modusa 0.4.21__tar.gz → 0.4.24__tar.gz

{modusa-0.4.21 → modusa-0.4.24}/PKG-INFO

@@ -1,6 +1,6 @@
 Metadata-Version: 2.1
 Name: modusa
-Version: 0.4.21
+Version: 0.4.24
 Summary: A modular signal analysis python library.
 Author-Email: Ankit Anand <ankit0.anand0@gmail.com>
 License: MIT
@@ -11,8 +11,7 @@ Requires-Dist: yt-dlp==2025.9.23
 Requires-Dist: IPython>=9.5.0
 Requires-Dist: sounddevice>=0.5.2
 Requires-Dist: ipywidgets>=8.1.7
-Requires-Dist: soundfile>=0.13.1
-Requires-Dist: scipy>=1.16.2
+Requires-Dist: imageio-ffmpeg>=0.6.0
 Description-Content-Type: text/markdown
 
 # modusa

{modusa-0.4.21 → modusa-0.4.24}/pyproject.toml

@@ -12,12 +12,11 @@ dependencies = [
     "IPython>=9.5.0",
     "sounddevice>=0.5.2",
     "ipywidgets>=8.1.7",
-    "soundfile>=0.13.1",
-    "scipy>=1.16.2",
+    "imageio-ffmpeg>=0.6.0",
 ]
 requires-python = ">=3.11"
 readme = "README.md"
-version = "0.4.21"
+version = "0.4.24"
 
 [project.license]
 text = "MIT"

{modusa-0.4.21 → modusa-0.4.24}/src/modusa/__init__.py

@@ -8,4 +8,4 @@ from modusa.tools import play, convert, record, save
 from modusa.tools import download
 from modusa.tools import load, load_ann
 
-__version__ = "0.4.21"
+__version__ = "0.4.24"

modusa-0.4.24/src/modusa/tools/audio_loader.py

@@ -0,0 +1,155 @@
+#!/usr/bin/env python3
+
+
+import subprocess
+import numpy as np
+import imageio_ffmpeg as ffmpeg
+from pathlib import Path
+import re
+
+def _get_audio_info_ffmpeg(path: Path):
+	"""
+	To get the original sampling rate and number of
+	channels of a given audio file by parsing the
+	metadata. (No extra tool required).
+
+	Parameters
+	----------
+	audiofp: PathLike
+		- Audio filepath
+	
+	Returns
+	-------
+	int
+		- Original sampling rate (hz)
+	int
+		- Number of channels
+	"""
+	ffmpeg_exe = ffmpeg.get_ffmpeg_exe()
+	cmd = [ffmpeg_exe, "-i", str(path)]
+	proc = subprocess.run(cmd, stderr=subprocess.PIPE, text=True)
+	text = proc.stderr
+	
+	# Example parse: "Stream #0:0: Audio: mp3, 44100 Hz, stereo, ..."
+	m = re.search(r'Audio:.*?(\d+)\s*Hz.*?(mono|stereo)', text)
+	if not m:
+		raise RuntimeError("Could not parse audio info")
+	sr = int(m.group(1))
+	channels = 1 if m.group(2) == "mono" else 2
+	return sr, channels
+
+def _load_audio_from_youtube(url: str):
+	"""
+	Download audio from a YouTube URL, convert it to WAV, and return the path.
+
+	Parameters
+	----------
+	url : str
+		YouTube video URL.
+
+	Returns
+	-------
+	Path
+		Path to the converted WAV file (you can delete it later).
+	"""
+	from modusa.tools.youtube_downloader import download
+	from modusa.tools.audio_converter import convert
+	import tempfile
+	
+	# Temporary directory to hold files (auto-created, not auto-deleted)
+	tmpdir = Path(tempfile.mkdtemp())
+	
+	# Download YouTube audio (e.g. .m4a or .webm)
+	audio_fp: Path = download(url=url, content_type="audio", output_dir=tmpdir)
+	
+	# Convert downloaded file to .wav
+	wav_audio_fp: Path = convert(inp_audio_fp=audio_fp, output_audio_fp=audio_fp.with_suffix(".wav"))
+	
+	# Return path to the WAV file
+	return wav_audio_fp
+
+#---------------------
+# Main Function
+#---------------------
+def load(path, sr=None, trim=None, ch=None):
+	"""
+	Lightweight audio loader using imageio-ffmpeg.
+
+	Parameters
+	----------
+	path: PathLike/str/URL
+		- Path to the audio file / YouTube video
+	sr: int
+		- Sampling rate to load the audio in.
+		- Default: None => Use the original sampling rate
+	trim: tuple[number, number]
+		- (start, end) in seconds to trim the audio clip.
+		- Default: None => No trimming
+	ch: int
+		- 1 for mono and 2 for stereo
+		- Default: None => Use the original number of channels.
+
+	Returns
+	-------
+	np.ndarray
+		- Audio signal Float32 waveform in [-1, 1].
+	int:
+		Sampling rate.
+	str:
+		File name stem.
+	"""
+	path = Path(path)
+	ffmpeg_exe = ffmpeg.get_ffmpeg_exe()
+	
+	yt = False # Is the path a youtube URL
+	
+	if ".youtube" in str(path):
+		yt = True
+		try:
+			path: Path = _load_audio_from_youtube(url=str(path))
+		except Exception as e:
+			raise ConnectionRefusedError("unable to download from YouTube")
+	
+	# Find the real sample rate from the file
+	if sr is None:
+		sr, _ = _get_audio_info_ffmpeg(path)
+		if not (sr > 100 and sr < 80000):
+			raise Exception(f"Error reading the metadata for original sampling rate {sr}, please set `sr` explicitly")
+	
+	# Find the real number of channels from the file
+	if ch is None:
+		_, ch = _get_audio_info_ffmpeg(path)
+		
+		if ch not in [1, 2]:
+			raise Exception(f"Error reading the metadata for number of channels {ch}, please set `ch` explicitly")
+		
+	cmd = [ffmpeg_exe]
+	
+	# Optional trimming
+	if trim is not None:
+		start, end = trim
+		duration = end - start
+		cmd += ["-ss", str(start), "-t", str(duration)]
+		
+	cmd += ["-i", str(path), "-f", "s16le", "-acodec", "pcm_s16le"]
+	cmd += ["-ar", str(sr)]
+	cmd += ["-ac", str(ch)]
+		
+	cmd += ["-"]
+	
+	proc = subprocess.Popen(cmd, stdout=subprocess.PIPE, stderr=subprocess.DEVNULL)
+	raw = proc.stdout.read()
+	proc.wait()
+	
+	audio = np.frombuffer(raw, np.int16).astype(np.float32) / 32768.0
+	
+	# Stereo reshaping if forced
+	if ch == 2:
+		audio = audio.reshape(-1, 2).T
+		
+	# Delete the file if downloaded from youtube
+	if yt:
+		path.unlink(missing_ok=True)
+		path.parent.rmdir()
+		
+	return audio, sr, path.stem

