comparison src/audacious/output.c @ 2538:719e0898ff3c trunk

[svn] - Patch to use libsamplerate for high-quality interpolated upsampling, similar to what foobar2000 can do. Patch by Teru KAMOGASHIRA.

2537:7aac1b5ef85d

#include "main.h"
#include "input.h"
#include "playback.h"

#include "playlist.h"

OutputPluginData op_data = {
    NULL,
    NULL
};

    OutputPlugin *op = get_current_output_plugin();

    return op->output_time();
}

gint
output_open_audio(AFormat fmt, gint rate, gint nch)
{
    gint ret;
    OutputPlugin *op;
    
    op = get_current_output_plugin();

    if (op == NULL)
        return -1;

void
output_close_audio(void)
{
    OutputPlugin *op = get_current_output_plugin();

    /* Do not close if there are still songs to play and the user has 
     * not requested a stop.  --nenolod
     */
    if (ip_data.stop == FALSE && 
    (playlist_get_position_nolock(playlist_get_active()) < playlist_get_length_nolock(playlist_get_active()) - 1))

    if (op == NULL)
        return 0;

    return op->buffer_playing();
}

/* called by input plugin when data is ready */
void
produce_audio(gint time,        /* position             */
              AFormat fmt,      /* output format        */

    guint myorder = G_BYTE_ORDER == G_LITTLE_ENDIAN ? FMT_S16_LE : FMT_S16_BE;
    int caneq = (fmt == FMT_S16_NE || fmt == myorder);
    OutputPlugin *op = get_current_output_plugin();
    int writeoffs;

    if (!caneq && cfg.equalizer_active) {    /* wrong byte order */
        byteswap(length, ptr);               /* so convert */
        ++swapped;
        ++caneq;
    }                                        /* can eq now, mark swapd */

2538:719e0898ff3c

#include "main.h"
#include "input.h"
#include "playback.h"

#include "playlist.h"
#include "libaudacious/configdb.h"
#ifdef USE_SRC
#include <samplerate.h>
#endif

OutputPluginData op_data = {
    NULL,
    NULL
};

    OutputPlugin *op = get_current_output_plugin();

    return op->output_time();
}

#ifdef USE_SRC
static SRC_STATE *src_state;
static SRC_DATA src_data;
static int overSamplingFs = 96000;
static int converter_type = SRC_SINC_BEST_QUALITY;
static int srcError = 0;
static void freeSRC()
{
  if(src_state != NULL)
    src_state = src_delete(src_state);
}
#endif

gint
output_open_audio(AFormat fmt, gint rate, gint nch)
{
    gint ret;
    OutputPlugin *op;
    
#ifdef USE_SRC
    ConfigDb *db;
    gboolean src_enabled;
    gint src_rate, src_type;
    db = bmp_cfg_db_open();
    
    if (bmp_cfg_db_get_bool(db, NULL, "enable_src", &src_enabled) == FALSE)
      src_enabled = FALSE;

    if (bmp_cfg_db_get_int(db, NULL, "src_rate", &src_rate) == FALSE)
      overSamplingFs = 48000;
    else
      overSamplingFs = src_rate;

    if (bmp_cfg_db_get_int(db, NULL, "src_type", &src_type) == FALSE)
      converter_type = SRC_SINC_BEST_QUALITY;
    else
      converter_type = src_type;
    
    bmp_cfg_db_close(db);
    
    freeSRC();
    
    if(src_enabled&&
       (fmt == FMT_S16_NE||(fmt == FMT_S16_LE && G_BYTE_ORDER == G_LITTLE_ENDIAN)||
	(fmt == FMT_S16_BE && G_BYTE_ORDER == G_BIG_ENDIAN)))
      {
	src_state = src_new(converter_type, nch, &srcError);
	if (src_state != NULL)
	  {
	    src_data.src_ratio = (float)overSamplingFs/(float)rate;
	    rate = overSamplingFs;
	  }
	else
	  fprintf(stderr, "src_new(): %s\n\n", src_strerror(srcError));
      }
#endif
    
    op = get_current_output_plugin();

    if (op == NULL)
        return -1;

void
output_close_audio(void)
{
    OutputPlugin *op = get_current_output_plugin();

#ifdef USE_SRC
    freeSRC();
#endif

    /* Do not close if there are still songs to play and the user has 
     * not requested a stop.  --nenolod
     */
    if (ip_data.stop == FALSE && 
    (playlist_get_position_nolock(playlist_get_active()) < playlist_get_length_nolock(playlist_get_active()) - 1))

    if (op == NULL)
        return 0;

    return op->buffer_playing();
}

#ifdef USE_SRC
static float *srcIn = NULL, *srcOut = NULL;
static short int *wOut = NULL;
static gboolean isSrcAlloc = FALSE;
#endif

/* called by input plugin when data is ready */
void
produce_audio(gint time,        /* position             */
              AFormat fmt,      /* output format        */

    guint myorder = G_BYTE_ORDER == G_LITTLE_ENDIAN ? FMT_S16_LE : FMT_S16_BE;
    int caneq = (fmt == FMT_S16_NE || fmt == myorder);
    OutputPlugin *op = get_current_output_plugin();
    int writeoffs;

#ifdef USE_SRC
    if(isSrcAlloc == TRUE)
      {
        free(srcIn);
        free(srcOut);
        free(wOut);
        isSrcAlloc = FALSE;
      }
    
    if(src_state != NULL&&length > 0)
      {
        int lrLength = length/2;
        int overLrLength = (int)floor(lrLength*(src_data.src_ratio+1));
        srcIn = (float*)malloc(sizeof(float)*lrLength);
        srcOut = (float*)malloc(sizeof(float)*overLrLength);
        wOut = (short int*)malloc(sizeof(short int)*overLrLength);
        src_short_to_float_array((short int*)ptr, srcIn, lrLength);
        isSrcAlloc = TRUE;
        src_data.data_in = srcIn;
        src_data.data_out = srcOut;
        src_data.end_of_input = 0;
        src_data.input_frames = lrLength/2;
        src_data.output_frames = overLrLength/2;
        if (srcError = src_process(src_state, &src_data))
          {
            fprintf(stderr, "src_process(): %s\n", src_strerror(srcError));
          }
        else
          {
            src_float_to_short_array(srcOut, wOut, src_data.output_frames_gen*2);
            ptr = wOut;
            length = src_data.output_frames_gen*4;
          }
      }
#endif

    if (!caneq && cfg.equalizer_active) {    /* wrong byte order */
        byteswap(length, ptr);               /* so convert */
        ++swapped;
        ++caneq;
    }                                        /* can eq now, mark swapd */