aboutsummaryrefslogtreecommitdiffstats
path: root/src/decoder/mp4ff_plugin.c
blob: fb0f007763e5f2a73623f402aa2950952babad1f (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
/* 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
 *
 * 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 "config.h"

#include "mp4ff.h"

#include <limits.h>
#include <faad.h>
#include <glib.h>
#include <stdlib.h>
#include <unistd.h>

#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "mp4ff"

/* all code here is either based on or copied from FAAD2's frontend code */

struct mp4_context {
	struct decoder *decoder;
	struct input_stream *input_stream;
};

static int
mp4_get_aac_track(mp4ff_t * infile)
{
	/* find AAC track */
	int i, rc;
	int num_tracks = mp4ff_total_tracks(infile);

	for (i = 0; i < num_tracks; i++) {
		unsigned char *buff = NULL;
		unsigned int buff_size = 0;
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
		mp4AudioSpecificConfig mp4ASC;
#else
		unsigned long dummy1_32;
		unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
		    dummy7_8, dummy8_8;
#endif

		mp4ff_get_decoder_config(infile, i, &buff, &buff_size);

		if (buff) {
#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
			rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
#else
			rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
						 &dummy3_8, &dummy4_8,
						 &dummy5_8, &dummy6_8,
						 &dummy7_8, &dummy8_8);
#endif
			free(buff);
			if (rc < 0)
				continue;
			return i;
		}
	}

	/* can't decode this */
	return -1;
}

static uint32_t
mp4_read(void *user_data, void *buffer, uint32_t length)
{
	struct mp4_context *ctx = user_data;

	return decoder_read(ctx->decoder, ctx->input_stream, buffer, length);
}

static uint32_t
mp4_seek(void *user_data, uint64_t position)
{
	struct mp4_context *ctx = user_data;

	return input_stream_seek(ctx->input_stream, position, SEEK_SET)
		? 0 : -1;
}

static faacDecHandle
mp4_faad_new(mp4ff_t *mp4fh, int track, struct audio_format *audio_format)
{
	faacDecHandle decoder;
	faacDecConfigurationPtr config;
	unsigned char *mp4_buffer;
	unsigned int mp4_buffer_size;
	uint32_t sample_rate;
#ifdef HAVE_FAAD_LONG
	/* neaacdec.h declares all arguments as "unsigned long", but
	   internally expects uint32_t pointers.  To avoid gcc
	   warnings, use this workaround. */
	unsigned long *sample_rate_r = (unsigned long*)&sample_rate;
#else
	uint32_t *sample_rate_r = &sample_rate;
#endif
	unsigned char channels;

	decoder = faacDecOpen();

	config = faacDecGetCurrentConfiguration(decoder);
	config->outputFormat = FAAD_FMT_16BIT;
#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
	config->downMatrix = 1;
#endif
#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
	config->dontUpSampleImplicitSBR = 0;
#endif
	faacDecSetConfiguration(decoder, config);

	mp4_buffer = NULL;
	mp4_buffer_size = 0;
	mp4ff_get_decoder_config(mp4fh, track, &mp4_buffer, &mp4_buffer_size);

	if (faacDecInit2(decoder, mp4_buffer, mp4_buffer_size,
			 sample_rate_r, &channels) < 0) {
		g_warning("Not an AAC stream.\n");
		faacDecClose(decoder);
		return NULL;
	}

	*audio_format = (struct audio_format){
		.bits = 16,
		.channels = channels,
		.sample_rate = sample_rate,
	};

	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);
		faacDecClose(decoder);
		return NULL;
	}

	return decoder;
}

static void
mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
{
	struct mp4_context ctx = {
		.decoder = mpd_decoder,
		.input_stream = input_stream,
	};
	mp4ff_callback_t callback = {
		.read = mp4_read,
		.seek = mp4_seek,
		.user_data = &ctx,
	};
	mp4ff_t *mp4fh;
	int32_t track;
	float file_time, total_time;
	int32_t scale;
	faacDecHandle decoder;
	struct audio_format audio_format;
	faacDecFrameInfo frame_info;
	unsigned char *mp4_buffer;
	unsigned int mp4_buffer_size;
	long sample_id;
	long num_samples;
	long dur;
	unsigned int sample_count;
	char *sample_buffer;
	size_t sample_buffer_length;
	unsigned int initial = 1;
	float *seek_table;
	long seek_table_end = -1;
	bool seek_position_found = false;
	long offset;
	uint16_t bit_rate = 0;
	bool seeking = false;
	double seek_where = 0;
	enum decoder_command cmd = DECODE_COMMAND_NONE;

	mp4fh = mp4ff_open_read(&callback);
	if (!mp4fh) {
		g_warning("Input does not appear to be a mp4 stream.\n");
		return;
	}

	track = mp4_get_aac_track(mp4fh);
	if (track < 0) {
		g_warning("No AAC track found in mp4 stream.\n");
		mp4ff_close(mp4fh);
		return;
	}

	decoder = mp4_faad_new(mp4fh, track, &audio_format);
	if (decoder == NULL) {
		mp4ff_close(mp4fh);
		return;
	}

	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
	scale = mp4ff_time_scale(mp4fh, track);

	if (scale < 0) {
		g_warning("Error getting audio format of mp4 AAC track.\n");
		faacDecClose(decoder);
		mp4ff_close(mp4fh);
		return;
	}
	total_time = ((float)file_time) / scale;

	num_samples = mp4ff_num_samples(mp4fh, track);
	if (num_samples > (long)(INT_MAX / sizeof(float))) {
		 g_warning("Integer overflow.\n");
		 faacDecClose(decoder);
		 mp4ff_close(mp4fh);
		 return;
	}

	file_time = 0.0;

	seek_table = g_malloc(sizeof(float) * num_samples);

	decoder_initialized(mpd_decoder, &audio_format,
			    input_stream->seekable,
			    total_time);

	for (sample_id = 0;
	     sample_id < num_samples && cmd != DECODE_COMMAND_STOP;
	     sample_id++) {
		if (cmd == DECODE_COMMAND_SEEK) {
			seeking = true;
			seek_where = decoder_seek_where(mpd_decoder);
		}

		if (seeking && seek_table_end > 1 &&
		    seek_table[seek_table_end] >= seek_where) {
			int i = 2;
			while (seek_table[i] < seek_where)
				i++;
			sample_id = i - 1;
			file_time = seek_table[sample_id];
		}

		dur = mp4ff_get_sample_duration(mp4fh, track, sample_id);
		offset = mp4ff_get_sample_offset(mp4fh, track, sample_id);

		if (sample_id > seek_table_end) {
			seek_table[sample_id] = file_time;
			seek_table_end = sample_id;
		}

		if (sample_id == 0)
			dur = 0;
		if (offset > dur)
			dur = 0;
		else
			dur -= offset;
		file_time += ((float)dur) / scale;

		if (seeking && file_time > seek_where)
			seek_position_found = true;

		if (seeking && seek_position_found) {
			seek_position_found = false;
			seeking = 0;
			decoder_command_finished(mpd_decoder);
		}

		if (seeking)
			continue;

		if (mp4ff_read_sample(mp4fh, track, sample_id, &mp4_buffer,
				      &mp4_buffer_size) == 0)
			break;

#ifdef HAVE_FAAD_BUFLEN_FUNCS
		sample_buffer = faacDecDecode(decoder, &frame_info, mp4_buffer,
					      mp4_buffer_size);
#else
		sample_buffer = faacDecDecode(decoder, &frame_info, mp4_buffer);
#endif

		if (mp4_buffer)
			free(mp4_buffer);
		if (frame_info.error > 0) {
			g_warning("faad2 error: %s\n",
				  faacDecGetErrorMessage(frame_info.error));
			break;
		}

		if (frame_info.channels != audio_format.channels) {
			g_warning("channel count changed from %u to %u",
				  audio_format.channels, frame_info.channels);
			break;
		}

#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
		if (frame_info.samplerate != audio_format.sample_rate) {
			g_warning("sample rate changed from %u to %lu",
				  audio_format.sample_rate,
				  (unsigned long)frame_info.samplerate);
			break;
		}
#endif

		if (audio_format.channels * (unsigned long)(dur + offset) > frame_info.samples) {
			dur = frame_info.samples / audio_format.channels;
			offset = 0;
		}

		sample_count = (unsigned long)(dur * audio_format.channels);

		if (sample_count > 0) {
			initial = 0;
			bit_rate = frame_info.bytesconsumed * 8.0 *
			    frame_info.channels * scale /
			    frame_info.samples / 1000 + 0.5;
		}

		sample_buffer_length = sample_count * 2;

		sample_buffer += offset * audio_format.channels * 2;

		cmd = decoder_data(mpd_decoder, input_stream,
				   sample_buffer, sample_buffer_length,
				   file_time, bit_rate, NULL);
	}

	g_free(seek_table);
	faacDecClose(decoder);
	mp4ff_close(mp4fh);
}

static struct tag *
mp4_tag_dup(const char *file)
{
	struct tag *ret = NULL;
	struct input_stream input_stream;
	struct mp4_context ctx = {
		.decoder = NULL,
		.input_stream = &input_stream,
	};
	mp4ff_callback_t callback = {
		.read = mp4_read,
		.seek = mp4_seek,
		.user_data = &ctx,
	};
	mp4ff_t *mp4fh;
	int32_t track;
	int32_t file_time;
	int32_t scale;
	int i;

	if (!input_stream_open(&input_stream, file)) {
		g_warning("Failed to open file: %s", file);
		return NULL;
	}

	mp4fh = mp4ff_open_read(&callback);
	if (!mp4fh) {
		input_stream_close(&input_stream);
		return NULL;
	}

	track = mp4_get_aac_track(mp4fh);
	if (track < 0) {
		mp4ff_close(mp4fh);
		input_stream_close(&input_stream);
		return NULL;
	}

	ret = tag_new();
	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
	scale = mp4ff_time_scale(mp4fh, track);
	if (scale < 0) {
		mp4ff_close(mp4fh);
		input_stream_close(&input_stream);
		tag_free(ret);
		return NULL;
	}
	ret->time = ((float)file_time) / scale + 0.5;

	for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
		char *item;
		char *value;

		mp4ff_meta_get_by_index(mp4fh, i, &item, &value);

		if (0 == strcasecmp("artist", item)) {
			tag_add_item(ret, TAG_ITEM_ARTIST, value);
		} else if (0 == strcasecmp("title", item)) {
			tag_add_item(ret, TAG_ITEM_TITLE, value);
		} else if (0 == strcasecmp("album", item)) {
			tag_add_item(ret, TAG_ITEM_ALBUM, value);
		} else if (0 == strcasecmp("track", item)) {
			tag_add_item(ret, TAG_ITEM_TRACK, value);
		} else if (0 == strcasecmp("disc", item)) {	/* Is that the correct id? */
			tag_add_item(ret, TAG_ITEM_DISC, value);
		} else if (0 == strcasecmp("genre", item)) {
			tag_add_item(ret, TAG_ITEM_GENRE, value);
		} else if (0 == strcasecmp("date", item)) {
			tag_add_item(ret, TAG_ITEM_DATE, value);
		} else if (0 == strcasecmp("writer", item)) {
			tag_add_item(ret, TAG_ITEM_COMPOSER, value);
		}

		free(item);
		free(value);
	}

	mp4ff_close(mp4fh);
	input_stream_close(&input_stream);

	return ret;
}

static const char *const mp4_suffixes[] = { "m4a", "mp4", NULL };
static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL };

const struct decoder_plugin mp4ff_decoder_plugin = {
	.name = "mp4",
	.stream_decode = mp4_decode,
	.tag_dup = mp4_tag_dup,
	.suffixes = mp4_suffixes,
	.mime_types = mp4_mime_types,
};