mpd/src/decoder/mp4ff_plugin.c

438 lines
10 KiB
C
Raw Normal View History

/*
* Copyright (C) 2003-2009 The Music Player Daemon Project
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "../decoder_api.h"
#include "config.h"
#include "tag_table.h"
#include <glib.h>
#include <mp4ff.h>
#include <faad.h>
#include <assert.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, faacDecHandle decoder,
uint32_t *sample_rate, unsigned char *channels_r)
{
#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
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;
if (mp4ff_get_track_type(infile, i) != 1)
/* not an audio track */
continue;
if (decoder == NULL)
/* have don't have a decoder to initialize -
we're done now, because we found an audio
track */
return i;
mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff == NULL)
continue;
rc = faacDecInit2(decoder, buff, buff_size,
sample_rate_r, channels_r);
free(buff);
if (rc >= 0)
/* found a valid AAC track */
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_r, struct audio_format *audio_format)
{
faacDecHandle decoder;
faacDecConfigurationPtr config;
int track;
uint32_t sample_rate;
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);
track = mp4_get_aac_track(mp4fh, decoder, &sample_rate, &channels);
if (track < 0) {
g_warning("No AAC track found");
faacDecClose(decoder);
return NULL;
}
*track_r = track;
*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;
}
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)(G_MAXINT / sizeof(float))) {
g_warning("Integer overflow.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
return;
}
file_time = 0.0;
seek_table = input_stream->seekable
? g_malloc(sizeof(float) * num_samples)
: NULL;
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) {
assert(seek_table != NULL);
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;
assert(seek_table != NULL);
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 (seek_table != NULL && 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
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 const char *const mp4ff_tag_names[TAG_NUM_OF_ITEM_TYPES] = {
[TAG_ITEM_ALBUM_ARTIST] = "album artist",
[TAG_ITEM_COMPOSER] = "writer",
[TAG_ITEM_PERFORMER] = "band",
};
static enum tag_type
mp4ff_tag_name_parse(const char *name)
{
enum tag_type type = tag_table_lookup(mp4ff_tag_names, name);
if (type == TAG_NUM_OF_ITEM_TYPES)
type = tag_name_parse_i(name);
return type;
}
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, NULL, NULL, NULL);
if (track < 0) {
mp4ff_close(mp4fh);
input_stream_close(&input_stream);
return NULL;
}
2008-08-29 09:38:21 +02:00
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);
2008-08-29 09:38:21 +02:00
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);
enum tag_type type = mp4ff_tag_name_parse(item);
if (type != TAG_NUM_OF_ITEM_TYPES)
tag_add_item(ret, type, 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,
};