linux_audio.h (4297B)
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 | #ifndef _LINUX_AUDIO_H_ #define _LINUX_AUDIO_H_ #include <stdio.h> #include <stdarg.h> #include <dlfcn.h> #include <alsa/asoundlib.h> #include "intrinsics.h" #include "log.h" struct AudioOutputDevice { snd_pcm_t * pcm; }; void audio_output_open( AudioOutputDevice * device ); void audio_output_close( AudioOutputDevice * device ); void audio_output_write( AudioOutputDevice * device, s16 * samples, size_t num_samples ); #endif // _LINUX_AUDIO_H_ #ifdef PLATFORM_AUDIO_IMPLEMENTATION // copied from alsa headers static int ( *_snd_lib_error_set_handler )( snd_lib_error_handler_t handler ) = NULL; static const char * ( *_snd_strerror )( int errnum ) = NULL; static int ( *_snd_pcm_open )( snd_pcm_t ** pcm, const char * name, snd_pcm_stream_t stream, int mode ) = NULL; static int ( *_snd_pcm_set_params )( snd_pcm_t * pcm, snd_pcm_format_t format, snd_pcm_access_t access, unsigned int channels, unsigned int rate, int soft_resample, unsigned int latency ) = NULL; static snd_pcm_sframes_t ( *_snd_pcm_writen )( snd_pcm_t * pcm, void ** bufs, snd_pcm_uframes_t size ) = NULL; static int ( *_snd_pcm_recover )( snd_pcm_t * pcm, int err, int silent ) = NULL; static int ( *_snd_pcm_close )( snd_pcm_t * pcm ) = NULL; static void audio_error_handler( const char * file, int line, const char * function, int err, const char * fmt, ... ) { printf( "error in %s at %s:%d\n", file, function, line ); va_list args; va_start( args, fmt ); vprintf( fmt, args ); va_end( args ); printf( "\n" ); } void audio_output_open( AudioOutputDevice * device ) { // TODO: make an audio_output_init_lib function to do this + dlload static bool audio_inited = false; if( !audio_inited ) { void * alsa_lib = dlopen( "libasound.so", RTLD_NOW ); if( alsa_lib == NULL ) { FATAL( "Couldn't open libasound.so: %s", dlerror() ); } _snd_lib_error_set_handler = ( int ( * )( snd_lib_error_handler_t ) ) dlsym( alsa_lib, "snd_lib_error_set_handler" ); _snd_strerror = ( const char * ( * )( int ) ) dlsym( alsa_lib, "snd_strerror" ); _snd_pcm_open = ( int ( * ) ( snd_pcm_t **, const char *, snd_pcm_stream_t, int ) ) dlsym( alsa_lib, "snd_pcm_open" ); _snd_pcm_set_params = ( int ( * ) ( snd_pcm_t *, snd_pcm_format_t, snd_pcm_access_t, unsigned int, unsigned int, int, unsigned int ) ) dlsym( alsa_lib, "snd_pcm_set_params" ); _snd_pcm_writen = ( snd_pcm_sframes_t ( * ) ( snd_pcm_t *, void **, snd_pcm_uframes_t ) ) dlsym( alsa_lib, "snd_pcm_writen" ); _snd_pcm_recover = ( int ( * )( snd_pcm_t *, int, int ) ) dlsym( alsa_lib, "snd_pcm_recover" ); _snd_pcm_close = ( int ( * ) ( snd_pcm_t * ) ) dlsym( alsa_lib, "snd_pcm_close" ); if( _snd_lib_error_set_handler == NULL || _snd_strerror == NULL || _snd_pcm_open == NULL || _snd_pcm_set_params == NULL || _snd_pcm_writen == NULL || _snd_pcm_recover == NULL || _snd_pcm_close == NULL ) { FATAL( "Couldn't load ALSA functions" ); } _snd_lib_error_set_handler( audio_error_handler ); audio_inited = true; } const int channels = 2; const int sample_rate = 44100; const int ms = 1000; const int latency = 10 * ms; int err_open = _snd_pcm_open( &device->pcm, "default", SND_PCM_STREAM_PLAYBACK, 0 ); if( err_open != 0 ) { // TODO: maybe don't kill the program FATAL( "Couldn't open sound output: %s", _snd_strerror( err_open ) ); } int err_params = _snd_pcm_set_params( device->pcm, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_NONINTERLEAVED, channels, sample_rate, 1, latency ); if( err_params != 0 ) { // TODO: maybe don't kill the program FATAL( "Couldn't configure sound output: %s", _snd_strerror( err_params ) ); } } void audio_output_close( AudioOutputDevice * device ) { int err = _snd_pcm_close( device->pcm ); if( err != 0 ) { // TODO: maybe don't kill the program FATAL( "Couldn't close sound output: %s", _snd_strerror( err ) ); } } void audio_output_write( AudioOutputDevice * device, s16 * samples, size_t num_samples ) { for( u32 i = 0; i < num_samples; ) { s16 * channels[ 2 ] = { samples + i, samples + i }; snd_pcm_sframes_t written = _snd_pcm_writen( device->pcm, ( void ** ) channels, num_samples - i ); if( written <= 0 ) { _snd_pcm_recover( device->pcm, written, 1 ); } else { i += written; } } } #endif |