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• Building OpenCV 2.4.11 on Linux with `libopencv_ffmpeg.so`

  28 décembre 2016, par SomethingSomething

  When downloading OpenCV sources, no matter whether it is for Windows or for Linux, the sources directory contains only the Windows already-compiled shared library 3rdparty/ffmpeg/opencv_ffmpeg.dll, that is essential for the VideoCapture module to work on the Python portings for OpenCV.

  In Linux, as this file is not provided, you’d expect it to be automatically built when you build OpenCV, especially when in CMakeCache.txt, you can spot the line WITH_FFMPEG:BOOL=ON.

  It is of course not built... Any idea ?

• codecs.ascii_decode(input, self.errors)[0] UnicodeDecodeError : 'ascii' codec can't decode byte 0xc2 in position 318 : ordinal not in range(128)

  6 janvier 2017, par dtidy

  I am trying to open and readlines a .txt file that contains a large amount of text. Below is my code, i dont know how to solve this problem. Any help would be very appreciated.

  file = input("Please enter a .txt file: ")
  
  myfile = open(file)
  
  x = myfile.readlines()
  
  print (x)

  when i enter the .txt file this is the full error message is displayed below :

  line 10, in <module> x = myfile.readlines()
  
  line 26, in decode return codecs.ascii_decode(input, self.errors)[0]
  
  UnicodeDecodeError: 'ascii' codec can't decode byte 0xc2 in position 318: ordinal not in range(128)
  
  </module>

• Raspberry Pi : how to configure audio sampling rate in ALSA, ffmpeg, mp3 encoding pipeline

  30 janvier 2014, par user3252022

  I have a Pi with the camera module and a USB sound card. The intent is to use this combination to generate the MPEG2-TS segments for HTTP Live Streaming (Apple's HLS.)

  raspivid -ih -e -w 1280 -h 720 -fps 25 -vf -hf -t 0 -b 1800000 -o - | \
  
  ffmpeg -y \
  
    -i - \
  
    -ac 1 \
  
    -f alsa -ar 22050 -i hw:0,0 \
  
    -c:v copy \
  
    -c:a libshine -strict -2 \
  
    -map 0:0 -map 1:0 \
  
    -f ssegment \
  
    -segment_list_entry_prefix live/ \
  
    -segment_time 10 \
  
    -segment_format mpegts \
  
    -segment_list "$base/live.m3u8" \
  
    -segment_list_size 20 \
  
    -segment_list_flags +live \
  
    -segment_list_type m3u8 \
  
    -segment_wrap 200 \
  
    live/%08d.ts"

  The above basically works. The heavy lifting of H.264 video encoding is done in hardware. However, MP3 encoding in software brings the ARM core to its knees. The option :

  -f alsa -ar 22050 -i hw:0,0

  is intended to lighten the CPU loading by reducing the ALSA audio sampling rate - from the 48kHz default down to 22.05kHz. The sampling rate, however, won't budge to anything below 44.1kHz.

  Any idea what I'm doing wrong ? Why can't I tell ALSA to sample at 22.05kHz (or 8kHz for that matter ?)

  Here's an excerpt output from running the encoding pipeline :

  ffmpeg version N-60125-g4b8c599 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Jan 29 2014 10:16:02 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  
    configuration: --enable-gpl --enable-nonfree --enable-libshine
  
    ...   
  
  Input #0, h264, from 'pipe:':
  
    Duration: N/A, bitrate: N/A
  
      Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 25 fps, 25 tbr, 1200k tbn, 50 tbc
  
  Guessed Channel Layout for  Input Stream #1.0 : mono
  
  Input #1, alsa, from 'hw:0,0':
  
    Duration: N/A, start: 1391066215.835499, bitrate: 705 kb/s
  
      Stream #1:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
  
      ...