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• php stream any mp4

  3 avril 2012, par GRaecuS

  I'm developing a web app that converts videos and allows to play them through Flowplayer.

  On the current status, I use ffmpeg to convert the videos to mp4 and qtfaststart to fix their metadata for streaming. Everything is working smoothly as I can download any converted mp4 and view it correctly.

  For serving the videos to Flowplayer, I use a php file which contains the following (summarized) code :

  header("Content-Type: {$mediatype}");
  
  
  
  if ( empty($_SERVER['HTTP_RANGE']) )
  
  {
  
      if ( $filetype == 'flv' && $seekPos != 0 )
  
      {
  
          header("Content-Length: " . ($filesize + 13));
  
          print('FLV');
  
          print(pack('C', 1));
  
          print(pack('C', 1));
  
          print(pack('N', 9));
  
          print(pack('N', 9));
  
      }
  
      else
  
      {           
  
          header("Content-Length: {$filesize}");
  
      }
  
  
  
      $fh = fopen($filepath, "rb") or die("Could not open file: {$filepath}");
  
  
  
      # seek to requested file position
  
      fseek($fh, $seekPos);
  
  
  
      # output file
  
      while(!feof($fh))
  
      {
  
          # output file without bandwidth limiting
  
          echo fread($fh, $filesize);
  
      }
  
      fclose($fh);
  
  }
  
  else //violes rfc2616, which requires ignoring  the header if it's invalid
  
  {   
  
      $fp = @fopen($file, 'rb');
  
  
  
      $size   = filesize($file); // File size
  
      $length = $size;           // Content length
  
      $start  = 0;               // Start byte
  
      $end    = $size - 1;       // End byte
  
      // Now that we've gotten so far without errors we send the accept range header
  
      /* At the moment we only support single ranges.
  
       * Multiple ranges requires some more work to ensure it works correctly
  
       * and comply with the spesifications: http://www.w3.org/Protocols/rfc2616/rfc2616-sec19.html#sec19.2
  
       *
  
       * Multirange support annouces itself with:
  
       * header('Accept-Ranges: bytes');
  
       *
  
       * Multirange content must be sent with multipart/byteranges mediatype,
  
       * (mediatype = mimetype)
  
       * as well as a boundry header to indicate the various chunks of data.
  
       */
  
      header("Accept-Ranges: 0-$length");
  
      // header('Accept-Ranges: bytes');
  
      // multipart/byteranges
  
      // http://www.w3.org/Protocols/rfc2616/rfc2616-sec19.html#sec19.2
  
      if (isset($_SERVER['HTTP_RANGE']))
  
      {
  
          $c_start = $start;
  
          $c_end   = $end;
  
          // Extract the range string
  
          list(, $range) = explode('=', $_SERVER['HTTP_RANGE'], 2);
  
          // Make sure the client hasn't sent us a multibyte range
  
          if (strpos($range, ',') !== false)
  
          {
  
              // (?) Shoud this be issued here, or should the first
  
              // range be used? Or should the header be ignored and
  
              // we output the whole content?
  
              header('HTTP/1.1 416 Requested Range Not Satisfiable');
  
              header("Content-Range: bytes $start-$end/$size");
  
              // (?) Echo some info to the client?
  
              exit;
  
          }
  
          // If the range starts with an '-' we start from the beginning
  
          // If not, we forward the file pointer
  
          // And make sure to get the end byte if spesified
  
          if ($range0 == '-')
  
          {
  
  
  
              // The n-number of the last bytes is requested
  
              $c_start = $size - substr($range, 1);
  
          }
  
          else
  
          {
  
              $range  = explode('-', $range);
  
              $c_start = $range[0];
  
              $c_end   = (isset($range[1]) && is_numeric($range[1])) ? $range[1] : $size;
  
          }
  
          /* Check the range and make sure it's treated according to the specs.
  
           * http://www.w3.org/Protocols/rfc2616/rfc2616-sec14.html
  
           */
  
          // End bytes can not be larger than $end.
  
          $c_end = ($c_end > $end) ? $end : $c_end;
  
          // Validate the requested range and return an error if it's not correct.
  
          if ($c_start > $c_end || $c_start > $size - 1 || $c_end >= $size)
  
          {
  
              header('HTTP/1.1 416 Requested Range Not Satisfiable');
  
              header("Content-Range: bytes $start-$end/$size");
  
              // (?) Echo some info to the client?
  
              exit;
  
          }
  
          $start  = $c_start;
  
          $end    = $c_end;
  
          $length = $end - $start + 1; // Calculate new content length
  
          fseek($fp, $start);
  
          header('HTTP/1.1 206 Partial Content');
  
      }
  
  
  
      // Notify the client the byte range we'll be outputting
  
      header("Content-Range: bytes $start-$end/$size");
  
      header("Content-Length: $length");
  
  
  
      // Start buffered download
  
      $buffer = 1024 * 8;
  
      while(!feof($fp) && ($p = ftell($fp)) <= $end)
  
      {
  
          if ($p + $buffer > $end)
  
          {
  
              // In case we're only outputtin a chunk, make sure we don't
  
              // read past the length
  
              $buffer = $end - $p + 1;
  
          }
  
          set_time_limit(0); // Reset time limit for big files
  
          echo fread($fp, $buffer);
  
          flush(); // Free up memory. Otherwise large files will trigger PHP's memory limit.
  
