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FFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files and streams. At its core is the command-line ffmpeg tool itself, designed for processing of video and audio files. It is widely used for format transcoding, basic editing (trimming and concatenation), video scaling, video post-production effects and standards compliance (SMPTE, ITU). FFmpeg also includes other tools: ffplay, a simple media player and ffprobe, a command-line tool to display media information. Among included libraries are libavcodec, an audio/video codec library used by many commercial and free software products, libavformat (Lavf), an audio/video container mux and demux library, and libavfilter, a library for enhancing and editing filters through a Gstreamer-like filtergraph. FFmpeg is part of the workflow of many other software projects, and its libraries are a core part of software media players such as VLC, and has been included in core processing for YouTube and Bilibili. Encoders and decoders for many audio and video file formats are included, making it highly useful for the transcoding of common and uncommon media files. FFmpeg is published under the LGPL-2.1-or-later or GPL-2.0-or-later, depending on which options are enabled. The project was started by Fabrice Bellard (using the pseudonym "Gérard Lantau") in 2000, and was led by Michael Niedermayer from 2004 until 2015. Some FFmpeg developers were also part of the MPlayer project. The name of the project is inspired by the MPEG video standards group, together with "FF" for "fast forward", so FFmpeg stands for "Fast Forward Moving Picture Experts Group". The logo uses a zigzag pattern that shows how MPEG video codecs handle entropy encoding. On March 13, 2011, a group of FFmpeg developers decided to fork the project under the name Libav. The event was related to an issue in project management, in which developers disagreed with the leadership of FFmpeg.
[2]Paolo Prandoni, [3]LCAV, EPFL
In this notebook we will implement a simple fractional resampler that
uses local Lagrange interpolation. The resampler can be used to perform
any rational sampling rate change (but beware of aliasing when
downsampling!)
In [1]:
import numpy as np
import IPython
from scipy.io import wavfile
Local Lagrange interpolation[4]¶
To perform resampling we will need to compute subsample values and, in [2]Paolo Prandoni, [3]LCAV, EPFL
This notebook shows a simple way to process audio files (in WAV format)
in a block-by-block fashion; this is useful in two cases
1. when you want to process a data stream in real time but you can
accept a small processing delay; in this case a buffered approach
can yield computational gains
2. when you need to process a large file that does not fit entirely
into [2]Paolo Prandoni, [3]LCAV, EPFL
In this notebook we will explore various ways to implement a system
that changes the voice of a speaker. Typical applications for these
techniques, strictly sorted by plausibility, include international spy
missions and prank telephone calls, and devices such as the one
displayed here can be easily bought on the web... or in a novelty shop.
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