Registered Member
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Hello,
Maybe it's a very stupid question. Before I edit my AVCHD video, I convert it using DNxHD. That works very well. But I ran always in trouble with rendering. 2pass x264 always fail. So, how can I render the video with less quality loss so I can convert it later to x264? Thanks. |
Registered Member
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I didn't test it with HDV, but I do the following.
I convert the original with transcode to FFv1 (lossless codec). Then I render again to FFv1, but not as predefinded with flac, but with pcm. Flac produces a lot of errors, when creating the mp3-file, pcm does not. Also I don't hear problems, when I listen to the mp3-file built from the flac file. The FFv1-file is converted then with mencoder to x264. If you need more details, let me know. |
Registered Member
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As a workaround you should be able to render to your project with a lossless codec (select Lossless in the first drop-down list of the render window), such as Huffyuv. The file will be very large, but the compression is lossless, so the quality will be the best possible.
You can then convert your lossless video to h.264 using a piece of software like Avidemux (works quite well, very simple to use). |
Registered Member
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Thank you both, i've got a very very big file but good quality. Now it's time to do a 2pass x264 rendering. Someone have good settings for ffmpeg or mencoder?
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Registered Member
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Well it depends very much on what you want for yout final encoded file. I suppose the most important would be a maximum size for the file (which could basically be the bitrate multiplied by the duration in seconds).
You can simply use a Kdenlive render profile. You should actually be able to convert your large file file in Kdenlive directly by adding the file to a new project and rendering using the 2 pass h.264 profile you prefer. But you can also do it from the command line with ffmpeg if you wish. The command line parameters are given if you select a profile in the render window and click the "new profile" button. For instance h.264 10000k 2pass (with AAC audio) is defined like this : f=mp4 hq=1 acodec=libfaac ab=384k ar=48000 pix_fmt=yuv420p vcodec=libx264 minrate=0 b=10000k b_strategy=1 subcmp=2 cmp=2 coder=1 flags=+loop flags2=dct8x8 qmax=51 subq=7 qmin=10 qcomp=0.6 qdiff=4 trellis=1 aspect=%dar pass=2 I suppose these are reasonable default parameters, and you can simply adjust the ab= and b= values (audio and video bitrates). |
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