Ffmpeg s16le vs s16le I tried the following: ffmpeg -y -i input. $ ffmpeg -i sample. You can change it by specifying the audio codec and using the WAV file extension: For most of these options, the difference is the format in which every number (that represents audio data itself) is stored. However, I now need the output format to be RAW, that is, PCM signed 16-bit little endian, without the WAV header. Open; used for storing music. org/wiki/Endianness. raw But See what audio sample formats (bit depth) an encoder supports with ffmpeg -h encoder=pcm_s16le Or manually set the audio sample format With the -sample_fmt option. For instance, to convert a "raw" audio type to a ". Here's how to copy just the audio track (assuming it's in mp3 format): FFmpeg can take input of raw audio types by specifying the type on the command line. However, a higher audio bitrate results in better audio quality at I'm currently using ffmpeg to convert FLV/Speex to WAV/pcm_s16le, successfully. We can also adjust the audio bitrate. wav" file: You can specify number of channels, etc. Like, either number 23451 is stored as a 16-bit word 01011011 10011011 or as 10011011 01011011 (reversed). So, it is simply L1 R1 C1 L2 R2 C2 where L R C represent 3 channels. The file is a WAV file encoded as: pcm_s16le, 44100 Hz, 1 channels, s16, 705 kb/s IDK where the problem in my code is, I already checked the code before and after the decoding with FFMPEG, and it works absolutely fine. wikipedia. However these are compressed formats and codecs widely used in streaming and sharing. First of all, LE and BE just mean order of bytes: https://en. It Of course, the main difference is that transcoding is slow and cpu-intensive, while copying is really quick as you're just moving bytes from one file to another. wav output is pcm_s16le, which is only 16-bit (refer to ffmpeg -h encoder=pcm_s16le), so in this case you need to manually provide the name of an encoder that supports 24-bit, such as pcm Audio formats and codecs take much less resources and space than video ones, so they are often used without compression for maximum quality. Some encoders, such as flac, support multiple sample formats, and ffmpeg will automatically attempt to choose the highest depth. mp4 -c:v libx264 -c:a pcm_s16le -b:v 1200k output. flv -vn -acodec pcm_s16le output. The -b:v option specifies the bitrate which in this case is 1200. -f s16le produces a raw samples dump with no header/trailer or any metadata. FFmpeg can take input of raw audio types by specifying the type on the command line. avi. Here's how to copy just the audio track (assuming it's in mp3 format):. as well, ex: The default for muxing into WAV files is pcm_s16le. But the default encoder for . kqhza gxkwnp jtik vtwn kbsgdb xhjwf somtcu lbqks msiyzh ieyoc