[Solved] JWPlayer - Video not found or access denied

[Gnewt]

I recently set up Subsonic to recognize and transcode video as well. I tweaked everything nicely, have my transcoder settings working etc, but I have an issue: when I try to play a video, nothing happens. If I go fullscreen and try, it shows: "Video not found or access denied: /{urlhere}"

If I try accessing the URL manually, a file does in fact start to download, so I know my permissions and transcoding settings are correct. I recently upgraded to JWPlayer 5.4 but to no avail. Help?

[Exrace]

Have you tried an external player?

[Gnewt]

Never mind, all is fixed. I stupidly forgot to symlink ffmpeg into the correct directory, so it wasn't in Subsonic's PATH.

[Exrace]

Thanks for reporting back.

[DViz]

Never mind, all is fixed. I stupidly forgot to symlink ffmpeg into the correct directory, so it wasn't in Subsonic's PATH.

I know I'm probably just being stupid, but can you explain to me exactly what you did here? I can see that ffmpeg is in the transcoding folder (I can listen to music files fine) but for videos, I'm just getting the same error you mentioned.

Oddly enough, SOME video files in my library work, just not all of them.

[kapz]

I'm getting the same error on SOME of my movies while other play. I compiled my own ffmpeg v0.6.1 on my slackware machine. I did create symlinks. They are all x264 mp4 files. Any clues???

My log file shows an error about resampling 6-channel audio.

Code: Select all

[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0(und): Video: flv, yuv420p, 653x272 [PAR 1:1 DAR 653:272], q=2-31, 1000 kb/s, 1k tbn, 23.98 tbc
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1(und): Audio: adpcm_swf, 44100 Hz, 2 channels, s16, 64 kb/s
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream mapping:
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0 -> #0.0
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1 -> #0.1
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Press [q] to stop encoding
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Resampling with input channels greater than 2 unsupported.
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Can not resample 6 channels @ 48000 Hz to 2 channels @ 44100 Hz

[jbaruch76]

I'm getting the same error on SOME of my movies while other play. I compiled my own ffmpeg v0.6.1 on my slackware machine. I did create symlinks. They are all x264 mp4 files. Any clues???

My log file shows an error about resampling 6-channel audio.

Code: Select all

[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0(und): Video: flv, yuv420p, 653x272 [PAR 1:1 DAR 653:272], q=2-31, 1000 kb/s, 1k tbn, 23.98 tbc
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1(und): Audio: adpcm_swf, 44100 Hz, 2 channels, s16, 64 kb/s
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream mapping:
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0 -> #0.0
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1 -> #0.1
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Press [q] to stop encoding
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Resampling with input channels greater than 2 unsupported.
[1/6/11 9:08:06 PM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Can not resample 6 channels @ 48000 Hz to 2 channels @ 44100 Hz

i get the same error, it seems to just be on my mkvs, but haven't thoroughly tested yet.

[stozher]

Can not resample 6 channels @ 48000 Hz to 2 channels @ 44100 Hz

Remove "-ar 44100" from transcoding row...

[kapz]

stozher,

I removed the "-ar 44100" from the following mp4 > flv and now get a new error in the subsonic log (in fact removing the -ar switch broke all the movies from playing)...

Code: Select all

ffmpeg -ss %o -i %s -async 1 -b %bk -s %wx%h -ac 2 -v 0 -f flv -

Code: Select all

[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0(und): Video: h264, yuv420p, 720x352 [PAR 6041:5000 DAR 54369:22000], 1698 kb/s, PAR 869:720 DAR 79:32, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1(und): Audio: aac, 48000 Hz, 5.1, s16, 129 kb/s
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) [adpcm_swf @ 0x8ab0c60]Sample rate must be 11025, 22050 or 44100
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Output #0, flv, to 'pipe:':
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0(und): Video: flv, yuv420p, 870x352 [PAR 1:1 DAR 435:176], q=2-31, 1000 kb/s, 90k tbn, 23.98 tbc
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1(und): Audio: adpcm_swf, 48000 Hz, 2 channels, s16, 64 kb/s
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream mapping:
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.0 -> #0.0
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Stream #0.1 -> #0.1
[1/7/11 9:44:38 AM PST]    DEBUG   InputStreamReaderThread   (/var/subsonic/data/transcode/ffmpeg) Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height

After searching around I was led down a path to compile ffmpeg using some optional ./configure parameters using this page as a guide.