      }
  
  
  
      fclose($fp);
  
  }

  Unfortunately, it is working only for the majority of the videos. For some of them, Flowplayer keeps returning Error 200, even though they were encoded correctly.

  How can I fix this ? Is it a coding problem or those videos are faulty ?

• Computer crashing when using python tools in same script

  5 février 2023, par SL1997

  I am attempting to use the speech recognition toolkit VOSK and the speech diarization package Resemblyzer to transcibe audio and then identify the speakers in the audio.

  


  Tools :

  


  https://github.com/alphacep/vosk-api
  
https://github.com/resemble-ai/Resemblyzer

  


  I can do both things individually but run into issues when trying to do them when running the one python script.

  


  I used the following guide when setting up the diarization system :

  


  https://medium.com/saarthi-ai/who-spoke-when-build-your-own-speaker-diarization-module-from-scratch-e7d725ee279

  


  Computer specs are as follows :

  


  Intel(R) Core(TM) i3-7100 CPU @ 3.90GHz, 3912 Mhz, 2 Core(s), 4 Logical Processor(s)
  
32GB RAM

  


  The following is my code, I am not to sure if using threading is appropriate or if I even implemented it correctly, how can I best optimize this code as to achieve the results I am looking for and not crash.

  


  from vosk import Model, KaldiRecognizer
from pydub import AudioSegment
import json
import sys
import os
import subprocess
import datetime
from resemblyzer import preprocess_wav, VoiceEncoder
from pathlib import Path
from resemblyzer.hparams import sampling_rate
from spectralcluster import SpectralClusterer
import threading
import queue
import gc



def recognition(queue, audio, FRAME_RATE):

    model = Model("Vosk_Models/vosk-model-small-en-us-0.15")

    rec = KaldiRecognizer(model, FRAME_RATE)
    rec.SetWords(True)

    rec.AcceptWaveform(audio.raw_data)
    result = rec.Result()

    transcript = json.loads(result)#["text"]

    #return transcript
    queue.put(transcript)



def diarization(queue, audio):

    wav = preprocess_wav(audio)
    encoder = VoiceEncoder("cpu")
    _, cont_embeds, wav_splits = encoder.embed_utterance(wav, return_partials=True, rate=16)
    print(cont_embeds.shape)

    clusterer = SpectralClusterer(
        min_clusters=2,
        max_clusters=100,
        p_percentile=0.90,
        gaussian_blur_sigma=1)

    labels = clusterer.predict(cont_embeds)

    def create_labelling(labels, wav_splits):

        times = [((s.start + s.stop) / 2) / sampling_rate for s in wav_splits]
        labelling = []
        start_time = 0

        for i, time in enumerate(times):
            if i > 0 and labels[i] != labels[i - 1]:
                temp = [str(labels[i - 1]), start_time, time]
                labelling.append(tuple(temp))
                start_time = time
            if i == len(times) - 1:
                temp = [str(labels[i]), start_time, time]
                labelling.append(tuple(temp))

        return labelling

    #return
    labelling = create_labelling(labels, wav_splits)
    queue.put(labelling)



def identify_speaker(queue1, queue2):

    transcript = queue1.get()
    labelling = queue2.get()

    for speaker in labelling:

        speakerID = speaker[0]
        speakerStart = speaker[1]
        speakerEnd = speaker[2]

        result = transcript['result']
        words = [r['word'] for r in result if speakerStart < r['start'] < speakerEnd]
        #return
        print("Speaker",speakerID,":",' '.join(words), "\n")





def main():

    queue1 = queue.Queue()
    queue2 = queue.Queue()

    FRAME_RATE = 16000
    CHANNELS = 1

    podcast = AudioSegment.from_mp3("Podcast_Audio/Film-Release-Clip.mp3")
    podcast = podcast.set_channels(CHANNELS)
    podcast = podcast.set_frame_rate(FRAME_RATE)

    first_thread = threading.Thread(target=recognition, args=(queue1, podcast, FRAME_RATE))
    second_thread = threading.Thread(target=diarization, args=(queue2, podcast))
    third_thread = threading.Thread(target=identify_speaker, args=(queue1, queue2))

    first_thread.start()
    first_thread.join()
    gc.collect()

    second_thread.start()
    second_thread.join()
    gc.collect()

    third_thread.start()
    third_thread.join()
    gc.collect()

