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FFMPEG results in a silent video when trying to combine video and audio tracks
23 septembre 2017, par Stuart ClarkeI’m using the following command to combine a video and an audio track.
ffmpeg -y -i /var/www/temp/merged.mp4 -i /var/www/temp/combined.mp3 -strict -2 /var/www/temp/videoExtouiulbjryzxlehjj2.mp4
Edit :
Here is the output from the first command
ffmpeg version 2.5.10-0ubuntu0.15.04.1 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/var/www/temp/merged.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf56.15.102
Duration: 00:02:31.80, start: 0.000000, bitrate: 2384 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 2381 kb/s, 25.43 fps, 29.97 tbr, 11988 tbn, 59.94 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 0 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #1, mp3, from '/var/www/temp/audioTrack.mp3':
Metadata:
encoder : Lavf56.15.102
Duration: 00:02:08.94, start: 0.011995, bitrate: 128 kb/s
Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavf
[libx264 @ 0x1e903a0] using SAR=1/1
[libx264 @ 0x1e903a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX AVX2 FMA3 LZCNT BMI2
[libx264 @ 0x1e903a0] profile High, level 4.0
[libx264 @ 0x1e903a0] 264 - core 142 r2495 6a301b6 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to '/var/www/temp/videoExtvzxvphotsyfpcbkd.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf56.15.102
Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 29.97 fps, 30k tbn, 29.97 tbc (default)
Metadata:
handler_name : VideoHandler
encoder : Lavc56.13.100 libx264
Stream #0:1(und): Audio: mp3 (libmp3lame) (i[0][0][0] / 0x0069), 48000 Hz, stereo, fltp (default)
Metadata:
handler_name : SoundHandler
encoder : Lavc56.13.100 libmp3lame
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Stream #0:1 -> #0:1 (aac (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
frame= 31 fps=0.0 q=0.0 size= 0kB time=00:00:01.05 bitrate= 0.4kbits/s dup=1 drop=0frame= 46 fps= 41 q=0.0 size= 0kB time=00:00:01.56 bitrate= 0.2kbits/s dup=1 drop=0frame= 52 fps= 32 q=29.0 size= 149kB time=00:00:01.75 bitrate= 698.4kbits/s dup=1 drop=frame= 64 fps= 28 q=29.0 size= 258kB time=00:00:02.16 bitrate= 977.2kbits/s dup=1 drop=frame= 74 fps= 26 q=29.0 size= 389kB
video:39673kB audio:211kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.143487%
[libx264 @ 0x1e903a0] frame I:27 Avg QP:18.78 size: 58834
[libx264 @ 0x1e903a0] frame P:2202 Avg QP:23.67 size: 14187
[libx264 @ 0x1e903a0] frame B:2323 Avg QP:25.75 size: 3356
[libx264 @ 0x1e903a0] consecutive B-frames: 24.7% 18.7% 9.0% 47.5%
[libx264 @ 0x1e903a0] mb I I16..4: 18.8% 74.7% 6.5%
[libx264 @ 0x1e903a0] mb P I16..4: 2.1% 5.0% 0.3% P16..4: 22.2% 6.0% 1.9% 0.0% 0.0% skip:62.5%
[libx264 @ 0x1e903a0] mb B I16..4: 0.1% 0.3% 0.0% B16..8: 19.9% 1.0% 0.1% direct: 0.3% skip:78.3% L0:42.3% L1:55.3% BI: 2.4%
[libx264 @ 0x1e903a0] 8x8 transform intra:68.2% inter:86.5%
[libx264 @ 0x1e903a0] coded y,uvDC,uvAC intra: 35.8% 52.3% 4.5% inter: 5.2% 7.4% 0.0%
[libx264 @ 0x1e903a0] i16 v,h,dc,p: 28% 23% 8% 40%
[libx264 @ 0x1e903a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 20% 22% 3% 6% 6% 7% 5% 4%
[libx264 @ 0x1e903a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 21% 14% 4% 8% 7% 7% 4% 3%
[libx264 @ 0x1e903a0] i8c dc,h,v,p: 56% 18% 20% 6%
[libx264 @ 0x1e903a0] Weighted P-Frames: Y:0.9% UV:0.2%
[libx264 @ 0x1e903a0] ref P L0: 73.4% 13.0% 10.0% 3.6% 0.0%
[libx264 @ 0x1e903a0] ref B L0: 92.2% 6.7% 1.1%
[libx264 @ 0x1e903a0] ref B L1: 96.6% 3.4%
[libx264 @ 0x1e903a0] kb/s:2139.74End Edit
I know this should work since the exact line does work for other files. But the result with these is a silent video.
