- #Ffmpeg pcm to wav install#
- #Ffmpeg pcm to wav update#
- #Ffmpeg pcm to wav Patch#
- #Ffmpeg pcm to wav software#
writewav.c - Create a wav file by piping raw samples to ffmpeg The sound is just a loud 1 kHz sine wave. The following example creates a 1-second long WAV audio file (PCM, 16-bit signed integer samples, 44.1 kHz sampling frequency). Ok, time for some practical examples… Writing a WAV audio file (or MP3, FLAC, or whatever format you like)
#Ffmpeg pcm to wav install#
I’m working on Xubuntu Linux, so to install FFmpeg I just did this:
#Ffmpeg pcm to wav software#
So far, I haven’t needed to read or write any file format that it couldn’t handle, although in some cases it did involve some googling to figure out the required command line arguments.įFmpeg is free software and cross-platform, so you should be able to install it on Windows and Mac without too much difficulty, but I haven’t actually tried that myself. It may seem like a bit of a hack, but it’s surprisingly effective and, provided that you can figure out the correct FFmpeg command line, extremely adaptable.įFmpeg is described on its project web page as “ complete, cross-platform solution to record, convert and stream audio and video.” That’s a pretty accurate description.
The basic idea is to launch FFmpeg via a pipe, which then converts raw samples to the required format (for writing) or decodes the file into raw samples (for reading). The article describes using FFmpeg to read and write video frames in Python and there’s a link to a second article showing the same thing for audio.
#Ffmpeg pcm to wav update#
* update to new upstream.C is my favourite programming language and the one I use most often. remaining changes:įfmpeg (4:0.6~svn20100505-1) experimental urgency=low * Many upstream changes, see upstream Changelog for details * Fixup lintian overrides for new upstream snapshot don't build against libfaad, libdirac and libopenjpeg (all in universe)įfmpeg (4:0.6~ svn20100505- 1) experimental urgency=low This bug was fixed in the package ffmpeg - 4:0.6~svn201005 05-1ubuntu1įfmpeg (4:0.6~ svn20100505- 1ubuntu1) maverick urgency=low $ hexdump -x diatonis_dts_secret-universe.wav -n 2 -s 0x14 Using "ffplay -f dts file.wav" or "ffmpeg -f dts -i file.wav output.ogg" skips the codec detection, and works properly. htmlĪs an aside, there's a workaround which I had thought didn't work because I was specifying the options in the wrong order.
#Ffmpeg pcm to wav Patch#
There was a patch proposed, but it wasn't adopted at the time. I would consider this a problem with the file, but according to the following thread on ffmpeg-devel, it's standard for programs writing DTS audio to do so with a broken header to maintain compatibility with broken CD burning programs. The file now plays properly in ffplay, and ffmpeg can decode it properly.
Using hexedit to change the bytes at offset 0x14 from 01 00 to 01 20 (0x2001 in little-endian format): $ hexdump -x diatonis_ dts_secret- universe. This problem arises because the relevant file's "fmt " chunk specifies type 0x01 (it's in little-endian format, two bytes at offset 0x14), which is uncompressed PCM data. On further investigation it's already open upstream. PATH=/ home/username/ bin:/usr/ local/sbin: /usr/local/ bin:/usr/ sbin:/usr/ bin:/sbin: /bin:/usr/ games
What happened: ffmpeg decoded/played it as a PCM WAV file, resulting in white noise. What I expected: I expected ffmpeg to decode the sound file as a DTS file, and ffplay to play it as one. This version of ffmpeg supposedly does support DTS audio: However, ffplay and ffmpeg assume that they're PCM WAV files, and output white noise. (They're quite large, which is why I'm not attaching one.) I can play them in mplayer, which uses libdca to decode them, without specifying any command-line options or forcing a particular codec. Consider any DTS WAV file from com/downloads_ dts_ac3.