_STATISTICS_WRITING_APP-eng: no_variable_data _STATISTICS_WRITING_APP: no_variable_data Here's what I get for output: ffmpeg version 2.8.4 Copyright (c) 2000-2015 the FFmpeg developersĬonfiguration: -enable-gpl -enable-version3 -disable-w32threads -enable-avisynth -enable-bzlib -enable-fontconfig -enable-frei0r -enable-gnutls -enable-iconv -enable-libass -enable-libbluray -enable-libbs2b -enable-libcaca -enable-libdcadec -enable-libfreetype -enable-libgme -enable-libgsm -enable-libilbc -enable-libmodplug -enable-libmp3lame -enable-libopencore-amrnb -enable-libopencore-amrwb -enable-libopenjpeg -enable-libopus -enable-librtmp -enable-libschroedinger -enable-libsoxr -enable-libspeex -enable-libtheora -enable-libtwolame -enable-libvidstab -enable-libvo-aacenc -enable-libvo-amrwbenc -enable-libvorbis -enable-libvpx -enable-libwavpack -enable-libwebp -enable-libx264 -enable-libx265 -enable-libxavs -enable-libxvid -enable-lzma -enable-decklink -enable-zlibĭuration: 00:24:04.78, start: 0.000000, bitrate: 3196 kb/s
This approach is a simpler and faster alternative to the classical convert, save then read.Output = subprocess.check_output()
This blog post introduced a small example of reading the ffmpeg command pipe output and parsing the resulting wave data into a numpy array. In the above table we notice that the audio data bytes start at byte 45 and therefore the first 44 bytes are the offset. Sample values are given above for a 16-bit stereo source. (BitsPerSample * Channels) / 8.1 - 8 bit mono2 - 8 bit stereo/16 bit mono4 - 16 bit stereo (Sample Rate * BitsPerSample * Channels) / 8. Sample Rate = Number of Samples per second, or Hertz. Type of format (1 is PCM) - 2 byte integer For our purposes, it always equals "WAVE".įormat chunk marker. Typically, you'd fill this in after creation.įile Type Header. Size of the overall file - 8 bytes, in bytes (32-bit integer). Table 4: Wave file structure and content # In order to know how many bytes to ignore, we need to examine the following table 3 of the wave data: The read data is essentially a wave file data including the header which must be ignored when passing the data to the We also define a buffer size to receive the read data. Note that the used FFmpeg command is slightly changed, to define the channel of choice and the encoding to use.įor more on that you can refer either to 1 or 2.
uint16, offset = 8 * 44 ) 21 sig, fs = audio_np, target_fs stderr ) 18 19 # read signal as numpy array and assign sampling rate 20 audio_np = np. run ( ffmpeg_command, 12 stdout = subprocess. 1 import subprocess 2 import numpy as np 3 4 # init command 5 ffmpeg_command = 9 10 # excute ffmpeg command 11 pipe = subprocess.
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