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• How can I remove the zero padding at the start of a .mp3 file generated by ffmpeg conversion ?

  22 juin 2022, par Miguel de Sousa

  After reading this and this, I understand that the .mp3 encoder appends zeros at the start and at the end of an audio.

  


  With this approach, the encoder can pass a sliding analysis window through the initial and final audio samples, just like in the rest of the audio.

  


  This sliding window is associated with a Fourier Transform-like algorithm, but I will not get into this technical detail.

  


  My problem is : When I generate an .mp3 with ffmpeg like :

  


  MacBook-Air-de-Miguel:faixas miguel$ ffmpeg -i primeira_caf.caf primeira_agora_vai.mp3
ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.3 (clang-1103.0.32.62)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_1 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[caf @ 0x7fe205009000] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, caf, from 'primeira_caf.caf':
  Duration: 00:00:08.73, start: 0.000000, bitrate: 1414 kb/s
    Stream #0:0: Audio: pcm_s16le (lpcm / 0x6D63706C), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'primeira_agora_vai.mp3':
  Metadata:
    TSSE            : Lavf58.45.100
    Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc58.91.100 libmp3lame
size=     137kB time=00:00:08.75 bitrate= 128.6kbits/s speed=  61x    
video:0kB audio:137kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.180156%


  


  my generated .mp3 has +- 50ms of latency. I'm not 100% sure that all this latency is caused by the zero padding that I mentioned.

  


  Converting .caf to other audio formats like .wav and .ogg do not give me this problem.

  


  Considering .mp3 as a constraint, is there any simple way to generate a .mp3 with zero latency ? Maybe a ffmpeg argument that cuts off the zero padding at the start ?

  


  If not, is it possible to do it manually ? How can I know or calculate how many samples should be cut off ?

  


  I have also tried sox. It adds 25ms of latency instead of 50ms.

  


• Pims.open Throws "UnkownFormat Error" with "Invalid Argument" On One Machine But Not on Another

  12 août 2021, par KaceBellamy

  I'm trying to do some video processing for a physics experiment, and I want to do it on my much more powerful Windows desktop computer as opposed to my Mac laptop.

  


  The following code works like a dream when run as a Jupyter notebook on my Mac :

  


  import matplotlib as mpl
from mpl_toolkits import mplot3d
import pims
import trackpy as tp 

@pims.pipeline
def gray(image):
    return image[:, :, 1]  # Take just the green channel

frames = gray(pims.open('output.mp4'))


  


  but on my Windows machine I get this error :

  


  ---------------------------------------------------------------------------&#xA;UnknownFormatError                        Traceback (most recent call last)&#xA;~\AppData\Local\Temp/ipykernel_12152/704954007.py in <module>&#xA;      1 #Actually converts the video. Might be VERY processor intensive... or not?&#xA;----> 2 frames = gray(pims.open(&#x27;output.mp4&#x27;)) #Make the File Name whatever file you like!&#xA;&#xA;~\miniconda3\lib\site-packages\pims\api.py in open(sequence, **kwargs)&#xA;    207             warn(message)&#xA;    208             exceptions &#x2B;= message &#x2B; &#x27;\n&#x27;&#xA;--> 209     raise UnknownFormatError("All handlers returned exceptions:\n" &#x2B; exceptions)&#xA;    210 &#xA;    211 &#xA;&#xA;UnknownFormatError: All handlers returned exceptions:&#xA;<class> errored: [Errno 22] Invalid argument: &#x27;output.mp4&#x27;&#xA;<class> errored: [Errno 22] Invalid argument: &#x27;output.mp4&#x27;&#xA;<class> errored: Could not load meta information&#xA;=== stderr ===&#xA;&#xA;ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers&#xA;  built with gcc 9.2.1 (GCC) 20200122&#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-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-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt&#xA;  libavutil      56. 31.100 / 56. 31.100&#xA;  libavcodec     58. 54.100 / 58. 54.100&#xA;  libavformat    58. 29.100 / 58. 29.100&#xA;  libavdevice    58.  8.100 / 58.  8.100&#xA;  libavfilter     7. 57.100 /  7. 57.100&#xA;  libswscale      5.  5.100 /  5.  5.100&#xA;  libswresample   3.  5.100 /  3.  5.100&#xA;  libpostproc    55.  5.100 / 55.  5.100&#xA;C:\Users\Callum\OneDrive - The University of Chicago\output.mp4: Invalid argument&#xA;<class> errored: MoviePy error: failed to read the duration of file output.mp4.&#xA;Here are the file infos returned by ffmpeg:&#xA;&#xA;ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers&#xA;  built with gcc 9.2.1 (GCC) 20200122&#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-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-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt&#xA;  libavutil      56. 31.100 / 56. 31.100&#xA;  libavcodec     58. 54.100 / 58. 54.100&#xA;  libavformat    58. 29.100 / 58. 29.100&#xA;  libavdevice    58.  8.100 / 58.  8.100&#xA;  libavfilter     7. 57.100 /  7. 57.100&#xA;  libswscale      5.  5.100 /  5.  5.100&#xA;  libswresample   3.  5.100 /  3.  5.100&#xA;  libpostproc    55.  5.100 / 55.  5.100&#xA;output.mp4: Invalid argument&#xA;</class></class></class></class></module>

  


  output.mp4 is just a run of the mill video file taken on a CCD camera ; I've tried converting it to a .MOV, I've tried other video files taken on different cameras, and I've tried running the file through FFmpeg to impose a 30 fps framerate ; everything I've tried works fine on my Mac and throws the above error on my Windows machine.

