path: root/src/decoder/audiofile_plugin.c

/* the Music Player Daemon (MPD)
 * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
 * This project's homepage is: http://www.musicpd.org
 * 
 * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
 * 
 * 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; either version 2 of the License, or
 * (at your option) any later version.
 *
 * 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, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

#include "../decoder_api.h"

#include <audiofile.h>
#include <af_vfs.h>
#include <glib.h>

#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "audiofile"

/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
#define CHUNK_SIZE		1020

static int getAudiofileTotalTime(const char *file)
{
	int total_time;
	AFfilehandle af_fp = afOpenFile(file, "r", NULL);
	if (af_fp == AF_NULL_FILEHANDLE) {
		return -1;
	}
	total_time = (int)
	    ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK)
	     / afGetRate(af_fp, AF_DEFAULT_TRACK));
	afCloseFile(af_fp);
	return total_time;
}

static ssize_t
audiofile_file_read(AFvirtualfile *vfile, void *data, size_t nbytes)
{
	struct input_stream *is = (struct input_stream *) vfile->closure;
	return input_stream_read(is, data, nbytes);
}

static long
audiofile_file_length(AFvirtualfile *vfile)
{
	struct input_stream *is = (struct input_stream *) vfile->closure;
	return is->size;
}

static long
audiofile_file_tell(AFvirtualfile *vfile)
{
	struct input_stream *is = (struct input_stream *) vfile->closure;
	return is->offset;
}

static void
audiofile_file_destroy(AFvirtualfile *vfile)
{
	struct input_stream *is = (struct input_stream *) vfile->closure;
	vfile->closure = NULL;
	input_stream_close(is);
}

static long
audiofile_file_seek(AFvirtualfile *vfile, long offset, int is_relative)
{
	struct input_stream *is = (struct input_stream *) vfile->closure;
	int whence = (is_relative ? SEEK_CUR : SEEK_SET);
	if (input_stream_seek(is, offset, whence)) { 
		return is->offset;
	} else {
		return -1;
	}
}

static AFvirtualfile *
setup_virtual_fops(struct input_stream *stream)
{
	AFvirtualfile *vf = g_malloc(sizeof(AFvirtualfile));
	vf->closure = stream;
	vf->write   = NULL;
	vf->read    = audiofile_file_read;
	vf->length  = audiofile_file_length;
	vf->destroy = audiofile_file_destroy;
	vf->seek    = audiofile_file_seek;
	vf->tell    = audiofile_file_tell;
	return vf;
}

static void
audiofile_streamdecode(struct decoder * decoder, struct input_stream *inStream)
{
	AFvirtualfile *vf;
	int fs, frame_count;
	AFfilehandle af_fp;
	int bits;
	struct audio_format audio_format;
	float total_time;
	uint16_t bitRate;
	int ret, current = 0;
	char chunk[CHUNK_SIZE];

	vf = setup_virtual_fops(inStream);

	af_fp = afOpenVirtualFile(vf, "r", NULL);
	if (af_fp == AF_NULL_FILEHANDLE) {
		g_warning("failed to input stream\n");
		return;
	}

	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
	                         AF_SAMPFMT_TWOSCOMP, 16);
	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
	audio_format.bits = (uint8_t)bits;
	audio_format.sample_rate =
	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
	audio_format.channels =
	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);

	if (!audio_format_valid(&audio_format)) {
		g_warning("Invalid audio format: %u:%u:%u\n",
			  audio_format.sample_rate, audio_format.bits,
			  audio_format.channels);
		afCloseFile(af_fp);
		return;
	}

	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);

	total_time = ((float)frame_count / (float)audio_format.sample_rate);

	bitRate = (uint16_t)(inStream->size * 8.0 / total_time / 1000.0 + 0.5);

	fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);

	decoder_initialized(decoder, &audio_format, true, total_time);

	do {
		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
			current = decoder_seek_where(decoder) *
				audio_format.sample_rate;
			afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
			decoder_command_finished(decoder);
		}

		ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
				   CHUNK_SIZE / fs);
		if (ret <= 0)
			break;

		current += ret;
		decoder_data(decoder, NULL,
			     chunk, ret * fs,
			     (float)current / (float)audio_format.sample_rate,
			     bitRate, NULL);
	} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);

	afCloseFile(af_fp);
}

static struct tag *audiofile_tag_dup(const char *file)
{
	struct tag *ret = NULL;
	int total_time = getAudiofileTotalTime(file);

	if (total_time >= 0) {
		ret = tag_new();
		ret->time = total_time;
	} else {
		g_debug("Failed to get total song time from: %s\n",
			file);
	}

	return ret;
}

static const char *const audiofile_suffixes[] = {
	"wav", "au", "aiff", "aif", NULL
};

static const char *const audiofile_mime_types[] = {
	"audio/x-wav",
	"audio/x-aiff",
	NULL 
};

const struct decoder_plugin audiofilePlugin = {
	.name = "audiofile",
	.stream_decode = audiofile_streamdecode,
	.tag_dup = audiofile_tag_dup,
	.suffixes = audiofile_suffixes,
	.mime_types = audiofile_mime_types,
};