I checked the audio file being used and it seemed fine when played. but I did get a weird warning when creating it. Basically I have 2 audio tracks, one is overlay music and the other is talking. I create the combined audio with this command.
ffmpeg -y -i /var/www/temp/audioTrack.mp3 -i /var/www/temp/musicTrack.mp3 -filter_complex amerge -c:a libmp3lame -q:a 4 /var/www/temp/combined.mp3
Here is the output from the second command.
ffmpeg version 2.5.10-0ubuntu0.15.04.1 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
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
Input #0, mp3, from '/var/www/temp/audioTrack.mp3':
Metadata:
encoder : Lavf56.15.102
Duration: 00:02:08.89, start: 0.025057, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavc56.13
Input #1, mp3, from '/var/www/temp/musicTrack.mp3':
Metadata:
encoder : Lavf56.15.102
Duration: 00:02:08.89, start: 0.025057, bitrate: 128 kb/s
Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavc56.13
[Parsed_amerge_0 @ 0x15e3760] No channel layout for input 1
[Parsed_amerge_0 @ 0x15e3760] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
Output #0, mp3, to '/var/www/temp/combined.mp3':
Metadata:
TSSE : Lavf56.15.102
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p (default)
Metadata:
encoder : Lavc56.13.100 libmp3lame
Stream mapping:
Stream #0:0 (mp3) -> amerge:in0
Stream #1:0 (mp3) -> amerge:in1
amerge -> Stream #0:0 (libmp3lame)
Press [q] to stop, [?] for help
[libmp3lame @ 0x1601000] Trying to remove 1152 samples, but the queue is empty
size= 2001kB time=00:02:08.88 bitrate= 127.2kbits/s
video:0kB audio:2001kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.012350%Of note is these sections (I believe).
[Parsed_amerge_0 @ 0x15e3760] No channel layout for input 1
[Parsed_amerge_0 @ 0x15e3760] Input channel layouts overlap: output layout will be determined by the number of distinct input channelsand
[libmp3lame @ 0x1601000] Trying to remove 1152 samples, but the queue is empty
As I said the output audio file sounds correct.
The Music and Audio tracks are created with AudioSegment in python from the pydub library. I’ve used this before with no issues. The code for this is as follows.
sound = AudioSegment.from_mp3(audio)
introSilence = AudioSegment.silent(duration=introLength)
creditsSilence = AudioSegment.silent(duration=creditsLength)
increasedAudio = sound + 12
talking = introSilence + increasedAudio + creditsSilence
talking.export(audioTrack, format="mp3")
mus = AudioSegment.from_mp3(music)
introMusic = mus[ : introLength]
videoMusic = mus[introLength - crossFade : introLength + videoLength + crossFade]
creditsMusic = mus[totalLength - creditsLength : totalLength]
lowerMusic = videoMusic - 6
totalMusic = introMusic.append(lowerMusic, crossfade=crossFade).append(creditsMusic, crossfade=crossFade).fade_out(fadeOut)
totalMusic.export(musicTrack, format="mp3")Here is the output of
ffprobe
on the 2 audio files.for audioTrack
ffprobe version 2.5.10-0ubuntu0.15.04.1 Copyright (c) 2007-2016 the FFmpeg developers
built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
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
Input #0, mp3, from 'temp/audioTrack.mp3':
Metadata:
encoder : Lavf56.15.102
Duration: 00:02:08.89, start: 0.025057, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavc56.13and for musicTrack
ffprobe version 2.5.10-0ubuntu0.15.04.1 Copyright (c) 2007-2016 the FFmpeg developers
built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
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
Input #0, mp3, from 'temp/musicTrack.mp3':
Metadata:
encoder : Lavf56.15.102
Duration: 00:02:08.89, start: 0.025057, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavc56.13I can’t see any issues with these.
For completeness here is the ffprobe result for the combined audio track.
ffprobe version 2.5.10-0ubuntu0.15.04.1 Copyright (c) 2007-2016 the FFmpeg developers
built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
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
Input #0, mp3, from 'temp/combined.mp3':
Metadata:
encoder : Lavf56.15.102
Duration: 00:02:08.91, start: 0.025057, bitrate: 127 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 127 kb/s
Metadata:
encoder : Lavc56.13Any help would be much appreciated.
