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• Pydub FFMPEG issue [closed]

  14 janvier, par Nikolai van den Hoven

  I am attempting to use FFMPEG with Pydub to create a program that chops .mp3 files into different words, each contained in their own .mp3 file, but when I run the script I am getting the following error :

  


  PS C:\Users\nik> & C:/Users/nik/AppData/Local/Microsoft/WindowsApps/python3.12.exe "d:/Python/Word Splitter.py"
C:\Users\nik\AppData\Local\Packages\PythonSoftwareFoundation.Python.3.12_qbz5n2kfra8p0\LocalCache\local-packages\Python312\site-packages\pydub\utils.py:170: RuntimeWarning: Couldn't find ffmpeg or avconv - defaulting to ffmpeg, but may not work
  warn("Couldn't find ffmpeg or avconv - defaulting to ffmpeg, but may not work", RuntimeWarning)


  


  This is the code I am using.

  


  import os
from pydub import AudioSegment
from pydub.silence import split_on_silence
import speech_recognition as sr
AudioSegment.ffmpeg = r"D:\Python\ffmpeg\bin\ffmpeg.exe"
def mp3_to_words(mp3_file, output_folder):
    # Ensure output folder exists
    os.makedirs(output_folder, exist_ok=True)

    # Load MP3 file
    print("Loading audio file...")
    audio = AudioSegment.from_mp3(mp3_file)

    # Split audio into chunks using silence detection
    print("Splitting audio into chunks...")
    chunks = split_on_silence(
        audio,
        min_silence_len=200,  # Minimum silence duration in ms to consider as a split point
        silence_thresh=audio.dBFS - 14,  # Silence threshold relative to average loudness
        keep_silence=100  # Retain some silence in chunks
    )

    recognizer = sr.Recognizer()

    for i, chunk in enumerate(chunks):
        print(f"Processing chunk {i + 1}/{len(chunks)}...")

        # Save the chunk temporarily
        temp_file = os.path.join(output_folder, f"chunk_{i}.wav")
        chunk.export(temp_file, format="wav")

        # Recognize words in the chunk
        with sr.AudioFile(temp_file) as source:
            audio_data = recognizer.record(source)
            try:
                text = recognizer.recognize_google(audio_data)
                words = text.split()

                # Export each word as its own MP3
                word_start = 0
                for j, word in enumerate(words):
                    word_duration = len(chunk) // len(words)  # Approximate duration per word
                    word_audio = chunk[word_start:word_start + word_duration]
                    word_file = os.path.join(output_folder, f"word_{i}_{j}.mp3")
                    word_audio.export(word_file, format="mp3")
                    word_start += word_duration

            except sr.UnknownValueError:
                print(f"Could not understand chunk {i + 1}.")
            except sr.RequestError as e:
                print(f"Could not request results; {e}")

        # Clean up temporary file
        os.remove(temp_file)

    print(f"Processed {len(chunks)} chunks. Word MP3s saved in {output_folder}.")

if __name__ == "__main__":
    input_file = input("Enter the path to the MP3 file: ").strip()
    output_dir = input("Enter the output folder path: ").strip()

    mp3_to_words(input_file, output_dir)


  


  I have added the Base FFMPEG folder and the bin folder within it to Windows PATH
My PATH variable on Windows 11,
But it does not show up in the variable when I typed PATH into cmd

  


• avcodec/msmpeg4dec : Don't process unnecessarily many coefficients

  1er juillet 2024, par Andreas Rheinhardt

  avcodec/msmpeg4dec : Don't process unnecessarily many coefficients
  This code has been added in f5957f3fe2b8dfda13a9750acc3e3a03cd5d39c9
  
  when support for WMV1 has been added. It was needed for decoding
  
  inter macroblocks, as WMV1 uses a different scantable and
  
  the ScanTable and ScanTable.raster_end was not yet introduced
  
  (the H263 unquantize function instead used a hardcoded version
  
  of ff_zigzag_direct's raster_end). In other words,
  
  there is no reason to continue to do so for WMV1 inter blocks.
  
  (Apart from that : WMV1 inter blocks don't use
  
  dct_unquantize_h263_inter any more, as unquantizing happens
  
  when parsing the block via specialized RL VLC tables.)
  It is also not needed for WMV1/2 intra blocks (for non-ac_pred
  
  it uses the correct ScanTable used by the h263 unquantize
  
  intra function ; for ac_pred, the number is overridden)
  
  as well as for WMV2 inter-blocks (raster_end is not really
  
  used in this case ; the ordinary ScanTable is used in case
  
  abt_type is zero, otherwise there are two different ScanTables
  
  and block_last_index is overridden in wmv2dec.c).
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavcodec/msmpeg4dec.c

• How to split audio file into equal-length segments with ffmpeg ?

  11 mars, par GPWR

  I want to split an audio file into several equal-length segments using FFmpeg. I want to specify the general segment duration (no overlap), and I want FFmpeg to render as many segments as it takes to go over the whole audio file (in other words, the number of segments to be rendered is unspecified).
Also, since I am not very experienced with FFmpeg (I only use it to make simple file conversions with few arguments), I would like a description of the code you should use to do this, rather than just a piece of code that I won't necessarily understand, if possible.
Thank you in advance.

  


  P.S. Here's the context for why I'm trying to do this :
I would like to sample a song into single-bar loops automatically, instead of having to chop them manually using a DAW. All I want to do is align the first beat of the song to the beat grid in my DAW, and then export that audio file and use it to generate one-bar loops in FFmpeg.

  


  In the future, I will try to do something like a batch command in which one can specify the tempo and key signature, and it will generate the loops using FFmpeg automatically (as long as the loop is aligned to the beat grid, as I've mentioned earlier). 😀