diff options
| -rw-r--r-- | Jamfile | 2 | ||||
| -rw-r--r-- | cvorbis.c | 5 | ||||
| -rw-r--r-- | cwav.c | 4 | ||||
| -rw-r--r-- | psdl.c | 72 |
4 files changed, 68 insertions, 15 deletions
@@ -5,5 +5,5 @@ CCFLAGS += -std=c99 ; Main deadbeef : cmod.c codec.c cvorbis.c cwav.c playlist.c psdl.c main.c ; -LINKLIBS on deadbeef = -lmikmod -lm -lvorbis -logg -lvorbisfile -lmikmod -lSDL ; +LINKLIBS on deadbeef = -lmikmod -lm -lvorbis -logg -lvorbisfile -lmikmod -lSDL -lsamplerate ; @@ -42,6 +42,7 @@ cvorbis_read (char *bytes, int size) { if (!file) return -1; + printf ("vorbis read %d bytes!\n", size); for (;;) { // read ogg @@ -60,6 +61,10 @@ cvorbis_read (char *bytes, int size) size -= ret; bytes += ret; } + else if (ret > size) { + printf ("read more than requested!\n"); + break; + } else { break; } @@ -31,7 +31,7 @@ int cwav_init (const char *fname) { wavHeader_t header; // read WAV header - if (fread (&header, sizeof (header), 1, file) != sizeof (header)) + if (fread (&header, sizeof (header), 1, file) != 1) { printf ("WARNING: WAV header not found in %s\n", fname); fclose (file); @@ -61,7 +61,7 @@ cwav_read (char *bytes, int size) { int bytesread = 0; int first = 1; for (;;) { - int bytesread = fread (bytes, size, 1, file); + int bytesread = fread (bytes, 1, size, file); if (bytesread == size) { break; } @@ -1,15 +1,24 @@ #include <SDL/SDL.h> +#include <samplerate.h> #include "psdl.h" #include "codec.h" #include "playlist.h" -static int sdl_player_numsamples = 2<<16; +static int sdl_player_numsamples = 2<<25; static int sdl_player_freq; -static float *sdl_buffer[2]; +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; @@ -18,8 +27,6 @@ psdl_init (void) { const char *fmtnames[] = { "16 bit signed integer" }; int fmt, frq; int success = 0; - sdl_buffer[0] = NULL; - sdl_buffer[1] = NULL; fprintf (stderr, "sdl_player_init\n"); for (fmt = 0; formats[fmt] != -1; fmt++) { for (frq = 0; freqs[frq] != -1; frq++) { @@ -42,26 +49,34 @@ psdl_init (void) { 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 * 10); + sdl_fbuffer = malloc (sdl_player_numsamples * sizeof (float) * 2 * 10); + sdl_srcbuffer = malloc (sdl_player_numsamples * sizeof (float) * 10); 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[0]) { - free (sdl_buffer[0]); - sdl_buffer[0] = NULL; - } - if (sdl_buffer[1]) { - free (sdl_buffer[1]); - sdl_buffer[1] = NULL; + 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; } @@ -90,7 +105,40 @@ psdl_callback (void* userdata, Uint8 *stream, int len) { memset (stream, 0, len); } else { - codec->read (stream, len); + int nsamples = len/4+3000; + // convert to codec samplerate +// nsamples = nsamples * codec->info.samplesPerSecond / sdl_player_freq + 100; + // add how many additional samples we need for src + printf ("reading nsamples=%d (codecleft=%d)\n", nsamples-codecleft, codecleft); + // 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); + ((int16_t*)stream)[i] = (int16_t)(sdl_fbuffer[i]*32767.f); + } + // calculate how many unused input samples left +// codecleft = nsamples - srcdata.input_frames_used; + codecleft = nsamples - len/4; +// printf ("codecleft: %d (used %d), in: %d, out: %d, outreq: %d, ratio: %f\n", codecleft, srcdata.input_frames_used, nsamples, srcdata.output_frames_gen,srcdata.output_frames, srcdata.src_ratio); + // copy spare samples for next update +// memmove (sdl_fbuffer, &sdl_fbuffer[srcdata.input_frames_used*2], codecleft * 4); + memmove (sdl_fbuffer, sdl_fbuffer+len/2, codecleft * 4); } } |