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-rw-r--r--src/decoder/mp4ff_decoder_plugin.c435
1 files changed, 435 insertions, 0 deletions
diff --git a/src/decoder/mp4ff_decoder_plugin.c b/src/decoder/mp4ff_decoder_plugin.c
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--- /dev/null
+++ b/src/decoder/mp4ff_decoder_plugin.c
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+/*
+ * Copyright (C) 2003-2010 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 "config.h"
+#include "decoder_api.h"
+#include "audio_check.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 mp4ff_input_stream {
+ mp4ff_callback_t callback;
+
+ 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 mp4ff_input_stream *mis = user_data;
+
+ return decoder_read(mis->decoder, mis->input_stream, buffer, length);
+}
+
+static uint32_t
+mp4_seek(void *user_data, uint64_t position)
+{
+ struct mp4ff_input_stream *mis = user_data;
+
+ return input_stream_seek(mis->input_stream, position, SEEK_SET, NULL)
+ ? 0 : -1;
+}
+
+static const mp4ff_callback_t mpd_mp4ff_callback = {
+ .read = mp4_read,
+ .seek = mp4_seek,
+};
+
+static mp4ff_t *
+mp4ff_input_stream_open(struct mp4ff_input_stream *mis,
+ struct decoder *decoder,
+ struct input_stream *input_stream)
+{
+ mis->callback = mpd_mp4ff_callback;
+ mis->callback.user_data = mis;
+ mis->decoder = decoder;
+ mis->input_stream = input_stream;
+
+ return mp4ff_open_read(&mis->callback);
+}
+
+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;
+ GError *error = NULL;
+
+ 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;
+ }
+
+ if (!audio_format_init_checked(audio_format, sample_rate,
+ SAMPLE_FORMAT_S16, channels,
+ &error)) {
+ g_warning("%s", error->message);
+ g_error_free(error);
+ faacDecClose(decoder);
+ return NULL;
+ }
+
+ *track_r = track;
+
+ return decoder;
+}
+
+static void
+mp4_decode(struct decoder *mpd_decoder, struct input_stream *input_stream)
+{
+ struct mp4ff_input_stream mis;
+ 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_input_stream_open(&mis, mpd_decoder, input_stream);
+ 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,
+ bit_rate);
+ }
+
+ g_free(seek_table);
+ faacDecClose(decoder);
+ mp4ff_close(mp4fh);
+}
+
+static const char *const mp4ff_tag_names[TAG_NUM_OF_ITEM_TYPES] = {
+ [TAG_ALBUM_ARTIST] = "album artist",
+ [TAG_COMPOSER] = "writer",
+ [TAG_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);
+
+ if (g_ascii_strcasecmp(name, "albumartist") == 0 ||
+ g_ascii_strcasecmp(name, "album_artist") == 0)
+ return TAG_ALBUM_ARTIST;
+
+ return type;
+}
+
+static struct tag *
+mp4_stream_tag(struct input_stream *is)
+{
+ struct mp4ff_input_stream mis;
+ int32_t track;
+ int32_t file_time;
+ int32_t scale;
+ int i;
+
+ mp4ff_t *mp4fh = mp4ff_input_stream_open(&mis, NULL, is);
+ if (mp4fh == NULL)
+ return NULL;
+
+ track = mp4_get_aac_track(mp4fh, NULL, NULL, NULL);
+ if (track < 0) {
+ mp4ff_close(mp4fh);
+ return NULL;
+ }
+
+ file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+ scale = mp4ff_time_scale(mp4fh, track);
+ if (scale < 0) {
+ mp4ff_close(mp4fh);
+ return NULL;
+ }
+
+ struct tag *tag = tag_new();
+ tag->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(tag, type, value);
+
+ free(item);
+ free(value);
+ }
+
+ mp4ff_close(mp4fh);
+
+ return tag;
+}
+
+static const char *const mp4_suffixes[] = {
+ "m4a",
+ "m4b",
+ "mp4",
+ NULL
+};
+
+static const char *const mp4_mime_types[] = { "audio/mp4", "audio/m4a", NULL };
+
+const struct decoder_plugin mp4ff_decoder_plugin = {
+ .name = "mp4ff",
+ .stream_decode = mp4_decode,
+ .stream_tag = mp4_stream_tag,
+ .suffixes = mp4_suffixes,
+ .mime_types = mp4_mime_types,
+};