view src/audacious/output.c @ 3555:a73951b8cd9f trunk

effect processing -> flow manager API / attached to postproc_flow.

/*  Audacious - Cross-platform multimedia player
 *  Copyright (C) 2005-2007  Audacious team
 *
 *  Based on BMP:
 *  Copyright (C) 2003-2004  BMP development team.
 *
 *  Based on XMMS:
 *  Copyright (C) 1998-2003  XMMS development team.
 *
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; under version 3 of the License.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program.  If not, see <http://www.gnu.org/licenses>.
 *
 *  The Audacious team does not consider modular code linking to
 *  Audacious or using our public API to be a derived work.
 */

#ifdef HAVE_CONFIG_H
#  include "config.h"
#endif

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

#include "playlist.h"
#include "configdb.h"

#include "flow.h"

#include "effect.h"
#include "volumecontrol.h"

#include <math.h>

#ifdef USE_SRC
#include <samplerate.h>
#endif

OutputPluginData op_data = {
    NULL,
    NULL
};

OutputPluginState op_state = {
    0,
    0,
    0
};

OutputPlugin psuedo_output_plugin = {
    .description = "XMMS reverse compatibility output plugin",
    .get_volume = output_get_volume,
    .set_volume = output_set_volume,

    .open_audio = output_open_audio,
    .write_audio = output_write_audio,
    .close_audio = output_close_audio,

    .flush = output_flush,
    .pause = output_pause,

    .buffer_free = output_buffer_free,
    .buffer_playing = output_buffer_playing,
    .output_time = get_output_time,
    .written_time = get_written_time,
};

OutputPlugin *
get_current_output_plugin(void)
{
    return op_data.current_output_plugin;
}

void
set_current_output_plugin(gint i)
{
    gboolean playing;
    OutputPlugin *op = get_current_output_plugin();

    GList *node = g_list_nth(op_data.output_list, i);
    if (!node) {
        op_data.current_output_plugin = NULL;
        return;
    }

    op_data.current_output_plugin = node->data;

    playing = playback_get_playing();

    if (playing == TRUE)
    {
        guint time, pos;
        PlaylistEntry *entry;

        /* don't stop yet, get the seek time and playlist position first */
        pos = playlist_get_position(playlist_get_active());
        time = op->output_time();

        /* reset the audio system */
        mainwin_stop_pushed();
        op->close_audio();

        g_usleep(300000);

        /* wait for the playback thread to come online */
        while (playback_get_playing())
            g_message("waiting for audio system shutdown...");

        /* wait for the playback thread to come online */
        playlist_set_position(playlist_get_active(), pos);
        entry = playlist_get_entry_to_play(playlist_get_active());
        playback_play_file(entry);

        while (!playback_get_playing())
        {
            gtk_main_iteration();
                g_message("waiting for audio system startup...");
        }

        /* and signal a reseek */
        if (playlist_get_current_length(playlist_get_active()) > -1 &&
            time <= (playlist_get_current_length(playlist_get_active())))
            playback_seek(time / 1000);
    }
}

GList *
get_output_list(void)
{
    return op_data.output_list;
}

void
output_about(gint i)
{
    OutputPlugin *out = g_list_nth(op_data.output_list, i)->data;
    if (out && out->about)
        out->about();
}

void
output_configure(gint i)
{
    OutputPlugin *out = g_list_nth(op_data.output_list, i)->data;
    if (out && out->configure)
        out->configure();
}

void
output_get_volume(gint * l, gint * r)
{
    *l = *r = -1;

    if (!op_data.current_output_plugin)
        return;

    if (!op_data.current_output_plugin->get_volume)
        return;

    if (cfg.software_volume_control)
        volumecontrol_get_volume_state(l, r);
    else
        op_data.current_output_plugin->get_volume(l, r);
}

void
output_set_volume(gint l, gint r)
{
    if (!op_data.current_output_plugin)
        return;

    if (!op_data.current_output_plugin->set_volume)
        return;

    if (cfg.software_volume_control)
        volumecontrol_set_volume_state(l, r);
    else
        op_data.current_output_plugin->set_volume(l, r);
}

void
output_set_eq(gboolean active, gfloat pre, gfloat * bands)
{
    int i;
    preamp[0] = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre;
    preamp[1] = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre;

    for (i = 0; i < 10; ++i)
    {
        set_gain(i, 0, 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]);
        set_gain(i, 1, 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]);
    }
}

/* this should be in BYTES, NOT gint16s */
static void
byteswap(size_t size,
         gint16 * buf)
{
    gint16 *it;
    size &= ~1;                  /* must be multiple of 2  */
    for (it = buf; it < buf + size / 2; ++it)
        *(guint16 *) it = GUINT16_SWAP_LE_BE(*(guint16 *) it);
}

/* called by input plugin to peek at the output plugin's write progress */
gint
get_written_time(void)
{
    OutputPlugin *op = get_current_output_plugin();

    return op->written_time();
}

/* called by input plugin to peek at the output plugin's output progress */
gint
get_output_time(void)
{
    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 float *srcIn = NULL, *srcOut = NULL;
static short int *wOut = NULL;
static int lengthOfSrcIn = 0;
static int lengthOfSrcOut = 0;

static void freeSRC()
{
  if(src_state != NULL)
    src_state = src_delete(src_state);
  free(srcIn);
  free(srcOut);
  free(wOut);
  srcIn = NULL;
  srcOut = NULL;
  wOut = NULL;
  lengthOfSrcIn = 0;
  lengthOfSrcOut = 0;
}

