Reviewer's Guide by Sourcery

This PR fixes sampling rate handling for non-SND files by introducing sampling rate parameters to audio conversion functions. The implementation extracts the sampling rate using mediainfo before conversion and passes it to both sox and ffmpeg converters. The changes ensure correct sampling rate preservation during audio file conversion.

Sequence diagram for audio file conversion with sampling rate

sequenceDiagram
    participant User
    participant AudioFile
    participant MediaInfo
    participant Sox
    participant FFmpeg
    participant SoundFile

    User->>AudioFile: Call read(file)
    AudioFile->>MediaInfo: Get sampling rate
    MediaInfo-->>AudioFile: Return sampling rate
    AudioFile->>Sox: Try convert(file, tmpfile, offset, duration, sampling_rate)
    alt Sox conversion fails
        AudioFile->>FFmpeg: Try convert(file, tmpfile, offset, duration, sampling_rate)
        alt FFmpeg conversion fails
            AudioFile->>User: Raise binary_missing_error("ffmpeg")
        else FFmpeg conversion succeeds
            FFmpeg-->>AudioFile: Conversion complete
        end
    else Sox conversion succeeds
        Sox-->>AudioFile: Conversion complete
    end
    AudioFile->>SoundFile: Read tmpfile
    SoundFile-->>AudioFile: Return signal, sampling_rate
    AudioFile-->>User: Return signal, sampling_rate

Updated class diagram for audio conversion functions

classDiagram
    class AudioFile {
        +read(file)
    }
    class MediaInfo {
        +get_sampling_rate(file)
    }
    class Sox {
        +run_sox(infile, outfile, offset, duration, sampling_rate)
    }
    class FFmpeg {
        +run_ffmpeg(infile, outfile, offset, duration, sampling_rate)
    }
    class Convert {
        +convert(infile, outfile, offset, duration, sampling_rate)
    }
    AudioFile --> MediaInfo : uses
    AudioFile --> Sox : uses
    AudioFile --> FFmpeg : uses
    AudioFile --> Convert : uses
    Convert --> Sox : calls
    Convert --> FFmpeg : calls

File-Level Changes

Assessment against linked issues

Possibly linked issues

• OPUS files and ground truth for sampling rate #157: PR fixes the sampling rate issue for OPUS files as described in the issue.

Codecov Report

All modified and coverable lines are covered by tests ✅

Project coverage is 100.0%. Comparing base (69b0c97) to head (ffe0389).
Report is 1 commits behind head on main.

I would be curious to understand what I could do as a review.

Code-wise there is not much to say, one could possibly describe the unusual behavior of opus in one sentence somewhere (don't even know where)

The unittests are all passing. My question whether it should entail testing against further external audio data. If not I would continue with approval.

I think you addressed the important points in your review, and I tried to improve on them.

There is not really a change how we test the opus file, as the test file was broken before, and the test was hence passing for the main branch before. At the moment we are testing only for a few non standard audio formats:

• mp4 (which is also not really a format, but a container)
• opus
• aac
• m4a
• amr

If you think we should expand on this, please open an issue.