    # transcript = recognition(podcast,FRAME_RATE)
    #
    # labelling = diarization(podcast)
    #
    # print(identify_speaker(transcript, labelling))


if __name__ == '__main__':
    main()


  


  When I say crash I mean everything freezes, I have to hold down the power button on the desktop and turn it back on again. No blue/blank screen, just frozen in my IDE looking at my code. Any help in resolving this issue would be greatly appreciated.

  


• slicing and seeking extremeley small sections of video in ffmpeg

  20 février 2021, par Zarc Rowden

  I am writing a program that maps midi data to timestamps in a video. The end result is a kind of automatic generation of audio visuals for beat making or techno heads. The program takes in midi, slices a video into chunks based on the midi events and mappings and finally joins the slices into a video with 1:1 timing of monophonic midi notes to sections of a video.

  


  When it is successful, the result is very cool and watching the video jump around and lock in to midi notes is very interesting

  


  However, I am affraid that the ffmpeg commands I use are not giving exact results.

  


  The code I feed to ffmpeg looks like this

  


  EVENTS : left is midinote number, right is time from start of recording in which note occurs.

  


    [{"note"=>"start", "timestamp"=>0.0},
   {"note"=>48, "timestamp"=>0.5700000037904829},
   {"note"=>51, "timestamp"=>383.7100000018836},
   {"note"=>45, "timestamp"=>884.3500000002678},
   {"note"=>48, "timestamp"=>999.0449999968405},
   {"note"=>51, "timestamp"=>1383.544999996957},
   {"note"=>45, "timestamp"=>1884.2599999989034},
   {"note"=>48, "timestamp"=>1998.890000002575},
   {"note"=>51, "timestamp"=>2383.4199999982957},
   {"note"=>45, "timestamp"=>2884.1000000029453},
   {"note"=>48, "timestamp"=>2998.7200000032317},
   {"note"=>51, "timestamp"=>3383.2800000018324},
   {"note"=>45, "timestamp"=>3883.894999999029},
   {"note"=>48, "timestamp"=>3998.6250000001746},
   {"note"=>51, "timestamp"=>4384.550000002491},
   {"note"=>45, "timestamp"=>4883.780000003753},
   {"note"=>48, "timestamp"=>4998.404999998456},
   {"note"=>51, "timestamp"=>5384.39500000095},
   {"note"=>45, "timestamp"=>5883.565000003728},
   {"note"=>48, "timestamp"=>5998.464999996941},
   {"note"=>51, "timestamp"=>6384.254999997211},
   {"note"=>45, "timestamp"=>6883.4550000028685},
   {"note"=>48, "timestamp"=>6998.585000001185},
   {"note"=>51, "timestamp"=>7384.055000002263},
   {"note"=>45, "timestamp"=>7883.249999998952},],


  


  MAPPINGS : left side is midi note, right is timestamp in seconds

  


  {
 48=>234.3489,
 45=>124.334489,
 51=>2789.34,
}


  


  That Events are a sequential array of midi notes and time taken from recordings or standard midi file. The number is in milliseconds but I convert for ffmpeg before feeding the arguments.

  


  The mappings are just in seconds and tell the program what to show when certain midi notes are encountered as we loop through the events and begin slicing the video.

  


  The command I send to ffmpeg is constructed like this :

  


  "ffmpeg -an -y -ss #{begin_at} -i #{project_tempfile_url} -t #{slice_duration} -c:v libx264 #{temp_url}"


  


  When I concatenate these slices, they only look exact when my notes are very consistent like a kickdrum doing 4/4 rythms. Anything too fast or varied creates unpleasant results.

  


  Is there a specific set of commands that will tell ffmpeg to cut down to the frame ? I think keyframe are not an ideal answer but not sure. I also think I can adjust by making sure that I only ever map the notes to keyframes, I can settle for it but it would be great if I could just cut almost anywhere between start and end like ANYWHERE like

  


    rand(0...video.length)
  # and then have
  332.3253613134


  


  But I may just be dreaming :P

  


  Do you think that I would be better off writing a custom c program to cut frames like this ? I understand that frame rates could be an issue and that there may actually not be any data at 7.34667898999 seconds and that it might be here instead : 7.356788722342 and that ffmpeg probably searches for the nearest frame from whatever timestamp you input, but I feel like there must be a way to get good results still despite these limitations.

  


  Thank you so much in advance for those who take the time to read this and understand this issue.