I had to re-compile my lame, faad, faac, libs so "make install" would put everything in the default locations (instead of using the --extra-cflags and --extra-libs switches). I was able to compile ffmpeg with the following ./configure

Code: Select all

./configure --enable-gpl --enable-nonfree --enable-libfaad --enable-libfaac --enable-libmp3lame

After replacing this new ffmpeg in the subsonic/transcode folder, I tried a movie that was giving me the down-sample error before. It still did not play. In fact none of my movies (mp4 - x264) or shows (avi divx/xvid) would play now

I was only getting this one line in the subsonic log...

Code: Select all

[1/7/11 12:06:49 PM PST]    DEBUG   TranscodeInputStream   Starting transcoder: [/var/subsonic/data/transcode/ffmpeg] [-ss] [0] [-i] [/mnt/user/VIDEO/_DVD/DVD_Movies/A Bug's Life (1998)/A Bug's Life (1998).mp4] [-async] [1] [-b] [1000k] [-s] [653x272] [-ar] [44100] [-ac] [2] [-v] [0] [-f] [flv] [-]

I know all this doesn't fix the issue, but I hope this information helps someone continue the search. I guess I could re-encode that movie forcing 2-channel audio. It'll only takes an hour.

UPDATE: seems its been known for a long time by the ffmpeg team https://roundup.ffmpeg.org/issue185

[stozher]

ffmpeg -v 0 -ss %o <"-async ..."> -i %s -f flv -b %bk <"-r ..." or noting for default> -s %wx%h -acodec libfaac <"-ab ..." or "-aq ..." or noting for default 64k> -ac 2 -

or two step encoding

ffmpeg -v 0 -ss %o <"-async ..."> -i %s -f flv -b %bk <"-r ..." or noting for default> -s %wx%h -acodec libfaac <"-ab ..." or "-aq ..." or noting for default 64k> -ac 2 - | ffmpeg -v 0 -f flv -i - -f flv -vcodec copy -acodec libfaac -ar 44100 <"-ab ..." or "-aq ..." or noting for default 64k> -

"-async 1" is a special case where only the start of the audio stream is corrected without any later correction.

For "-async" read http://ffmpeg-users.933282.n4.nabble.com/async-td935787.html

See also http://forum.subsonic.org/forum/viewtopic.php?p=19422#19422
http://forum.subsonic.org/forum/viewtopic.php?p=18197#18197
http://forum.subsonic.org/forum/viewtopic.php?t=5121

[stozher]

FFmpeg version SVN-r25838, Copyright (c) 2000-2010 the FFmpeg developers
built on Nov 29 2010 15:39:30 with gcc 4.4.5
configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --extra-libs=-lgcrypt --disable-altivec --disable-armv5te --disable-armv6 --disable-vis
libavutil 50.33. 0 / 50.33. 0
libavcore 0.14. 0 / 0.14. 0
libavcodec 52.97. 2 / 52.97. 2
libavformat 52.87. 1 / 52.87. 1
libavdevice 52. 2. 2 / 52. 2. 2
libavfilter 1.65. 0 / 1.65. 0
libswscale 0.12. 0 / 0.12. 0
libpostproc 51. 2. 0 / 51. 2. 0
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...

[kapz]

Stozher,

Using your first method works now, but I had to remove the "-ac 2" switch.

Code: Select all

ffmpeg -ss %o -i %s -async 1 -b %bk -s %wx%h -acodec libfaac -v 0 -f flv -

Now all my movies work!

[stozher]

Move this "-async 1" before "-i"... read link above for "-async"...

Synchronization from input file not from output... Output is a blank at begin of synchronization... and after this FFmpeg go to encoding output file.

ffmpeg -v 0 -ss %o -async 1 -i %s -f flv -b %bk -s %wx%h -acodec libfaac -

[stozher]

In your new situation just use "-acodec copy" and don't transcode audio:

ffmpeg -v 0 -ss %o -async 1 -i %s -f flv -b %bk -s %wx%h -acodec copy -

Flash with 5.1 sound...

[kapz]

Thanks, I used the transcode string you suggest.

As for "-acodec copy", bandwidth is the name of the game and I rather use it for video quality, but its good to know.