#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;

    /* don't resample if sampling rates are the same --nenolod */
    if (rate == overSamplingFs)
      src_enabled = FALSE;

    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;

    /* Is our output port already open? */
    if ((op_state.rate != 0 && op_state.nch != 0) &&
        (op_state.rate == rate && op_state.nch == nch && op_state.fmt == fmt))
    {
        /* Yes, and it's the correct sampling rate. Reset the counter and go. */
        op->flush(0);
        return 1;
    }
    else if (op_state.rate != 0 && op_state.nch != 0)
        op->close_audio();

    ret = op->open_audio(fmt, rate, nch);

    if (ret == 1)            /* Success? */
    {
        op_state.fmt = fmt;
        op_state.rate = rate;
        op_state.nch = nch;
    }

    return ret;
}

void
output_write_audio(gpointer ptr, gint length)
{
    OutputPlugin *op = get_current_output_plugin();

    /* Sanity check. */
    if (op == NULL)
        return;

    op->write_audio(ptr, length);
}

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(playlist_get_active()) - 1))
        return;

    /* Sanity check. */
    if (op == NULL)
        return;

    op->close_audio();

    /* Reset the op_state. */
    op_state.fmt = op_state.rate = op_state.nch = 0;
}

void
output_flush(gint time)
{
    OutputPlugin *op = get_current_output_plugin();

    if (op == NULL)
        return;

    op->flush(time);
}

void
output_pause(gshort paused)
{
    OutputPlugin *op = get_current_output_plugin();

    if (op == NULL)
        return;

    op->pause(paused);
}

gint
output_buffer_free(void)
{
    OutputPlugin *op = get_current_output_plugin();

    if (op == NULL)
        return 0;

    return op->buffer_free();
}

gint
output_buffer_playing(void)
{
    OutputPlugin *op = get_current_output_plugin();

    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        */
              gint nch,         /* channels             */
              gint length,      /* length of sample     */
              gpointer ptr,     /* data                 */
              int *going        /* 0 when time to stop  */
              )
{
    static Flow *postproc_flow = NULL;
    static int init = 0;
    int swapped = 0;
    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;
    AFormat new_format;
    gint new_rate;
    gint new_nch;

    if (postproc_flow == NULL)
    {
        postproc_flow = flow_new();
        flow_link_element(postproc_flow, effect_flow);
        flow_link_element(postproc_flow, volumecontrol_flow);
    }

#ifdef USE_SRC
    if(src_state != NULL&&length > 0)
      {
        int lrLength = length/2;
        int overLrLength = (int)floor(lrLength*(src_data.src_ratio+1));
	if(lengthOfSrcIn < lrLength)
	  {
	    lengthOfSrcIn = lrLength;
	    free(srcIn);
	    srcIn = (float*)malloc(sizeof(float)*lrLength);
	  }
	if(lengthOfSrcOut < overLrLength)
	  {
	    lengthOfSrcOut = overLrLength;
	    free(srcOut);
	    free(wOut);
	    srcOut = (float*)malloc(sizeof(float)*overLrLength);
	    wOut = (short int*)malloc(sizeof(short int)*overLrLength);
	  }
        src_short_to_float_array((short int*)ptr, srcIn, lrLength);
        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)) > 0)
          {
            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 */
    else if (caneq && !cfg.equalizer_active) /* right order but no eq */
        caneq = 0;                           /* so don't eq */

    if (caneq) {                /* if eq enab */
        if (!init) {            /* if first run */
            init_iir();         /* then init eq */
            ++init;
        }

        iir(&ptr, length, nch);

        if (swapped)               /* if was swapped */
            byteswap(length, ptr); /* swap back for output */
    }                           

    /* do vis plugin(s) */
    input_add_vis_pcm(time, fmt, nch, length, ptr);

    /* do effect plugin(s) */
    new_format = op_state.fmt;
    new_rate = op_state.rate;
    new_nch = op_state.nch;

    effect_do_query_format(&new_format, &new_rate, &new_nch);

    if (new_format != op_state.fmt ||
        new_rate != op_state.rate ||
        new_nch != op_state.nch)
    {
        /*
         * The effect plugin changes the stream format. Reopen the
         * audio device.
         */
        if (!output_open_audio(new_format, new_rate, new_nch))
            return;
    }

    length = effect_do_mod_samples(&ptr, length, op_state.fmt, op_state.rate, 
        op_state.nch);

    flow_execute(postproc_flow, ptr, length, op_state.fmt, op_state.rate, op_state.nch);

    writeoffs = 0;
    while (writeoffs < length)
    {
        int writable = length - writeoffs;

        if (writable > 2048)
            writable = 2048;

        if (writable == 0)
            return;

        while (op->buffer_free() < writable)   /* wait output buf */
        {
            if (going && !*going)              /* thread stopped? */
                return;                        /* so finish */

            if (ip_data.stop)                  /* has a stop been requested? */
                return;                        /* yes, so finish */

            g_usleep(10000);                   /* else sleep for retry */
        }

        if (ip_data.stop)
            return;

        if (going && !*going)                  /* thread stopped? */
            return;                            /* so finish */

        /* do output */
        op->write_audio(((guint8 *) ptr) + writeoffs, writable);

        writeoffs += writable;
    }
}