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Sur d’autres sites (11997)

  • Mimic Audacity amplification with Pydub

    16 août 2022, par Unisionzz

    For my music library I have used Audacity for recent years to amplify the music to similar levels of loudness ; technically speaking this is not completely true, but for me it is sufficient. However, as it is tedious to do this all by hand, I decided to write a Python code to automate this process for me. The code after the imported package(s) and defined functions will run in a loop in which the filename changes depending on which song is processed.

    


    The difficult part is that I have not yet been able to find a consistent way to amplify different songs so that when the output files are put through Audacity, it will not want to change the amplitude by more than 0.1 dB(FS).

    


    Below are two attempts which seem to have come closest to the desirable output ; other methods that I have tried were either less succesfull or resulted in clipping.

    


    The first attempt finds the maximum dBFS of the song and then applies a gain in order for the maximum dBFS to equal 0 (I have also tried this method with sound.dBFS and sound.apply_gain, but results seem more mixed than the attempt below) :

    


    from pydub import AudioSegment

def change_amplitude(sound, target_dBFS):
    change_in_dBFS = target_dBFS - sound.max_dBFS
    return sound.apply_gain_stereo(change_in_dBFS)

# Audio is gathered from a hard coded path
s = AudioSegment.from_file(Dir+filename+".mp3", "mp3")
amp_s = change_amplitude(s, 0)
amp_s.export(Dir+filename+".mp3", format = "mp3")


    


    The second attempt finds the amplitude and the maximum allowable amplitude (before clipping), recalculates both to dB and then adds the dB_diff to the sound :

    


    import numpy as np
from pydub import AudioSegment

s = AudioSegment.from_file(Dir+filename+".mp3", "mp3")

# Get dB amplitude of song and maximum allowable value
dB_sound = 20*np.log10(s.max)
dB_max = 20*np.log10(s.max_possible_amplitude)
dB_diff = dB_max - dB_sound

amp_sound = s + dB_diff


    


    Summarizing, I would like to import a music file, amplify it similar to Audacity amplification and then export the file again.

    


  • Trouble with converting webm into mp3 with pydub in python

    15 août 2020, par rc_marty

    so basically I want to convert song what I downloaded from youtube in webm and convert to into mp3

    


    when I wanted export song just with song.export("neco.mp3") it didn't work too

    


    I have in workfolder ffmpeg.exe and ffprobe.exe

    


    here is the code

    


    from pydub import AudioSegment

song = AudioSegment.from_file(downloaded.webm,"webm")
print("Loaded")
song.export("neco.mp3", format="mp3", bitrate="320k")
print("Converted and saved")


    


    here is the output of the console

    


    Loaded&#xA;Traceback (most recent call last):&#xA;  File "e:/martan/projekty/Python/programek na pisnicky/songDownloader.py", line 188, in <module>&#xA;    song.export("neco.mp3", format="mp3", bitrate="320k")&#xA;  File "C:\Users\BIBRAIN\AppData\Local\Programs\Python\Python38\lib\site-packages\pydub\audio_segment.py", line 911, in export&#xA;    raise CouldntEncodeError(&#xA;pydub.exceptions.CouldntEncodeError: Encoding failed. ffmpeg/avlib returned error code: 1&#xA;&#xA;Command:[&#x27;ffmpeg&#x27;, &#x27;-y&#x27;, &#x27;-f&#x27;, &#x27;wav&#x27;, &#x27;-i&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpo20ooz_z&#x27;, &#x27;-b:a&#x27;, &#x27;320k&#x27;, &#x27;-f&#x27;, &#x27;mp3&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpiqpl57g7&#x27;]&#xA;&#xA;Output from ffmpeg/avlib:&#xA;&#xA;ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers&#xA;  built with gcc 10.2.1 (GCC) 20200726&#xA;  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf&#xA;  libavutil      56. 51.100 / 56. 51.100&#xA;  libavcodec     58. 91.100 / 58. 91.100&#xA;  libavformat    58. 45.100 / 58. 45.100&#xA;  libavdevice    58. 10.100 / 58. 10.100&#xA;  libavfilter     7. 85.100 /  7. 85.100&#xA;  libswscale      5.  7.100 /  5.  7.100&#xA;  libswresample   3.  7.100 /  3.  7.100&#xA;  libpostproc    55.  7.100 / 55.  7.100&#xA;Guessed Channel Layout for Input Stream #0.0 : stereo&#xA;Input #0, wav, from &#x27;C:\Users\BIBRAIN\AppData\Local\Temp\tmpo20ooz_z&#x27;:&#xA;  Duration: 00:03:54.71, bitrate: 3072 kb/s&#xA;    Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (mp3_mf))&#xA;Press [q] to stop, [?] for help&#xA;[mp3_mf @ 00000000004686c0] could not find any MFT for the given media type&#xA;[mp3_mf @ 00000000004686c0] could not create MFT&#xA;Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height&#xA;Conversion failed!&#xA;</module>

    &#xA;

    I think it is something with codec but I have no idea what to do

    &#xA;

  • Getting accurate time from FFMPeg with Objective C (Audio Queue Services)

    2 avril 2012, par Winston

    My iPhone app plays an audio file using FFMPeg.

    I'm getting the elapsed time (to show to user) from the playing audio (in minutes and seconds after converting from microseconds, given by FFMPeg) like so :

    AudioTimeStamp currentTimeStamp;
    
AudioQueueGetCurrentTime (audioQueue, NULL, &amp;currentTimeStamp, NULL);
    

    
getFFMPEGtime = currentTimeStamp.mSampleTime/self.basicAudioDescription.mSampleRate;
    

    
self.currentAudioTime = [NSString stringWithFormat: @"%02d:%02d",
    
                            (int) getFFMPEGtime / (int)60000000,
    
                            (int) ((getFFMPEGtime % 60000000)/1000000)];

    Everything works fine, but when I scrub back or forward to play another portion of the song, the elapsed time will go back to zero, no matter the current position. The timer will always zero out.

    I know I'm suposed to do some math to keep track of the old time and the new time, maybe constructing another clock or so, perhaps implementing another callback function, etc... I'm not sure what way I should go.

    My questions are :

    1) What's the best approach to keep track of the elapsed time when going back/forward in a song, avoiding the clock to always going back to zero ?

    2) Should I look deeply into FFMPeg functions or should I stick with Objective-C and Cocoa Touch for solving this problem ?

    Please, I need some advices/ideas from experienced programmers. I'm stuck. Thanks beforehand !

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