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#include <SDL/SDL.h>
#include <samplerate.h>
#include "psdl.h"
#include "codec.h"
#include "playlist.h"
static int sdl_player_numsamples = 2<<25;
static int sdl_player_freq;
static char *sdl_buffer;
static float *sdl_fbuffer;
static float *sdl_srcbuffer;
static SDL_AudioSpec spec;
static void psdl_callback (void *userdata, Uint8 *stream, int len);
static codec_t *codec;
static SRC_STATE *src;
static SRC_DATA srcdata;
static int codecleft;
static float sdl_volume = 1;
int
psdl_init (void) {
SDL_AudioSpec obt;
int formats[] = { AUDIO_S16, -1 };
int freqs[] = { 48000, 44100, -1 };
const char *fmtnames[] = { "16 bit signed integer" };
int fmt, frq;
int success = 0;
fprintf (stderr, "sdl_player_init\n");
for (fmt = 0; formats[fmt] != -1; fmt++) {
for (frq = 0; freqs[frq] != -1; frq++) {
fprintf(stderr, "SDL: trying %s @ %dHz\n", fmtnames[fmt], freqs[frq]);
spec.freq = freqs[frq];
spec.format = formats[fmt];
spec.channels = 2;
spec.samples = sdl_player_numsamples;
spec.callback = psdl_callback;
if (SDL_OpenAudio(&spec, &obt) < 0) {
fprintf(stderr, "SDL: couldn't open audio: %s\n", SDL_GetError());
}
else {
success = 1;
break;
}
}
}
if (!success) {
return -1;
}
sdl_player_numsamples = obt.samples; //numsamples;
// source samplerate may be up to 96KHz, so need bigger buffers
sdl_buffer = malloc (sdl_player_numsamples * sizeof (uint16_t) * 2 * 2);
sdl_fbuffer = malloc (sdl_player_numsamples * sizeof (float) * 2 * 2);
sdl_srcbuffer = malloc (sdl_player_numsamples * sizeof (float) * 2);
sdl_player_freq = obt.freq;
fprintf (stderr, "SDL: got %d frame size (requested %d), %dHz\n", obt.samples, sdl_player_numsamples, sdl_player_freq);
// src = src_new (SRC_SINC_BEST_QUALITY, 2, NULL);
src = src_new (SRC_LINEAR, 2, NULL);
codecleft = 0;
return 0;
}
void
psdl_free (void) {
SDL_CloseAudio ();
if (sdl_buffer) {
free (sdl_buffer);
sdl_buffer = NULL;
}
src_delete (src);
src = NULL;
}
int
psdl_play (struct playItem_s *it) {
printf ("psdl_play\n");
if (!it) {
return -1;
}
codec = it->codec;
if (codec->init (it->fname)) {
return -1;
}
SDL_PauseAudio (0);
return 0;
}
int
psdl_stop (void) {
SDL_PauseAudio (1);
if (codec) {
codec->free ();
codec = NULL;
}
}
int
psdl_pause (void) {
SDL_PauseAudio (1);
}
void
psdl_set_volume (float vol) {
sdl_volume = vol;
}
static void
psdl_callback (void* userdata, Uint8 *stream, int len) {
if (!codec) {
memset (stream, 0, len);
}
else if (codec->info.samplesPerSecond == sdl_player_freq) {
codec->read (sdl_buffer, len);
}
else {
int nsamples = len/4;
// convert to codec samplerate
// add 5 extra samples for SRC
nsamples = nsamples * codec->info.samplesPerSecond / sdl_player_freq + 5;
// read data at source samplerate (with some room for SRC)
codec->read (sdl_buffer, (nsamples - codecleft) * 4);
// convert to float, and apply soft volume
int i;
float *fbuffer = sdl_fbuffer + codecleft*2;
for (i = 0; i < (nsamples - codecleft) * 2; i++) {
fbuffer[i] = ((int16_t *)sdl_buffer)[i]/32767.f;
sdl_fbuffer[i+codecleft*2] *= sdl_volume;
}
// convert samplerate
srcdata.data_in = sdl_fbuffer;
srcdata.data_out = sdl_srcbuffer;
srcdata.input_frames = nsamples;
srcdata.output_frames = len/4;
srcdata.src_ratio = (double)sdl_player_freq/codec->info.samplesPerSecond;
srcdata.end_of_input = 0;
src_process (src, &srcdata);
// convert back to s16 format
for (i = 0; i < len/2; i++) {
((int16_t*)stream)[i] = (int16_t)(sdl_srcbuffer[i]*32767.f);
}
// calculate how many unused input samples left
codecleft = nsamples - srcdata.input_frames_used;
// copy spare samples for next update
memmove (sdl_fbuffer, &sdl_fbuffer[srcdata.input_frames_used*2], codecleft * 8);
}
}
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