L'octidi 28 messidor, an CCXXV, Hans Carlson a écrit : > "... not implemented yet"? Does that mean it's currently in the works? Or > just something that has been considered, but no work has been done?
It was considered, but as far as I know nobody is actively working on it. > $ ffmpeg -f s16le -ac 2 -i > 'concat:subfile,,start,5292000,end,47628000,,:file1.raw|subfile,,start,1764000,end,28224000,,:file2.raw' > -codec mp3 test.mp3 > > To calculate the subfile offset sizes, I multiplied the start/end times by > 176400 (which is the filesize=[52920320] / duration=[300.001814]). So, for > file1.raw, inpoint=30 = (176400 * 30 = 5292000) and outpoint=270 = (176400 * > 270 = 47628000). Your ffmpeg command-line has a problem: it does not specify the sample rate, and thus falls back to whatever default libavformat is using. I strongly advise against relying on default values in this kind of cases, always specify the relevant parameters. Only you can know the correct sample rate. Once you know it, the octets sizes are easy: duration times number of samples per second (= sample rate) times number of channels times number of octets per channel value. The value you used, 176400, is consistent to libaformat's current default sample rate for PCM files. > I previously looked into the concat FILTER, but wasn't sure how to extract a > section from each file. I assume you mean "atrim" in my case and not > "trim". Indeed. It is one of these filters that exist in both video and audio variant, with the 'a' prefix for the audio. > Thanks for the various options. No problem, glad it solved the issue. Regards, -- Nicolas George
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