convert .ogg to .flv in 8.04

[stoppage]

When I try to convert .ogg (from RMD) to .flv with..........

ffmpeg -i out2.ogg -b 128 -s 400x240 -pass 1 -passlogfile log-file output.flv

terminal output reads...

FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al.

configuration: --enable-gpl --enable-pp --enable-swscaler --enable-pthreads --enable-libvorbis --enable-libtheora --enable-libogg --enable-libgsm --enable-dc1394 --disable-debug --enable-libmp3lame --enable-libfaadbin --enable-libfaad --enable-libfaac --enable-xvid --enable-x264 --enable-liba52 --enable-amr_nb --enable-amr_wb --enable-shared --prefix=/usr

libavutil version: 1d.49.3.0

libavcodec version: 1d.51.38.0

libavformat version: 1d.51.10.0

built on Mar 17 2009 21:37:49, gcc: 4.2.4 (Ubuntu 4.2.4-1ubuntu3)

[theora @ 0xb7e1b9a8]Theora bitstream version 30201

[theora @ 0xb7e1b9a8]544 bits left in packet 81

[theora @ 0xb7e1b9a8]7 bits left in packet 82

Input #0, ogg, from 'out2.ogg':

Duration: 00:00:58.9, start: 0.066667, bitrate: 1051 kb/s

Stream #0.0: Video: theora, yuv420p, 1440x896, 15.00 fps(r)

Stream #0.1: Audio: vorbis, 48000 Hz, stereo, 499 kb/s

Output #0, flv, to 'output.flv':

Stream #0.0: Video: flv, yuv420p, 400x240, q=2-31, pass 1, 0 kb/s, 15.00 fps(c)

Stream #0.1: Audio: mp3, 48000 Hz, stereo, 64 kb/s

Stream mapping:

Stream #0.0 -> #0.0

Stream #0.1 -> #0.1

[flv @ 0xb7e1b9a8]removing common factors from framerate

[theora @ 0xb7e1b9a8]Theora bitstream version 30201

[theora @ 0xb7e1b9a8]544 bits left in packet 81

[theora @ 0xb7e1b9a8]7 bits left in packet 82

[mp3 @ 0xb7e1b9a8]flv doesnt support that sample rate, choose from (44100, 22050, 11025)

Could not write header for output file #0 (incorrect codec parameters ?)


Conversion to .avi is o.k.
Anybody any ideas please?

[Rubberman]

It didn't handle the audio data correctly. You need to specify the audio data rate as one of the arguments you didn't specify to be one of those supported. This is because the input file audio isn't compatible without resampling and you have 3 rates to choose from, depending upon how faithful you want your audio to sound. Use the -ar bitrate option where 'bitrate' would be 44100, 22050, or 11025. Try that and see if it finishes ok. FWIW, that seems to be the only real issue in the conversion.

[stoppage]

Tried all three....unfortunately quality of convert is disastrous

[Rubberman]

You might want to extract the audio and video streams, convert the audio to mp3 with as good fidelity as possible, then convert the video and merge the streams to the output .flv file. Part of your problem is likely the interlacing of the audio/video streams, which you can also (I think) adjust with ffmpeg. Anyway, when doing non-standard conversions like this, some experimentation is usually in order until you know what to do for best results in the future.