modusa-0.4.24/tests/test_load.py

@@ -0,0 +1,158 @@
+#!/usr/bin/env python3
+
+#---------------------------------
+# Author: Ankit Anand
+# Date: 21/10/25
+# Email: ankit0.anand0@gmail.com
+#---------------------------------
+
+import pytest
+import modusa as ms
+from pathlib import Path
+
+
+#----------------------------------
+# Loading different audio formats
+#----------------------------------
+this_dir = Path(__file__).parents[0].resolve()
+def test_load_aac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aac")
+	assert title == "sample"
+
+def test_load_aiff():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aiff")
+	assert title == "sample"
+	
+def test_load_flac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.flac")
+	assert title == "sample"
+	
+def test_load_m4a():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.m4a")
+	assert title == "sample"
+	
+def test_load_mp3():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.mp3")
+	assert title == "sample"
+
+def test_load_opus():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.opus")
+	assert title == "sample"
+
+def test_load_wav():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.wav")
+	assert title == "sample"
+
+#----------------------------------
+# Resampe feature
+#----------------------------------
+SR = 16000 # Hz
+def test_load_with_resample_aac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aac", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+	
+def test_load_with_resample_aiff():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aiff", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+	
+def test_load_with_resample_flac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.flac", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+	
+def test_load_with_resample_m4a():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.m4a", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+	
+def test_load_with_resample_mp3():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.mp3", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+	
+def test_load_with_resample_opus():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.opus", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+	
+def test_load_with_resample_wav():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.wav", sr=SR)
+	assert sr == SR
+	assert title == "sample"
+
+#----------------------------------
+# Trim feature
+#----------------------------------
+def test_load_with_trim_aac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aac", trim=(0, 5.3))
+	assert title == "sample"
+	
+def test_load_with_trim_aiff():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aiff", trim=(0, 5.3))
+	assert title == "sample"
+	
+def test_load_with_trim_flac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.flac", trim=(0, 5.3))
+	assert title == "sample"
+	
+def test_load_with_trim_m4a():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.m4a", trim=(0, 5.3))
+	assert title == "sample"
+	
+def test_load_with_trim_mp3():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.mp3", trim=(0, 5.3))
+	assert title == "sample"
+	
+def test_load_with_trim_opus():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.opus", trim=(0, 5.3))
+	assert title == "sample"
+	
+def test_load_with_trim_wav():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.wav", trim=(0, 5.3))
+	assert title == "sample"
+
+#----------------------------------
+# Mono feature
+#----------------------------------
+def test_load_in_stereo_aac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aac", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+	
+def test_load_in_stereo_aiff():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.aiff", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+	
+def test_load_in_stereo_flac():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.flac", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+	
+def test_load_in_stereo_m4a():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.m4a", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+	
+def test_load_in_stereo_mp3():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.mp3", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+	
+def test_load_in_stereo_opus():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.opus", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+	
+def test_load_in_stereo_wav():
+	y, sr, title = ms.load(this_dir / "testdata/audio-formats/sample.wav", ch=2)
+	assert y.ndim == 2
+	assert title == "sample"
+
+#----------------------------------
+# Load from YouTube
+#----------------------------------
+def test_load_youtube_1():
+	y, sr, title = ms.load("https://www.youtube.com/watch?v=DIU_vmElPkU", ch=1)

