feat(audio): use PyAV instead of ffmpeg (#31)

* 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

---------

Co-authored-by: github-actions[bot] <github-actions[bot]@users.noreply.github.com>

infer/modules/uvr5/modules.py

@@ -4,7 +4,7 @@ import logging
 
 logger = logging.getLogger(__name__)
 
-import ffmpeg
+from infer.lib.audio import resample_audio, get_audio_properties
 import torch
 
 from configs import Config
@@ -46,27 +46,24 @@ def uvr(model_name, inp_root, save_root_vocal, paths, save_root_ins, agg, format
             need_reformat = 1
             done = 0
             try:
-                info = ffmpeg.probe(inp_path, cmd="ffprobe")
-                if (
-                    info["streams"][0]["channels"] == 2
-                    and info["streams"][0]["sample_rate"] == "44100"
-                ):
-                    need_reformat = 0
+                channels, rate = get_audio_properties(inp_path)
+
+                # Check the audio stream's properties
+                if channels == 2 and rate == 44100:
                     pre_fun._path_audio_(
                         inp_path, save_root_ins, save_root_vocal, format0, is_hp3=is_hp3
                     )
+                    need_reformat = 0
                     done = 1
-            except:
+            except Exception as e:
                 need_reformat = 1
-                traceback.print_exc()
+                print(f"Exception {e} occured. Will reformat")
             if need_reformat == 1:
                 tmp_path = "%s/%s.reformatted.wav" % (
                     os.path.join(os.environ["TEMP"]),
                     os.path.basename(inp_path),
                 )
-                os.system(
-                    f'ffmpeg -i "{inp_path}" -vn -acodec pcm_s16le -ac 2 -ar 44100 "{tmp_path}" -y'
-                )
+                resample_audio(inp_path, tmp_path, "pcm_s16le", "s16", 44100, "stereo")
                 inp_path = tmp_path
             try:
                 if done == 0: