comparison src/flacng/plugin.c @ 2360:ead24454f4b7

native 24bit output for FLAC

2359:b3475063c000

 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
 */

#include "flacng.h"
#include <audacious/util.h>
#include <audacious/output.h>
#include <audacious/i18n.h>

            test_info->stream.channels);
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    if ((16 != test_info->stream.bits_per_sample) &&
        (24 != test_info->stream.bits_per_sample) &&
        (8 != test_info->stream.bits_per_sample)) {
        _ERROR("This number of bits (%d) is currently not supported, stream not handled by this plugin",
            test_info->stream.bits_per_sample);
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    _LEAVE ret;
}

/* --- */

void squeeze_audio(gint32* src, void* dst, guint count, guint src_res, guint dst_res) {

    /*
     * Changes the sample width of count samples in src from
     * src_res to dst_res and places the result in dst
     */

    gint i;
    gint32* rp = src;
    gint8* wp = dst;
    gint16* wp2 = dst;
    gint32* wp4 = dst;

    _ENTER;

    _DEBUG("Converting %d samples %d->%d", count, src_res, dst_res);

    if ((src_res % 8 != 0) || (dst_res % 8 != 0)) {
        _ERROR("Can not convert from %d bps to %d bps: not a multiple of 8",
                src_res, dst_res);
        _LEAVE;
    }

    if (16 == dst_res) {
        if (8 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = (*rp << 8) & 0xffff;
            }
        } else if (16 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = *rp & 0xffff;
            }
        } else if (24 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = (*rp >> 8) & 0xffff;
            }
        } else if (32 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = (*rp >> 16) & 0xffff;
            }
        }
    } else if (8 == dst_res) {
        if (8 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = *rp & 0xff;
            }
        } else if (16 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = (*rp >> 8) & 0xff;
            }
        } else if (24 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = (*rp >> 16) & 0xff;
            }
        } else if (32 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = (*rp >> 24) & 0xff;
            }
        }
    } else if (32 == dst_res) {
        if (8 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = (*rp << 24) & 0xffffffff;
            }
        } else if (16 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = (*rp << 16) & 0xffffffff;
            }
        } else if (24 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = (*rp << 8) & 0xffffffff;
            }
        } else if (32 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = *rp;
            }
        }
    }

    _LEAVE;
}

    stream_info.samplerate = main_info->stream.samplerate;
    stream_info.channels = main_info->stream.channels;
    main_info->metadata_changed = FALSE;

    if (!playback->output->open_audio(FMT_S16_NE,
        main_info->stream.samplerate,
        main_info->stream.channels)) {
        playback->playing = FALSE;
        _ERROR("Could not open output plugin!");
        _LEAVE NULL;
    }

        while ((TRUE == playback->playing) && (0 != elements_left)) {

            sample_count = MIN(OUTPUT_BLOCK_SIZE, elements_left);

            /*
             * Convert the audio.
             * Currently this is hardcoded to 16 bit.
             * 8 and 24 bit are sampled up/down linear.
             */
            _DEBUG("Converting %d samples to FMT_S16_NE", sample_count);
            squeeze_audio(read_pointer, play_buffer, sample_count, main_info->frame.bits_per_sample, 16);

            _DEBUG("Copying %d samples to output plugin", sample_count);

            playback->pass_audio(playback, FMT_S16_NE, main_info->frame.channels, sample_count * sizeof(gint16), play_buffer, NULL);

            read_pointer += sample_count;
            elements_left -= sample_count;

            _DEBUG("%d elements left to be output", elements_left);

2360:ead24454f4b7

 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
 */
/* #define FLACNG_DEBUG */

#include "flacng.h"
#include <audacious/util.h>
#include <audacious/output.h>
#include <audacious/i18n.h>

            test_info->stream.channels);
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    if ((8  != test_info->stream.bits_per_sample) &&
        (16 != test_info->stream.bits_per_sample) &&
        (24 != test_info->stream.bits_per_sample) &&
        (32 != test_info->stream.bits_per_sample)) {
        _ERROR("This number of bits (%d) is currently not supported, stream not handled by this plugin",
            test_info->stream.bits_per_sample);
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    _LEAVE ret;
}

/* --- */

void squeeze_audio(gint32* src, void* dst, guint count, guint res) {

    gint i;
    gint32* rp = src;
    gint8* wp = dst;
    gint16* wp2 = dst;
    gint32* wp4 = dst;

    _ENTER;

    _DEBUG("Converting %d samples to %d bit", count, res);

    if (res % 8 != 0) {
        _ERROR("Can not convert to %d bps: not a multiple of 8", res);
        _LEAVE;
    }

    if (res == 8) {
        for (i=0; i<count; i++, wp++, rp++) {
            *wp = *rp & 0xff;
        }
    } else if (res == 16) {
        for (i=0; i<count; i++, wp2++, rp++) {
            *wp2 = *rp & 0xffff;
        }
    } else if (res == 24 || res == 32) { /* 24bit value stored in lowest 3 bytes */
       for (i=0; i<count; i++, wp4++, rp++) {
           *wp4 = *rp;
       }
    }

    _LEAVE;
}

    stream_info.samplerate = main_info->stream.samplerate;
    stream_info.channels = main_info->stream.channels;
    main_info->metadata_changed = FALSE;

    if (!playback->output->open_audio(SAMPLE_FMT(main_info->stream.bits_per_sample),
                                      main_info->stream.samplerate,
                                      main_info->stream.channels)) {
        playback->playing = FALSE;
        _ERROR("Could not open output plugin!");
        _LEAVE NULL;
    }

        while ((TRUE == playback->playing) && (0 != elements_left)) {

            sample_count = MIN(OUTPUT_BLOCK_SIZE, elements_left);

            squeeze_audio(read_pointer, play_buffer, sample_count, main_info->stream.bits_per_sample);

            _DEBUG("Copying %d samples to output plugin", sample_count);

            playback->pass_audio(playback,
                                 SAMPLE_FMT(main_info->stream.bits_per_sample),
                                 main_info->stream.channels,
                                 sample_count * SAMPLE_SIZE(main_info->stream.bits_per_sample),
                                 play_buffer,
                                 NULL);

            read_pointer += sample_count;
            elements_left -= sample_count;

            _DEBUG("%d elements left to be output", elements_left);