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• Could not read frame error when trying to decompress mp4 file with ffmpeg and Python's threading module

  23 janvier 2017, par mdornfe1

  I’m training constitutional neural networks with video data. So far the bottle neck of my application is decompressing the mp4 files before passing the images to the CNN for training. I had the idea to try to have multiple cpu threads decompress the images concurrently and having one thread pass images to the CNN for training. I made a class VideoStream which makes connection to the mp4 file using the ImageIO module which is built on top of ffmpeg. The structure of my program is a follows :

  1) Generate random ints which represent the frame numbers of the mp4 file that will be used in training. Store these ints in list frame_idxs.

  2) Pass this list of ints and an empty list called frame_queue to the worker function decompress_video_data.

  3) Each worker function makes a connection to the mp4 file using VideoStream.

  4) Each worker function then pops of elements of frame_idxs, decompresses that frame, and then stores that frame as numpy array in list frame_queue.

  Here is the code

  import numpy as np
  
  import os, threading, multiprocessing
  
  
  
  
  
  def decompress_video_data(frame_queue, frame_idxs, full_path):
  
      vs = VideoStream(full_path)
  
      while len(frame_idxs) >1 0:
  
          i = frame_idxs.pop()
  
          frame = vs[i]
  
          frame_queue.append(frame)
  
  
  
  video_folder = '/mnt/data_drive/frame_viewer_client'
  
  video_files = os.listdir(video_folder)
  
  video_file = video_files[0]
  
  full_path = os.path.join(video_folder, video_file)
  
  vs = VideoStream(full_path)
  
  
  
  num_samples = 1000
  
  batch_size = 1
  
  frame_queue = []
  
  decompress_threads = []
  
  frame_idxs = list(np.random.randint(0, len(vs), 
  
      size = batch_size * num_samples))
  
  num_cores = multiprocessing.cpu_count()
  
  
  
  for n in range(num_cores - 1):
  
      decompress_thread = threading.Thread(target=decompress_video_data, 
  
          args=(frame_queue, frame_idxs, full_path))
  
      decompress_threads.append(decompress_thread)
  
      decompress_thread.start()

  The program will sucessfuly decompress approximately 200 frames, and then ImageIO will throw an RuntimeError : Could not read frame. The full error is here. Does anyone know why this is happening ? Is it possible to do what I’m trying to do ? Does ffmpeg just not work with multi threading ?

• Get audio amplitude list using FFmpeg, which contains n (could be set manually) items [closed]

  26 juin 2024, par Xavier Hugo

  I am working on a recorder app and need to generate audio spectrum for my files. I need to get amplitude list from audio file so that I can draw audio wave with Canvas by my own.

  


  I have read the fantastic answer by llogan. However, I don't know how to set the time interval manually. By the way, when I try to run the command, I got a list full of negetive numbers : command-running-image. Hope someone can explain it.

  


• Emscripten and Web Audio API

  29 avril 2015, par Multimedia Mike — HTML5

  Ha ! They said it couldn’t be done ! Well, to be fair, I said it couldn’t be done. Or maybe that I just didn’t have any plans to do it. But I did it– I used Emscripten to cross-compile a CPU-intensive C/C++ codebase (Game Music Emu) to JavaScript. Then I leveraged the Web Audio API to output audio and visualize the audio using an HTML5 canvas.

  Want to see it in action ? Here’s a demonstration. Perhaps I will be able to expand the reach of my Game Music site when I can drop the odd Native Client plugin. This JS-based player works great on Chrome, Firefox, and Safari across desktop operating systems.

  But this endeavor was not without its challenges.

  Programmatically Generating Audio
  First, I needed to figure out the proper method for procedurally generating audio and making it available to output. Generally, there are 2 approaches for audio output :

  1. Sit in a loop and generate audio, writing it out via a blocking audio call
  2. Implement a callback that the audio system can invoke in order to generate more audio when needed

  Option #1 is not a good idea for an event-driven language like JavaScript. So I hunted through the rather flexible Web Audio API for a method that allowed something like approach #2. Callbacks are everywhere, after all.

  I eventually found what I was looking for with the ScriptProcessorNode. It seems to be intended to apply post-processing effects to audio streams. A program registers a callback which is passed configurable chunks of audio for processing. I subverted this by simply overwriting the input buffers with the audio generated by the Emscripten-compiled library.

  The ScriptProcessorNode interface is fairly well documented and works across multiple browsers. However, it is already marked as deprecated :

  Note : As of the August 29 2014 Web Audio API spec publication, this feature has been marked as deprecated, and is soon to be replaced by Audio Workers.

  Despite being marked as deprecated for 8 months as of this writing, there exists no appreciable amount of documentation for the successor API, these so-called Audio Workers.

  Vive la web standards !

  Visualize This
  The next problem was visualization. The Web Audio API provides the AnalyzerNode API for accessing both time and frequency domain data from a running audio stream (and fetching the data as both unsigned bytes or floating-point numbers, depending on what the application needs). This is a pretty neat idea. I just wish I could make the API work. The simple demos I could find worked well enough. But when I wired up a prototype to fetch and visualize the time-domain wave, all I got were center-point samples (an array of values that were all 128).

  Even if the API did work, I’m not sure if it would have been that useful. Per my reading of the AnalyserNode API, it only returns data as a single channel. Why would I want that ? My application supports audio with 2 channels. I want 2 channels of data for visualization.

  How To Synchronize
  So I rolled my own visualization solution by maintaining a circular buffer of audio when samples were being generated. Then, requestAnimationFrame() provided the rendering callbacks. The next problem was audio-visual sync. But that certainly is not unique to this situation– maintaining proper A/V sync is a perennial puzzle in real-time multimedia programming. I was able to glean enough timing information from the environment to achieve reasonable A/V sync (verify for yourself).

  Pause/Resume
  The next problem I encountered with the Web Audio API was pause/resume facilities, or the lack thereof. For all its bells and whistles, the API’s omission of such facilities seems most unusual, as if the design philosophy was, “Once the user starts playing audio, they will never, ever have cause to pause the audio.”

  Then again, I must understand that mine is not a use case that the design committee considered and I’m subverting the API in ways the designers didn’t intend. Typical use cases for this API seem to include such workloads as :

  • Downloading, decoding, and playing back a compressed audio stream via the network, applying effects, and visualizing the result
  • Accessing microphone input, applying effects, visualizing, encoding and sending the data across the network
  • Firing sound effects in a gaming application
  • MIDI playback via JavaScript (this honestly amazes me)

  What they did not seem to have in mind was what I am trying to do– synthesize audio in real time.

  I implemented pause/resume in a sub-par manner : pausing has the effect of generating 0 values when the ScriptProcessorNode callback is invoked, while also canceling any animation callbacks. Thus, audio output is technically still occurring, it’s just that the audio is pure silence. It’s not a great solution because CPU is still being used.

  Future Work
  I have a lot more player libraries to port to this new system. But I think I have a good framework set up.