Im trying to use ffmpeg to convert my flv files to mp4 to play them on iOS devices but the converted video has a much worse quality than the original one.
Here is the command i use:
ffmpeg -i input.flv -ar 22050 output.mp4
I would really appreciate if someone could provide me with the best settings for flv to mp4 conversion.
Depending on the codecs used in your FLV you may be able to get away with simply re-wrapping it in an mp4 container. You'll need H.264 or MPEG4 simple profile video and AAC audio. You can find out some info on your source file with ffmpeg -i input.flv
I'm not sure whether simply having H.264/MPEG4 Simple + AAC is good enough or if there are specific options to the codecs that are supported. It's easy enough to test:
Try using
ffmpeg -i input.flv -c copy -copyts output.mp4
-copyts is copy timestamps. It will help audio sync.
If that doesn't work, try forcing audio and video codecs. This will re-encode the file:
ffmpeg -i input.flv -c:v libx264 -crf 23 -c:a aac -b:a 160k output.mp4
If you want to change only the audio or only the video codec, you can use -c:a copy or -c:v copy to copy the one you want to keep.
To improve the video quality used when transcoding, you can use a lower CRF value, e.g. anything down to 18. To get a smaller file, use a higher CRF, but note that this will degrade quality.
To inprove the audio quality, choose a higher bitrate (160k in the example above).
With both the audio and video quality, your results will vary depending on the quality of the source. Be aware that there's only so much you can do with any original source - at best, you'll end up with the same quality as the source. It is definitely worthwhile to experiment with different quality levels and/or bitrates.
more info on FFMPEG aac encoding (I've been referring to the "native" encoder described on the ffmpeg site).
Regarding the ffmpeg command suggested in the question...
-ar refers to the audio sample rate. I recommend not messing with this parameter at all. If you want to play with audio encoding, adjust the bitrate (-b:a 160k above) and let the encoder choose what to do based on that.
For background, though, CD quality is 44100Hz sampling; typical video uses 48000Hz.
You may note that 22050 in the original question's example is 1/2 the cd quality sample rate. if you are downconverting CD material this would be a good choice. If you're starting with 48KHz source i'd use 24Khz instead. (but again, I'd recommend just picking a target bitrate and letting the encoder change the sample rate if needed.)
You should get info on the sample rates of the original material from the ffmpeg -i command I suggested at the top.