Thanks,
Stu.
-
FFMPEG conversion from MOV to MP4 Issue : Only 1st second is converted
24 avril 2012, par Jeffi'm trying to convert an MOV file to MP4.
I've tried so many options but the output file still only 1 second length.This is one of the FFMPEG comand I used :
ffmpeg -i NY_BTS1PastryHQnew.mov -f mp4 -vcodec copy -acodec copy
output.mp4The output.mp4 is only 1sec long.
The output of the FFMPEG is :ffmpeg version git-2012-02-22-534a82a Copyright (c) 2000-2012 the FFmpeg developers
built on Feb 22 2012 14:44:38 with gcc 4.4.5
configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab
libavutil 51. 40.100 / 51. 40.100
libavcodec 54. 4.100 / 54. 4.100
libavformat 54. 1.100 / 54. 1.100
libavdevice 53. 4.100 / 53. 4.100
libavfilter 2. 62.101 / 2. 62.101
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 7.100 / 0. 7.100
libpostproc 52. 0.100 / 52. 0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'NY_BTS1PastryHQnew.mov':
Metadata:
major_brand : qt
minor_version : 512
compatible_brands: qt
creation_time : 2011-06-17 01:38:11
encoder : Lavf54.1.100
Duration: 00:00:01.45, start: 0.000000, bitrate: 225 kb/s
Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720, 277 kb/s, 25 fps, 25 tbr, 2500 tbn, 5k tbc
Metadata:
creation_time : 2011-06-17 01:38:11
handler_name :
DataHandler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 122 kb/s
Metadata:
creation_time : 2011-06-17 01:38:11
handler_name :
DataHandler
File 'output.mp4' already exists. Overwrite ? [y/N] y
Output #0, mp4, to 'output.mp4':
Metadata:
major_brand : qt
minor_version : 512
compatible_brands: qt
creation_time : 2011-06-17 01:38:11
encoder : Lavf54.1.100
Stream #0:0(eng): Video: h264 (![0][0][0] / 0x0021), yuv420p, 1280x720, q=2-31, 277 kb/s, 25 fps, 2500 tbn, 2500 tbc
Metadata:
creation_time : 2011-06-17 01:38:11
handler_name :
DataHandler
Stream #0:1(eng): Audio: aac (@[0][0][0] / 0x0040), 48000 Hz, stereo, 122 kb/s
Metadata:
creation_time : 2011-06-17 01:38:11
handler_name :
DataHandler
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 12 fps= 0 q=-1.0 Lsize= 40kB time=00:00:00.40 bitrate= 814.6kbits/s
video:16kB audio:22kB global headers:0kB muxing overhead 4.766554%I can't see any error, do you have a clue ?
Thanks
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transcoding from amr to flac using JAVE(ffmpeg)
25 avril 2012, par user824440I want to transcode amr audio files to flac using JAVE (http://www.sauronsoftware.it/projects/jave/). it uses ffmpeg to do the job.
The transcoding procedure runs successfully on Windows but fails on Linux (centOS).
I checked the JAVE code and found it uses the following command :
ffmpeg -i 1.amr -vn -acodec flac -ac 1 -ar 8000 -f flac -y 1.flac
the output is :
FFmpeg version SVN-r11179, Copyright (c) 2000-2007 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-pp --enable-swscaler --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-memalign-hack --extra-ldflags=-static -lm -lz
libavutil version: 49.5.0
libavcodec version: 51.48.0
libavformat version: 52.1.0
built on Dec 7 2007 15:35:14, gcc: 4.1.2 20070626 (Red Hat 4.1.2-14)
Input #0, amr, from '1.amr':
Duration: N/A, bitrate: N/A
Stream #0.0: Audio: samr / 0x726D6173, 8000 Hz, mono
Input #1, amr, from '1.amr':
Duration: N/A, bitrate: N/A
Stream #1.0: Audio: samr / 0x726D6173, 8000 Hz, mono
Output #0, flac, to '1.flacffmpeg':
Stream #0.0: Audio: flac, 8000 Hz, mono, 64 kb/s
Output #1, flac, to '1.flac':
Stream #1.0: Audio: flac, 8000 Hz, mono, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Stream #1.0 -> #1.0
Unsupported codec (id=73728) for input stream #0.0So the problem might be unsuppported codec ? How to solve this ?
Thanks.