support
get opencore source from here https://sourceforge.net/projects/opencore-amr
compile the source and install to /path/to/opencore-amr
get the source of ffmpeg
configure ffmpeg with the options
$./configure --enable-libopencore-amrnb \
--enable-libopencore-amrwb \
--enable-version3 \
...
--extra-cflags="-I/path/to/opencore-amr/include" \
--extra-ldflags="-L/path/to/opencore-amr/lib"
...
# more configure options
compile and install ffmpeg, then your command will work

find in bash to convert files with ffmpeg - paths being changed

Trying to convert a bunch of files with ffmpeg and find.
find -name "*.mkv" | while read f
do ffmpeg -i "$f" -c copy -y "${f%.*}.mp4"
done
This should work right? It results in:
+ read f
+ ffmpeg -i '/file.mkv' -c copy -y '/file.mp4'
ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
built on Mar 7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
/file.mkv: No such file or directory
So I naturally try sticking the missing dot in front of the variable:
find -name "*.mkv" | while read f
do ffmpeg -i ".$f" -c copy -y ".${f%.*}.mp4"
done
The result:
+ read f
+ ffmpeg -i '../file.mkv' -c copy -y '../file.mp4'
ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
built on Mar 7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
../file.mkv: No such file or directory
It's specifically removing the dot at the worst possible time. Any idea what's causing this and how to fix it?
PS: -print0 in find has the same problem, and none of the files have newlines etc.
Edit: As requested, the echo. The echo works as expected:
find -name "*.mkv" | while read f
do echo ffmpeg -i "$f" -c copy -y "${f%.*}.mp4"
done
+ read f
+ echo ffmpeg -i './file.mkv' -c copy -y './file.mp4'
ffmpeg -i ./file.mkv -c copy -y ./file.mp4
Edit2: With a file with a space and one without, the filename without is changed. If I rm the one with a space the one without works fine.
$ ls
+ ls
file.mkv file two.mkv
$ find -name "*.mkv" | while read f; do ffmpeg -i "$f" -c copy -y "${f%.*}.mp4"; done
+ find -name '*.mkv'
+ read f
+ ffmpeg -i './file two.mkv' -c copy -y './file two.mp4'
ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
built on Mar 7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
configuration: --arch=amd64 --enable-pthreads --enable-libopencv --enable-librtmp --enable-libopenjpeg --enable-libopus --enable-libspeex --enable-libtheora --enable-vaapi --enable-runtime-cpudetect --enable-libvorbis --enable-zlib --enable-swscale --enable-libcdio --enable-bzlib --enable-libdc1394 --enable-frei0r --enable-gnutls --enable-libgsm --enable-libmp3lame --enable-libpulse --enable-vdpau --enable-libvpx --enable-gpl --enable-x11grab --enable-libx264 --enable-filters
libavutil 52. 66.101 / 52. 66.101
libavcodec 55. 52.102 / 55. 52.102
libavformat 55. 33.101 / 55. 33.101
libavdevice 55. 11.100 / 55. 11.100
libavfilter 4. 3.100 / 4. 3.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, matroska,webm, from './file two.mkv':
Metadata:
COMPATIBLE_BRANDS: isommp42
MAJOR_BRAND : mp42
MINOR_VERSION : 0
ENCODER : Lavf55.33.101
Duration: 00:08:15.54, start: 0.000000, bitrate: 194 kb/s
Stream #0:0(und): Audio: aac, 44100 Hz, stereo, fltp (default)
Metadata:
CREATION_TIME : 2014-03-07 04:33:17
LANGUAGE : und
HANDLER_NAME : IsoMedia File Produced by Google, 5-11-2011
Output #0, mp4, to './file two.mp4':
Metadata:
COMPATIBLE_BRANDS: isommp42
MAJOR_BRAND : mp42
MINOR_VERSION : 0
encoder : Lavf55.33.101
Stream #0:0(und): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo (default)
Metadata:
CREATION_TIME : 2014-03-07 04:33:17
LANGUAGE : und
HANDLER_NAME : IsoMedia File Produced by Google, 5-11-2011
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size= 11698kB time=00:08:15.53 bitrate= 193.4kbits/s
video:0kB audio:11614kB subtitle:0 data:0 global headers:0kB muxing overhead 0.724668%
+ read f
+ ffmpeg -i /file.mkv -c copy -y /file.mp4
ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
built on Mar 7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
configuration: --arch=amd64 --enable-pthreads --enable-libopencv --enable-librtmp --enable-libopenjpeg --enable-libopus --enable-libspeex --enable-libtheora --enable-vaapi --enable-runtime-cpudetect --enable-libvorbis --enable-zlib --enable-swscale --enable-libcdio --enable-bzlib --enable-libdc1394 --enable-frei0r --enable-gnutls --enable-libgsm --enable-libmp3lame --enable-libpulse --enable-vdpau --enable-libvpx --enable-gpl --enable-x11grab --enable-libx264 --enable-filters
libavutil 52. 66.101 / 52. 66.101
libavcodec 55. 52.102 / 55. 52.102
libavformat 55. 33.101 / 55. 33.101
libavdevice 55. 11.100 / 55. 11.100
libavfilter 4. 3.100 / 4. 3.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
/file.mkv: No such file or directory
+ read f
ffmpeg, for reasons I do not know, reads from standard input. Redirect from /dev/null to avoid interfering with the read statement in the while loop.
find -name "*.mkv" | while read f
do ffmpeg -i "$f" -c copy -y "${f%.*}.mp4" < /dev/null
done
Alternatively, you could execute ffmepg on each file as it's found:
find . -type f -name "*.mkv" -exec bash -c 'ffmpeg -i "$0" -codec copy "${0%.*}.mp4"' "{}" \;

FFmpeg unable to stream

I have a simple http server for incoming MPEG stream and I'm trying to stream video from another machine on the same network. It works using ffmpeg like this
ffmpeg -f video4linux2 -i /dev/video0 -f mpeg1video http://myaddress.com:port
But from one specific computer ffmpeg refuses to stream and I can't find out why. All I can get is this output:
ffmpeg version 0.7.8, Copyright (c) 2000-2011 the FFmpeg developers
built on Sep 16 2012 09:30:22 with gcc 4.5.3
configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib
--mandir=/usr/share/man --enable-shared --cc=i686-pc-linux-gnu-gcc
--disable-static --enable-gpl --enable-postproc --enable-avfilter
--disable-stripping --disable-debug --disable-doc --disable-network
--disable-vaapi --disable-ffplay --disable-vdpau --disable-indev=v4l
--disable-indev=oss --disable-indev=jack --disable-outdev=oss
--disable-altivec --disable-avx --cpu=atom --enable-hardcoded-tables
libavutil 50. 43. 0 / 50. 43. 0
libavcodec 52.123. 0 / 52.123. 0
libavformat 52.111. 0 / 52.111. 0
libavdevice 52. 5. 0 / 52. 5. 0
libavfilter 1. 80. 0 / 1. 80. 0
libswscale 0. 14. 1 / 0. 14. 1
libpostproc 51. 2. 0 / 51. 2. 0
Your ffmpeg build is compiled with --disable-network which I assume is the problem.

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