I've got audio samples; how to play them correctly with the SDL? -

i'm in process of writing simple frontend libretro cores, , i'm attacking audio part, hardest (i never ever dealt sound-related before, it's possible got totally wrong).

the way works, libretro cores regularly call function prototyped such, have implement myself:

typedef size_t (*retro_audio_sample_batch_t)(const int16_t *data, size_t count); 

the data parameter array of interlaced stereo data ({ left, right, left, ...}), , count number of samples (so array element count count * 2).

here these data:

• when libretro core sending me samples
  • i concatenate them audio buffer
• when sdl calling audio callback method,
  • i copy as requested , available audio buffer stream buffer
  • if not enough data available, fill rest of stream buffer 0's
  • if buffer has more data requested, remove played , keep rest

however, isn't right. despite being recognizable, sound garbled , full of noise. i'm not sure might missing : seems retroarch frontend (the official frontend libretro cores) doing sort of resampling, have no idea is. resampling required? can me issues come that?

here full code, partly reproduced below:

static int16_t * g_audio_samples = null; static size_t g_audio_sample_count = 0;  static void audio_write( int16_t const * samples, size_t count ) {     sdl_lockaudio( );      int16_t * previous_samples = g_audio_samples;     size_t previous_count = g_audio_sample_count;      g_audio_sample_count = previous_count + count;     g_audio_samples = calloc( g_audio_sample_count * 2, sizeof( int16_t ) );      ( size_t index = 0; index < previous_count; ++ index )         g_audio_samples[ index ] = previous_samples[ index ];      ( size_t index = 0; index < count; ++ index )         g_audio_samples[ previous_count + index ] = samples[ index ];      free( previous_samples );      sdl_unlockaudio( ); }  static void audio_callback( void * userdata, uint8 * stream, int length ) {     int16_t * available_samples = g_audio_samples;     size_t available_sample_count = g_audio_sample_count;      size_t requested_sample_count = length / 2;     size_t requested_byte_length = requested_sample_count * 2;      size_t providen_sample_count = requested_sample_count < available_sample_count ? requested_sample_count : available_sample_count;     size_t providen_byte_length = providen_sample_count * 2;      memcpy( stream, available_samples, providen_byte_length );     memset( stream + providen_byte_length, 0, requested_byte_length - providen_byte_length );      g_audio_samples = null;     g_audio_sample_count = 0;      if ( providen_sample_count < available_sample_count )         audio_write( available_samples + providen_sample_count, available_sample_count - providen_sample_count );      free( available_samples ); }  size_t bridge_virtjs_audio_push_sample_batch( int16_t const * samples, size_t count ) {     audio_write( samples, count );      return count; }