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ffmpeg tutorial - dranger.com
http://dranger.com/ffmpeg/tutorial03.html
The "S" in "S16SYS" stands for "signed", the 16 says that each sample is 16 bits long, and "SYS" means that the endian-order will depend on the system you are on. This is the format that avcodec_decode_audio2 will give us the audio in. channels: Number of audio channels. silence: This is the value that indicated silence. Since the audio is signed, 0 is of course the usual value.
SDL_mixer 1.2.10: Mix_OpenAudio - Simple DirectMedia …
https://www.libsdl.org/projects/SDL_mixer/docs/SDL_mixer_11.html
AUDIO_U16 same as AUDIO_U16LSB (for backwards compatability probably) AUDIO_S16 same as AUDIO_S16LSB (for backwards compatability probably) AUDIO_U16SYS Unsigned 16-bit samples, in system byte order AUDIO_S16SYS Signed 16-bit samples, in system byte order MIX_DEFAULT_FORMAT is the same as AUDIO_S16SYS. Returns: 0 on success, -1 on errors
SDL mixer:Tutorials:Playing a WAV Sound File | ModDB Wiki ...
https://moddb.fandom.com/wiki/SDL_mixer:Tutorials:Playing_a_WAV_Sound_File
If you're using 8-bit WAV file samples, then use the constant AUDIO_S8. If you are using 16-bit WAV file samples, then use the constant AUDIO_S16SYS. AUDIO_S16SYS will automatically match the user's system's byte order, so you don't have to worry about the endianness of the target system! Now, if you're not sure whether or not your samples are 8-bit or 16-bit, simply try …
audio - Simple sound wave generator with SDL in c++ ...
https://stackoverflow.com/questions/10110905/simple-sound-wave-generator-with-sdl-in-c
#include <math.h> #include <SDL.h> #include <SDL_audio.h> const int AMPLITUDE = 28000; const int SAMPLE_RATE = 44100; void audio_callback(void *user_data, Uint8 *raw_buffer, int bytes) { Sint16 *buffer = (Sint16*)raw_buffer; int length = bytes / 2; // 2 bytes per sample for AUDIO_S16SYS int &sample_nr(*(int*)user_data); for(int i = 0; i < length; i++, …
ffmpeg tutorial 05 with SDL2 and audio resampling fix · GitHub
https://gist.github.com/d7d935e87a44410966ed8cad066953bc
wanted_spec. format = AUDIO_S16SYS; wanted_spec. channels = codecCtx-> channels; wanted_spec. silence = 0; wanted_spec. samples = SDL_AUDIO_BUFFER_SIZE; wanted_spec. …
man SDL::Audio (3): SDL Bindings for Audio
http://manpages.org/sdlaudio/3
use SDL; use SDL::Audio; use SDL::AudioSpec; SDL::init(SDL_INIT_AUDIO); # Converting some WAV data to hardware format my $desired = SDL::AudioSpec->new(); my $obtained = SDL::AudioSpec->new(); # Set desired format $desired->freq(22050); $desired->channels(1); $desired->format(AUDIO_S16); $desired->samples(8192); # Open the audio device if( …
GitHub - realnc/SDL_audiolib: An audio decoding ...
https://github.com/realnc/SDL_audiolib
Set the output sample rate to // 44.1kHz, the audio format to 16-bit signed, use 2 output channels // (stereo), and an 8kB output buffer. if (Aulib::init (44100, AUDIO_S16SYS, 2, 8192) != 0) { std::cerr << " Couldn't initialize audio: " << SDL_GetError << ' \n '; return EXIT_FAILURE; } // Create an audio stream that will play our Vorbis file using a Vorbis // decoder and a Speex …
SDL_AudioSpec - SDL Wiki
https://wiki.libsdl.org/SDL_AudioSpec
The Nyquist Theorem states that the audio sampling frequency must be exactly twice the highest frequency represented in the audio. Humans can hear up to slightly under 20kHz, declining to 16kHz or lower as we age.
ffmpeg - How to make SDL play FLTP samples (AAC)? - Stack ...
https://stackoverflow.com/questions/32392571/how-to-make-sdl-play-fltp-samples-aac
The tutorials showed this way, SDL_AudioSpec wanted_spec, spec; wanted_spec.freq = aCodecCtx->sample_rate; wanted_spec.format = AUDIO_S16SYS; wanted_spec.channels = aCodecCtx->channels;
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