> On 7 Feb 2023, at 11:05, Alexander Bieliaev via ffmpeg-user > <ffmpeg-user@ffmpeg.org> wrote: > > I am processing audio chunks programmatically by using the ffmpeg library > for C#. First I divide the input audio of .wav format into chunks of 1 > minute each (I can't process the whole audio for specific reasons), then > prepend it's header to each chunk so it can be recognized and processed,
I don’t get it. If you speak C, and you have uncompressed Wave input, is as simple as opening the file, find the DATA riff and copy out X bytes (depending on the bit depth / sample rate) > then I get raw PCM of each chunk and replace some parts of PCM with sine > wave data (adding beeps) and then I transform that PCM chunks to mp3 chunks > and write those to stream. The final process of concating audio parts is *NOT > *performed by ffmpeg, I just write data chunks to the destination stream. I > am facing the problem that there are noticeable transitions between 1 > minute chunks in result audio(clicks/silence/change of volume/shifting). > How can I smooth out the start/end of each chunk so when I'm putting them > together there are no noticeable transitions? From experience I know that the size of -f segment for Wave does NOT result in a sample accurate amount of data, not sure if that is what you are facing. (Should not matter if there is a pure concat of the data.) It ’should’ be seamless. So, if I can do this kind of stuff in pure python, why find the cause and not do it yourself in C? Bouke _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
