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audio types – FFmpeg
https://trac.ffmpeg.org/wiki/audio%20types#:~:text=Raw%20audio%20in%20FFmpeg%20can%20take%20several%20different,sample%20formats%20by%20inspecting%20the%20ffmpeg%20-formats%20output%3A
FFmpeg: Audio sample formats
https://www.ffmpeg.org/doxygen/3.1/group__lavu__sampfmts.html
The data layout as used in av_samples_fill_arrays()and elsewhere in FFmpeg (such as AVFramein libavcodec) is as follows: For planar sample formats, each audio channel is in a separate data plane, and linesize is the buffer size, in bytes, for a single plane. All data planes must be the same size.
audio types – FFmpeg
https://trac.ffmpeg.org/wiki/audio%20types
These represent all the built-in raw audio sample formats. Reading and Writing Raw Audio. FFmpeg can take input of raw audio types by specifying the type on the command line. For instance, to convert a "raw" audio type to a ".wav" file: ffmpeg -f s32le input_filename.raw output.wav You can specify number of channels, etc. as well, ex: ffmpeg -f u16le -ar 44100 -ac …
ffmpeg: Audio sample formats - doxygen documentation ...
https://fossies.org/dox/ffmpeg-5.0/group__lavu__sampfmts.html
Get the planar alternative form of the given sample format. More... char * av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt) Generate a string corresponding to the sample format with sample_fmt, or a header if sample_fmt is negative. More... int av_get_bytes_per_sample (enum AVSampleFormat sample_fmt)
FFmpeg Formats Documentation
https://ffmpeg.org/ffmpeg-formats.html
ffmpeg -i sample.mpeg \ -f hls -hls_time 3 -hls_list_size 5 \ -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \ -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
video - audio sample format s16p, ffmpeg or audio codec ...
https://stackoverflow.com/questions/35226255/audio-sample-format-s16p-ffmpeg-or-audio-codec-bug
Stream #0:1 [0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 256 kb/s Stream #0:2 [0x1c1]: Audio: mp2, 48000 Hz, stereo, s16p, 256 kb/s. With the same video, its sample format changes to s16p. I implemented a simple video player which uses ffmpeg.
How to use FFMpeg to do Simple Audio Conversion
https://www.howtoforge.com/tutorial/ffmpeg-audio-conversion/
ffmpeg -i filename.mp3 newfilename.wav newfilename.ogg newfilename.mp4. This will result in converting 3 output audio files (wav,ogg,mp4) from one mp3 file. Alternatively, you can set the desired codec using the -c command like this: ffmpeg -i filename.mp4 c:a libopus newfilename.ogg
audio streaming - How do I change the bit depth in FFmpeg ...
https://superuser.com/questions/547929/how-do-i-change-the-bit-depth-in-ffmpeg
By default, the FFmpeg FLAC encoder takes the bit depth of the original. The bit depth can be changed with the sample_fmt option, e.g. ffmpeg -i … -c:a flac -sample_fmt s16 output.flac Note that not all formats are supported by every encoder. For a list of all supported sample formats, run: ffmpeg -sample_fmts
music - Changing sample format and bit depth on audio ...
https://askubuntu.com/questions/1306318/changing-sample-format-and-bit-depth-on-audio-files-with-ffmpeg
Or manually set the audio sample format. With the -sample_fmt option. ffmpeg -i input.wav -sample_fmt s16 -ar 44100 output.wav See a list of audio sample formats (bit depth) with ffmpeg -sample_fmts; Or use the aformat filter ffmpeg -i input.wav -af "aformat=sample_fmts=s16:sample_rates=44100" output.wav
ffmpeg Documentation
https://ffmpeg.org/ffmpeg.html
-sample_fmt[:stream_specifier] sample_fmt (output,per-stream) Set the audio sample format. Use -sample_fmts to get a list of supported sample formats. -af filtergraph (output) Create the filtergraph specified by filtergraph and use it to filter the stream. This is an alias for -filter:a, see the -filter option. 5.8 Advanced Audio options
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