Trim an audio file in C# · oneaudio cookbook

Open the file, Seek to the start of the part you want, Read until its end and write what you read with a WavEncoder. Seeking is sample-exact, so the cut lands where you ask. The file here is a three-second synthetic test file from the oneaudio repository; the program keeps 0.5 s to 2.0 s.

Terminal

dotnet new console -o TrimAudio && cd TrimAudio
dotnet add package RecursiveAI.OneAudio --version 1.4.0
t=https://raw.githubusercontent.com/recursive-ai-oss/oneaudio-csharp/v1.4.0/testdata
curl -L -o song.flac $t/flac/stereo16_seektable.flac

Put this in Program.cs and run dotnet run:

using OneAudio;
 
TimeSpan from = TimeSpan.FromSeconds(0.5), to = TimeSpan.FromSeconds(2.0);
 
using IAudioDecoder decoder = Audio.Open("song.flac");
AudioFormat f = decoder.Format;
long start = (long)(from.TotalSeconds * f.SampleRate);
long end = Math.Min((long)(to.TotalSeconds * f.SampleRate), decoder.LengthFrames);
long fade = f.SampleRate / 100;                 // 10 ms at each end
 
decoder.Seek(start);
// A lossless source keeps its bit depth; a lossy one is written as 16-bit.
using (var wav = new WavEncoder(File.Create("cut.wav"), f, new WavEncoderSettings(),
                                leaveOpen: false))
{
    float[] block = new float[4096 * f.Channels];
    long position = start;
    while (position < end)
    {
        int want = (int)Math.Min(4096, end - position);
        int read = decoder.Read(block.AsSpan(0, want * f.Channels));
        if (read == 0)
            break;
        for (int i = 0; i < read; i++)          // fade in and out
        {
            long fromStart = position + i - start, fromEnd = end - (position + i) - 1;
            float gain = Math.Min(1f, Math.Min(fromStart, fromEnd) / (float)fade);
            for (int c = 0; c < f.Channels; c++)
                block[i * f.Channels + c] *= gain;
        }
        wav.Write(block.AsSpan(0, read * f.Channels));
        position += read;
    }
}
 
using IAudioDecoder cut = Audio.Open("cut.wav");
Console.WriteLine($"song.flac: {decoder.LengthFrames / (double)f.SampleRate:F3} s");
Console.WriteLine($"cut.wav: {cut.LengthFrames} frames, " +
                  $"{cut.LengthFrames / (double)f.SampleRate:F3} s, " +
                  $"{cut.Format.NativeType}");

Text

song.flac: 2.993 s
cut.wav: 33075 frames, 1.500 s, PcmSigned16

The 10 ms fades stop the cut from clicking where it lands mid-waveform. Leave the loop out (and use ReadInteger with WriteInteger) for a bit-exact cut of a lossless file.

Limits

Works the same for MP3, Ogg Vorbis, Opus, FLAC, AAC, AMR and WAV inputs: only the input name changes. MP3 and AAC seeks are exact because the decoder rebuilds its state from frames before the target.
The output is WAV here. To keep it compressed, write the WAV to FLAC or Opus with Audio.Transcode (see Convert audio files). A lossy file can’t be cut without decoding, so the result is re-encoded, never a byte-for-byte copy of the original frames.
A stream that can’t seek (a pipe or an HTTP body) throws NotSupportedException from Seek; read and discard frames up to the start instead.

To join clips end to end, see Join audio files.