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Workflow Playbook
Diagnose codec, container, sample-rate, bit-depth, and channel mismatches, then choose a bounded conversion path for a target player or processing chain.
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The useful question is not whether an old file can be converted. It is which exact file the next player or processor will accept. Keep the source, collect the target's container, codec, sample-rate, bit-depth, channel, and profile requirements, and test with a representative clip before touching a production archive.
Start with audio-format-info. A WAV file normally carries PCM, but WAV is a container rather than a guarantee of one sample format. A file can therefore have the expected extension and still fail because its PCM packing, rate, bit depth, channel layout, or embedded stream is outside the receiver's support. Record the mismatch before converting.
Use AMR or GSM only when a telephone, voicemail, handset, or gateway contract asks for that codec. For general processing, MP3 to WAV provides an uncompressed PCM/WAV hand-off, while WAV to AAC, MP3, OGG, or Opus is appropriate only when the delivery environment names that codec. WAV to FLAC is the better branch when lossless compression is wanted.
For sample compatibility, change one parameter at a time: resample for the required rate, change sample format or reduce bit depth for the required PCM representation, dither when reducing depth is appropriate, and remap channels only for a known mono or stereo contract. If DSD or DTS is unsupported, use DSD to PCM or DTS to AC3 according to the destination rather than guessing.
Run audio-format-info on the output and compare every required field with the target contract. Then open the exact output in the real player or processing stage and check for decoder errors, silence, clipped duration, missing channels, and unexpected metadata changes. Link this record with audio-convert for broader format work, audio-analyze for evidence, audio-channel-routing for channel behavior, or audio-utility for adjacent handling.
Workflow playbook
Run audio-format-info on the original and record its container, codec, bitrate, sample rate, bit depth, channel count, layout, duration, and any profile information. Compare those facts with the target contract so a container problem is not confused with a codec or sample-format problem.
For a target that explicitly speaks AMR or GSM, select the direction that matches the hand-off: use AMR to MP3 or GSM to MP3 to make a copy easier to inspect, and MP3 to AMR or MP3 to GSM only when the receiving system requires that voice codec. Do not treat MP3 as a proof of quality recovery.
Use MP3 to WAV when the next processor needs uncompressed PCM in a WAV container. From WAV, choose AAC, MP3, OGG, or Opus for a specified compressed player path, or FLAC for lossless compression; retain the source because transcoding from a lossy file cannot restore discarded detail.
Choose the matching AMR or GSM direction only for a telephony, voicemail, handset, or legacy gateway contract. If the target accepts ordinary media formats, do not transcode into a narrowband voice codec merely because the source is old.
Use MP3 to WAV when a downstream tool needs WAV carrying PCM. Choose WAV to AAC, MP3, OGG, or Opus only when the target specifies a compressed delivery codec; choose WAV to FLAC when lossless compression is the useful trade-off.
Keep the original when the target supports the legacy stream. Otherwise use DSD to PCM for a PCM/WAV-compatible path or DTS to AC3 for a target that accepts Dolby Digital, then verify the resulting channels and playback behavior.
Apply audio-resample only for a stated sample-rate requirement. Use audio-change-sample-format or audio-bit-depth-reducer when the receiver needs a different PCM representation, add audio-dither when reducing bit depth where quantization noise matters, and use audio-channel-resample for an explicit mono or stereo requirement. Recheck that the change did not alter the intended channel mapping.
If the destination cannot decode DSD, use DSD to PCM to make a PCM/WAV-compatible output. If it cannot decode DTS but accepts AC3, use DTS to AC3 and confirm the requested channel layout and bitrate. Run audio-format-info on the output, then test the exact file in the target player or processing stage.