Whith this command you can extract the track you want: For track01. ffmpeg -f libcdio -ss 0 -t 494.773333 -i /dev/sr0 track01.wav. For track02. ffmpeg -f libcdio -ss 494.773333 -t 263.466667 -i /dev/sr0 track02.wav. Where 263.466667 is the difference between end time and start time of the track 02.
I want to use FFmpeg to convert FLAC to MP3; meanwhile, change the channel from stereo to mono and set the bitrate to 320kb/s. It is easy to do with the following command: ffmpeg -i test_audio.flac -ac 1 -ab 320k output.mp3. For sure, FFmpeg is a great tool for processing multimedia files with command lines.
Option placement matters with ffmpeg. You're attempting to apply an output option to the input. ffmpeg [input options] input [output options] output Corrected command: fluidsynth -T raw -F - sound_font_file.sf3 input.mid | ffmpeg -y -f s32le -i - -b:a 192k output.mp3 Fore more info about MP3 encoding with ffmpeg see FFmpeg Wiki: MP3. For example - to convert the contents of the directory input/, containing files of type .m4a and .flac, outputting to directory output/, converting to type .mp3 run: audioconvert convert input/ output/--output-format.mp3 Experimental. Audio can be passed to be converted to specific codecs. 1. I am using following command to create a hls aac audio file for web streaming. ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8. This command works only with some audio files. With many mp3 files I receive following output: ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3. Merely preserving an attached picture should be a simple matter of copying the picture stream to the mp3, though you don't mention what format you're converting from and some might store artwork differently. Share. The solution written there is to used these arguments when using ffmpeg. > I must use ffmpeg -i -f s16le tmp.dat and then mplayer -demuxer rawaudio -rawaudio rate=44100:channels=2:samplesize=2 -ao pcm tmp.dat to get "normal" wav file, many program can't read ffmpeg's wav file, they read 44 bytes of header, and others data use as sound spqaepZ.