There is a bug in librosa 0.9.1.
https://github.com/librosa/librosa/pull/1594
As a result, an error occurs when executing the "Vocals/Accompaniment Separation & Reverberation Removal" function.
To address this issue, librosa has been upgraded to version 0.10.2.
Additionally, torchcrepe has been upgraded due to its dependency on librosa.
* feat(audio): use PyAV instead of ffmpeg
replaced usage of ffmpeg in favor of PyAV (`av`)
* refactor(audio): store all of the audio related functions in the `infer.lib.audio`
refactors previous commit to have singular functions for each task, all located in `infer.lib.audio`
* fix(audio): remove downsample_audio from mdxnet.py
it is no longer needed, since it's imported from infer.lib.audio
* docs: remove every ffmpeg mention in the documentation to avoid confusion
* chore(requirements): remove ffmpeg-python and ffmpy from all requirements
* fix(audio): fix loading for UVR
wrapped gathering of META info from the stream into a function
fixes loading for UVR
* fix(audio): use np.frombuffer() instead of direct conversion of the resampled frames
this fixes traceback on preprocessing
* feat(audio): pre-allocate decoded_audio array in the load_audio function
this should improve performance, even if just a little
* Revert "docs: remove every ffmpeg mention in the documentation to avoid confusion"
This reverts commit 1e05bbce03.
* chore(format): run black on dev
* fix(requirements): revert removal of ffmpeg in unitest.yml and Dockerfile
* Revert "fix(requirements): revert removal of ffmpeg in unitest.yml and Dockerfile"
This reverts commit e28a0eebb2.
* feat(audio): pre-allocate numpy array to store the AudioFrame data in ndarray of dtype float32
* chore(format): run black on dev
* fix(audio): fix the decoded_audio size estimation
in estimated_total_samples we multiply by `sr` instead of `container.streams.audio[0].rate` since we want to estimate size of the OUTPUT file, not the input one. - Added dynamic resizing, in case something goes wrong and the size of decoded_audio is estimated incorrectly
Fixed function `load_audio` when the input audio's samplerate does not match the desired samplerate (`sr`)
* chore(format): run black on dev
* refactor(audio): remove `clean_path()` function as it serves no purpose anymore
* docs: remove everything related to ffmpeg
this includes everything except for formats support specification in the training_tips docs, since it has nothing to do with what ffmpeg does/did but rather what audio formats are supported (all the ones that ffmpeg supports!)
* docs: fix order of the steps in preparation in the READMEs
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