- LosslessConversionQuality model with bit depth/sample rate caps,
applied only when they reduce source quality
- FFmpegService probes sample rate and appends codec-specific args
(-ar, -sample_fmt, -bits_per_raw_sample) for FLAC/ALAC/WAV/AIFF
- Batch + single-track convert sheets expose quality cap options
- Persist real converted bit depth/sample rate to history/library DB
- track_metadata: recognize and convert to WAV/AIFF targets
- convertedAudioQualityLabel reflects actual output quality
WAV/AIFF: library scan, quality probe, native tag read/write via embedded ID3 chunk (RIFF id3 / AIFF ID3), cover art, ReadFileMetadata, ExtractLyrics, and FLAC<->WAV/AIFF conversion (PCM, bit-depth preserved via ffprobe). Treat WAV/AIFF as lossless across all convert sheets (no bitrate picker, Lossless labels) via isLosslessConversionTarget. Native MIME maps for SAF. Redesign the track metadata three-dot menu to a settings-style grouped card with a single divider above Share.
Audio Analysis:
- Add rescan capability by bumping cache version
- Display channel layout (stereo, 5.1, etc.) and bitrate
- Use astats filter for more accurate peak/RMS measurements
- Support more formats: mp4, ac3, eac3, mka, wv, ape, tta, aif
- Only report bit depth for codecs that store it (FLAC, ALAC, WAV)
- Validate cache for SAF content:// URIs
Conversion:
- Add AAC as conversion target format
- Recognize ALAC as lossless source
- Prevent accidental deletion when source and target URI match
- Store format and bitrate in database after conversion
Utilities:
- Add audio_conversion_utils.dart for centralized conversion logic
- Add isSameContentUri() helper for safe URI comparison
zarzet