annotate src/alac/plugin.c @ 2284:d19b53359b24

cleaned up the sndfile wav plugin, currently limiting it ONLY TO WAV PLAYBACK. if somebody is more experienced with it and wants to restore the other formats, go ahead (maybe change the name of the plugin too?).

/*
 * Copyright (c) 2005 David Hammerton
 * All rights reserved.
 *
 * Adapted from example program by William Pitcock <[e-mail]>
 *
 * Permission is hereby granted, free of charge, to any person
 * obtaining a copy of this software and associated documentation
 * files (the "Software"), to deal in the Software without
 * restriction, including without limitation the rights to use,
 * copy, modify, merge, publish, distribute, sublicense, and/or
 * sell copies of the Software, and to permit persons to whom the
 * Software is furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be
 * included in all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
 * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
 * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
 * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
 * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
 * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
 * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
 * OTHER DEALINGS IN THE SOFTWARE.
 */

#include "config.h"

#include <ctype.h>
#include <stdio.h>

#if HAVE_STDINT_H
# include <stdint.h>
#else
# if HAVE_INTTYPES_H
# include <inttypes.h>
# endif
#endif

#include <stdlib.h>
#include <errno.h>
#include <string.h>

#include <glib.h>
#include <audacious/i18n.h>

#include <audacious/plugin.h>
#include <audacious/main.h>
#include <audacious/output.h>
#include <audacious/vfs.h>
#include <audacious/util.h>

#include "demux.h"
#include "decomp.h"
#include "stream.h"

static int input_opened = 0;

gpointer decode_thread(void *args);
static GThread *playback_thread;
static int going = 0;
static int seek_to = -1;

extern void set_endian();

static void alac_about(void)
{
	static GtkWidget *aboutbox;

	if(aboutbox != NULL)
		return;

	aboutbox = audacious_info_dialog(_("About Apple Lossless Audio Plugin"),
				     _("Copyright (c) 2006 Audacious team\n"
				     "Portions (c) 2005-2006 David Hammerton <crazney -at- crazney.net>"),
				     _("Ok"), FALSE, NULL, NULL);

	g_signal_connect(G_OBJECT(aboutbox), "destroy",
			 G_CALLBACK(gtk_widget_destroyed),
			 &aboutbox);
}

gboolean is_our_fd(char *filename, VFSFile* input_file)
{
    demux_res_t demux_res;
    stream_t *input_stream;

    input_stream = stream_create_file(input_file, 1);

    set_endian();

    if (!input_stream)
        return FALSE;

    /* if qtmovie_read returns successfully, the stream is up to
     * the movie data, which can be used directly by the decoder */
    if (!qtmovie_read(input_stream, &demux_res))
    {
        stream_destroy(input_stream);
        return FALSE;
    }

    stream_destroy(input_stream);

    return TRUE;
}

Tuple *build_aud_tuple_from_demux(demux_res_t *demux_res, char *path)
{
    Tuple *ti = aud_tuple_new_from_filename(path);

    if (demux_res->tuple.art != NULL)
        aud_tuple_associate_string(ti, FIELD_ARTIST, NULL, demux_res->tuple.art);
    if (demux_res->tuple.nam != NULL)
        aud_tuple_associate_string(ti, FIELD_TITLE, NULL, demux_res->tuple.nam);
    if (demux_res->tuple.alb != NULL)
        aud_tuple_associate_string(ti, FIELD_ALBUM, NULL, demux_res->tuple.alb);
    if (demux_res->tuple.gen != NULL)
        aud_tuple_associate_string(ti, FIELD_GENRE, NULL, demux_res->tuple.gen);
    if (demux_res->tuple.cmt != NULL)
        aud_tuple_associate_string(ti, FIELD_COMMENT, NULL, demux_res->tuple.cmt);
    if (demux_res->tuple.day != NULL)
        aud_tuple_associate_int(ti, FIELD_YEAR, NULL, atoi(demux_res->tuple.day));

    aud_tuple_associate_string(ti, FIELD_CODEC, NULL, "Apple Lossless (ALAC)");
    aud_tuple_associate_string(ti, FIELD_QUALITY, NULL, "lossless");

    return ti;
}

Tuple *build_tuple(char *filename)
{
    demux_res_t demux_res;
    VFSFile *input_file;
    stream_t *input_stream;

    input_file = aud_vfs_fopen(filename, "rb");
    input_stream = stream_create_file(input_file, 1);

    set_endian();

    if (!input_stream)
    {
	aud_vfs_fclose(input_file);
        return NULL;
    }

    /* if qtmovie_read returns successfully, the stream is up to
     * the movie data, which can be used directly by the decoder */
    if (!qtmovie_read(input_stream, &demux_res))
    {
        stream_destroy(input_stream);
        aud_vfs_fclose(input_file);
        return NULL;
    }

    stream_destroy(input_stream);
    aud_vfs_fclose(input_file);

    return build_aud_tuple_from_demux(&demux_res, filename);
}

static InputPlayback *playback;

static void play_file(InputPlayback *data)
{
    char *filename = data->filename;
    going = 1;
    playback = data;
    playback_thread = g_thread_self();
    playback->set_pb_ready(playback);
    decode_thread(filename);
}

static void stop(InputPlayback * data)
{
    going = 0;
    g_thread_join(playback_thread);
    data->output->close_audio();
}