modusa-0.4.21/src/modusa/tools/audio_loader.py

@@ -1,109 +0,0 @@
-#!/usr/bin/env python3
-
-
-import soundfile as sf
-from scipy.signal import resample
-import numpy as np
-from pathlib import Path
-import tempfile
-from scipy.signal import resample
-from .youtube_downloader import download
-from .audio_converter import convert
-
-def load(path, sr=None, trim=None, mono=True):
-	"""
-	Loads audio file from various sources.
-
-	.. code-block:: python
-		
-		import modusa as ms
-		audio_fp = ms.load(
-			"https://www.youtube.com/watch?v=lIpw9-Y_N0g",
-			sr = None, trim=(5, 10))
-
-	Parameters
-	----------
-	path: str
-		- Path to the audio file.
-		- YouTube URL.
-	sr: int | None
-		- Sampling rate to load the audio in.
-	trim: number | tuple[number, number] | None
-		- Segment of the audio to load.
-		- Example: 10 => First 10 seconds, (5, 10) => 5 to 10 seconds.
-		- Default: None => Entire audio.
-	mono: bool
-		- If True, loads the signal in mono.
-
-	Return
-	------
-	np.ndarray
-		- Audio signal.
-	int
-		- Sampling rate of the loaded audio signal.
-	title
-		- Title of the loaded audio.
-		- Filename without extension or YouTube title.
-	"""
-	# Check if the path is YouTube
-	if ".youtube." in str(path):
-		# Download the audio in temp directory using tempfile module
-		with tempfile.TemporaryDirectory() as tmpdir:
-			# Download
-			audio_fp: Path = download(url=path, content_type="audio", output_dir=Path(tmpdir))
-			
-			# Convert the audio to ".wav" form for loading
-			wav_audio_fp: Path = convert(inp_audio_fp=audio_fp, output_audio_fp=audio_fp.with_suffix(".wav"))
-			
-			# Load the audio in memory
-			audio_data, audio_sr = sf.read(wav_audio_fp)
-			title = audio_fp.stem
-	else:
-		# Check if the file exists
-		fp = Path(path)
-		
-		if not fp.exists():
-			raise FileNotFoundError(f"{path} does not exist.")
-			
-		# Load the audio in memory
-		audio_data, audio_sr = sf.read(fp)
-		title = fp.stem
-		
-	# Convert to mono if requested and it's multi-channel
-	if mono and audio_data.ndim > 1:
-		audio_data = audio_data.mean(axis=1)
-		
-	# Resample if needed
-	if sr is not None and audio_sr != sr:
-		n_samples = int(len(audio_data) * sr / audio_sr)
-		
-		if audio_data.ndim == 1:
-			# Mono
-			audio_data = resample(audio_data, n_samples)
-		else:
-			# Stereo or multi-channel: resample each channel independently
-			audio_data = np.stack([
-				resample(audio_data[:, ch], n_samples)
-				for ch in range(audio_data.shape[1])
-			], axis=1)
-			
-		audio_sr = sr
-		
-	# Trim if requested
-	if trim is not None:
-		if isinstance(trim, (int, float)):
-			trim = (0, trim)
-		elif isinstance(trim, tuple) and len(trim) > 1:
-			trim = (trim[0], trim[1])
-		else:
-			raise ValueError(f"Invalid trim type or length: {type(trim)}, len={len(trim)}")
-			
-		start = int(trim[0] * audio_sr)
-		end = int(trim[1] * audio_sr)
-		audio_data = audio_data[start:end]
-		
-	# Clip to avoid out-of-range playback issues
-	if np.issubdtype(audio_data.dtype, np.floating):
-		audio_data = np.clip(audio_data, -1.0, 1.0)
-		
-	return audio_data.T, audio_sr, title