  


  For reference, I installed necessary packages for this code on both machines this morning, so it should all be up to date and the same on both.

  


  Any ideas as to what's up ? Thanks !

  


• Managing Music Playback Channels

  30 juin 2013, par Multimedia Mike — General

  My Game Music Appreciation site allows users to interact with old video game music by toggling various channels, as long as the underlying synthesizer engine supports it.

  
  
  

  Users often find their way to the Nintendo DS section pretty quickly. This is when they notice an obnoxious quirk with the channel toggling feature : specifically, one channel doesn’t seem to map to a particular instrument or track.

  When it comes to computer music playback methodologies, I have long observed that there are 2 general strategies : Fixed channel and dynamic channel allocation.

  Fixed Channel Approach
  One of my primary sources of computer-based entertainment used to be watching music. Sure I listened to it as well. But for things like Amiga MOD files and related tracker formats, there was a rich ecosystem of fun music playback programs that visualized the music. There exist music visualization modes in various music players these days (such as iTunes and Windows Media Player), but those largely just show you a single wave form. These files were real time syntheses based on multiple audio channels and usually showed some form of analysis for each channel. My personal favorite was Cubic Player :

  
  
  

  Most of these players supported the concept of masking individual channels. In doing so, the user could isolate, study, and enjoy different components of the song. For many 4-channel Amiga MOD files, I observed that the common arrangement was to use the 4 channels for beat (percussion track), bass line, chords, and melody. Thus, it was easy to just listen to, e.g., the bass line in isolation.

  MODs and similar formats specified precisely which digital audio sample to play at what time and on which specific audio channel. To view the internals of one of these formats, one gets the impression that they contain an extremely computer-centric view of music.

  Dynamic Channel Allocation Algorithm
  MODs et al. enjoyed a lot of popularity, but the standard for computer music is MIDI. While MOD and friends took a computer-centric view of music, MIDI takes, well, a music-centric view of music.

  There are MIDI visualization programs as well. The one that came with my Gravis Ultrasound was called PLAYMIDI.EXE. It looked like this…

  
  
  

  … and it confused me. There are 16 distinct channels being visualized but some channels are shown playing multiple notes. When I dug into the technical details, I learned that MIDI just specifies what notes need to be played, at what times and frequencies and using which instrument samples, and it was the MIDI playback program’s job to make it happen.

  Thus, if a MIDI file specifies that track 1 should play a C major chord consisting of notes C, E, and G, it would transmit events “key-on C ; delta time 0 ; key-on E ; delta time 0 ; key-on G ; delta time … ; [other commands]“. If the playback program has access to multiple channels (say, up to 32, in the case of the GUS), the intuitive approach would be to maintain a pool of all available channels. Then, when it’s time to process the “key-on C” event, fetch the first available channel from the pool, mark it as in-use, play C on the channel, and return that channel to the pool when either the sample runs its course or the corresponding “key-off C” event is encountered in the MIDI command stream.

  About That Game Music
  Circling back around to my game music website, numerous supported systems use the fixed channel approach for playback while others use dynamic channel allocation approach, including evey Nintendo DS game I have so far analyzed.

  Which approach is better ? As in many technical matters, there are trade-offs either way. For many systems, the fixed channel approach is necessary because for many older audio synthesis systems, different channels had very specific purposes. The 8-bit NES had 5 channels : 2 square wave generators (used musically for melody/treble), 1 triangle wave generator (usually used for bass line), a noise generator (subverted for all manner of percussive sounds), and a limited digital channel (was sometimes assigned richer percussive sounds). Dynamic channel allocation wouldn’t work here.

  But the dynamic approach works great on hardware with 16 digital channels available like, for example, the Nintendo DS. Digital channels are very general-purpose. What about the SNES, with its 8 digital channels ? Either approach could work. In practice, most games used a fixed channel approach : Games might use 4-6 channels for music while reserving the remainder for various in-game sound effects. Some notable exceptions to this pattern were David Wise’s compositions for Rare’s SNES games (think Battletoads and the various Donkey Kong Country titles). These clearly use some dynamic channel approach since masking all but one channel will give you a variety of instrument sounds.

  Epilogue
  There ! That took a long time to explain but I find it fascinating for some reason. I need to distill it down to far fewer words because I want to make it a FAQ on my website for “Why can’t I isolate specific tracks for Nintendo DS games ?”

  Actually, perhaps I should remove the ability to toggle Nintendo DS channels in the first place. Here’s a funny tale of needless work : I found the Vio2sf engine for synthesizing Nintendo DS music and incorporated it into the program. It didn’t support toggling of individual channels so I figured out a way to add that feature to the engine. And then I noticed that most Nintendo DS games render that feature moot. After I released the webapp, I learned that I was out of date on the Vio2sf engine. The final insult was that the latest version already supports channel toggling. So I did the work for nothing. But then again, since I want to remove that feature from the UI, doubly so.