static void do_pause(InputPlayback *data, short paused)
{
    data->output->pause(paused);
}

static void seek(InputPlayback * data, gint time)
{
    seek_to = time;
}

static int get_sample_info(demux_res_t *demux_res, uint32_t samplenum,
                           uint32_t *sample_duration,
                           uint32_t *sample_byte_size)
{
    unsigned int duration_index_accum = 0;
    unsigned int duration_cur_index = 0;

    if (samplenum >= demux_res->num_sample_byte_sizes)
        return 0;

    if (!demux_res->num_time_to_samples)
        return 0;

    while ((demux_res->time_to_sample[duration_cur_index].sample_count + duration_index_accum)
            <= samplenum)
    {
        duration_index_accum += demux_res->time_to_sample[duration_cur_index].sample_count;
        duration_cur_index++;

        if (duration_cur_index >= demux_res->num_time_to_samples)
            return 0;
    }

    *sample_duration = demux_res->time_to_sample[duration_cur_index].sample_duration;
    *sample_byte_size = demux_res->sample_byte_size[samplenum];

    return 1;
}

void GetBuffer(demux_res_t *demux_res)
{
    unsigned long destBufferSize = 1024*16; /* 16kb buffer = 4096 frames = 1 alac sample */
    void *pDestBuffer = malloc(destBufferSize);
    int bytes_read = 0;

    unsigned int buffer_size = 1024*128;
    void *buffer;

    unsigned int i;

    buffer = malloc(buffer_size);

    for (i = 0; i < demux_res->num_sample_byte_sizes && going == 1; i++)
    {
        uint32_t sample_duration;
        uint32_t sample_byte_size;

        int outputBytes;

#if 0
	/* XXX: Horribly inaccurate seek. -nenolod */
	if (seek_to != -1)
	{
	    gulong duration =
		(demux_res->num_sample_byte_sizes * (float)((1024 * demux_res->sample_size) - 1.0) /
	       		(float)(demux_res->sample_rate / 251));

	    i = (duration - seek_to) / demux_res->num_sample_byte_sizes;

	    g_print("seek to ALAC frame: %d\n", i);

	    seek_to = -1;
	}
#endif

        /* just get one sample for now */
        if (!get_sample_info(demux_res, i,
                             &sample_duration, &sample_byte_size))
            return;

        if (buffer_size < sample_byte_size)
            return;

        stream_read(demux_res->stream, sample_byte_size, buffer);

        /* now fetch */
        outputBytes = destBufferSize;
        decode_frame(demux_res->alac, buffer, pDestBuffer, &outputBytes);

        /* write */
        bytes_read += outputBytes;

        playback->pass_audio(playback, FMT_S16_LE, demux_res->num_channels, outputBytes, pDestBuffer, &going);
    }

    free(buffer);
    free(pDestBuffer);
}

static gchar *fmts[] = { "m4a", "alac", NULL };

InputPlugin alac_ip = {
    .description = "Apple Lossless Plugin",
    .about = alac_about,
    .play_file = play_file,
    .stop = stop,
    .pause = do_pause,
    .seek = seek,
    .get_song_tuple = build_tuple,
    .is_our_file_from_vfs = is_our_fd,
    .vfs_extensions = fmts,
};

InputPlugin *alac_iplist[] = { &alac_ip, NULL };

DECLARE_PLUGIN(alac, NULL, NULL, alac_iplist, NULL, NULL, NULL, NULL, NULL);

gpointer decode_thread(void *args)
{
    demux_res_t demux_res;
    gulong duration = 0;	/* samples added up */
    VFSFile *input_file;
    stream_t *input_stream;
    Tuple *ti;
    gchar *title;

    memset(&demux_res, 0, sizeof(demux_res));

    set_endian();

    input_file = aud_vfs_fopen((char *) args, "rb");
    input_stream = stream_create_file(input_file, 1);

    if (!input_stream)
        return NULL;

    /* if qtmovie_read returns successfully, the stream is up to
     * the movie data, which can be used directly by the decoder */
    if (!qtmovie_read(input_stream, &demux_res))
        return NULL;

    demux_res.stream = input_stream;

    /* Get the titlestring ready. */
    ti = build_aud_tuple_from_demux(&demux_res, (char *) args);
    title = aud_tuple_formatter_make_title_string(ti, aud_get_gentitle_format());

    /* initialise the sound converter */
    demux_res.alac = create_alac(demux_res.sample_size, demux_res.num_channels);
    alac_set_info(demux_res.alac, demux_res.codecdata);

    /* Sample rates are multiples of 251?! Apple is *fucking* *insane*! -nenolod */
    duration = (demux_res.num_sample_byte_sizes * (float)((1024 * demux_res.sample_size) - 1.0) /
	(float)(demux_res.sample_rate / 251));

    playback->output->open_audio(FMT_S16_LE, demux_res.sample_rate, demux_res.num_channels);
    playback->set_params(playback, title, duration, -1, demux_res.sample_rate, demux_res.num_channels);

    /* will convert the entire buffer */
    GetBuffer(&demux_res);

    going = 0;

    stream_destroy(input_stream);

    if (input_opened)
        aud_vfs_fclose(input_file);

    playback->output->close_audio();

    return NULL;
}