renamed src/inputPlugins/ to src/decoder/
These plugins are not input plugins, they are decoder plugins. No CamelCase in the directory name.
This commit is contained in:
Max Kellermann
423
src/decoder/mp4_plugin.c
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423
src/decoder/mp4_plugin.c
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@@ -0,0 +1,423 @@
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/* the Music Player Daemon (MPD)
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* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
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* This project's homepage is: http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "../decoder_api.h"
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#include "../utils.h"
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#include "../log.h"
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#include "mp4ff.h"
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#include <limits.h>
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#include <faad.h>
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/* all code here is either based on or copied from FAAD2's frontend code */
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static int mp4_getAACTrack(mp4ff_t * infile)
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{
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/* find AAC track */
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int i, rc;
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int numTracks = mp4ff_total_tracks(infile);
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for (i = 0; i < numTracks; i++) {
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unsigned char *buff = NULL;
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unsigned int buff_size = 0;
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#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
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mp4AudioSpecificConfig mp4ASC;
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#else
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unsigned long dummy1_32;
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unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
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dummy7_8, dummy8_8;
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#endif
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mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
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if (buff) {
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#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
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rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
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#else
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rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
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&dummy3_8, &dummy4_8,
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&dummy5_8, &dummy6_8,
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&dummy7_8, &dummy8_8);
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#endif
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free(buff);
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if (rc < 0)
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continue;
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return i;
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}
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}
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/* can't decode this */
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return -1;
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}
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static uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer,
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uint32_t length)
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{
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return readFromInputStream((InputStream *) inStream, buffer, length);
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}
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static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
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{
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return seekInputStream((InputStream *) inStream, position, SEEK_SET);
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}
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static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
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{
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mp4ff_t *mp4fh;
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mp4ff_callback_t *mp4cb;
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int32_t track;
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float file_time, total_time;
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int32_t scale;
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faacDecHandle decoder;
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faacDecFrameInfo frameInfo;
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faacDecConfigurationPtr config;
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struct audio_format audio_format;
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unsigned char *mp4Buffer;
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unsigned int mp4BufferSize;
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uint32_t sample_rate;
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unsigned char channels;
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long sampleId;
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long numSamples;
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long dur;
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unsigned int sampleCount;
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char *sampleBuffer;
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size_t sampleBufferLen;
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unsigned int initial = 1;
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float *seekTable;
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long seekTableEnd = -1;
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bool seekPositionFound = false;
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long offset;
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uint16_t bitRate = 0;
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bool seeking = false;
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double seek_where = 0;
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bool initialized = false;
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mp4cb = xmalloc(sizeof(mp4ff_callback_t));
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mp4cb->read = mp4_inputStreamReadCallback;
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mp4cb->seek = mp4_inputStreamSeekCallback;
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mp4cb->user_data = inStream;
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mp4fh = mp4ff_open_read(mp4cb);
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if (!mp4fh) {
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ERROR("Input does not appear to be a mp4 stream.\n");
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free(mp4cb);
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return -1;
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}
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track = mp4_getAACTrack(mp4fh);
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if (track < 0) {
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ERROR("No AAC track found in mp4 stream.\n");
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mp4ff_close(mp4fh);
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free(mp4cb);
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return -1;
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}
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decoder = faacDecOpen();
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config = faacDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
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config->downMatrix = 1;
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#endif
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#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
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config->dontUpSampleImplicitSBR = 0;
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#endif
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faacDecSetConfiguration(decoder, config);
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audio_format.bits = 16;
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mp4Buffer = NULL;
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mp4BufferSize = 0;
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mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
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if (faacDecInit2
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(decoder, mp4Buffer, mp4BufferSize, &sample_rate, &channels) < 0) {
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ERROR("Error not a AAC stream.\n");
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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free(mp4cb);
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return -1;
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}
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audio_format.sample_rate = sample_rate;
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audio_format.channels = channels;
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file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
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scale = mp4ff_time_scale(mp4fh, track);
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if (mp4Buffer)
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free(mp4Buffer);
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if (scale < 0) {
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ERROR("Error getting audio format of mp4 AAC track.\n");
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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free(mp4cb);
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return -1;
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}
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total_time = ((float)file_time) / scale;
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numSamples = mp4ff_num_samples(mp4fh, track);
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if (numSamples > (long)(INT_MAX / sizeof(float))) {
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ERROR("Integer overflow.\n");
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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free(mp4cb);
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return -1;
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}
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file_time = 0.0;
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seekTable = xmalloc(sizeof(float) * numSamples);
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for (sampleId = 0; sampleId < numSamples; sampleId++) {
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if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
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seeking = true;
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seek_where = decoder_seek_where(mpd_decoder);
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}
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if (seeking && seekTableEnd > 1 &&
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seekTable[seekTableEnd] >= seek_where) {
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int i = 2;
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while (seekTable[i] < seek_where)
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i++;
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sampleId = i - 1;
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file_time = seekTable[sampleId];
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}
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dur = mp4ff_get_sample_duration(mp4fh, track, sampleId);
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offset = mp4ff_get_sample_offset(mp4fh, track, sampleId);
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if (sampleId > seekTableEnd) {
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seekTable[sampleId] = file_time;
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seekTableEnd = sampleId;
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}
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if (sampleId == 0)
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dur = 0;
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if (offset > dur)
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dur = 0;
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else
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dur -= offset;
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file_time += ((float)dur) / scale;
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if (seeking && file_time > seek_where)
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seekPositionFound = true;
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if (seeking && seekPositionFound) {
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seekPositionFound = false;
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decoder_clear(mpd_decoder);
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seeking = 0;
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decoder_command_finished(mpd_decoder);
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}
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if (seeking)
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continue;
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if (mp4ff_read_sample(mp4fh, track, sampleId, &mp4Buffer,
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&mp4BufferSize) == 0)
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break;
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#ifdef HAVE_FAAD_BUFLEN_FUNCS
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sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer,
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mp4BufferSize);
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#else
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sampleBuffer = faacDecDecode(decoder, &frameInfo, mp4Buffer);
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#endif
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if (mp4Buffer)
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free(mp4Buffer);
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if (frameInfo.error > 0) {
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ERROR("faad2 error: %s\n",
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faacDecGetErrorMessage(frameInfo.error));
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break;
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}
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if (!initialized) {
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channels = frameInfo.channels;
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#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
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scale = frameInfo.samplerate;
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#endif
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audio_format.sample_rate = scale;
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audio_format.channels = frameInfo.channels;
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decoder_initialized(mpd_decoder, &audio_format,
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total_time);
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initialized = true;
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}
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if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
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dur = frameInfo.samples / channels;
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offset = 0;
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}
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sampleCount = (unsigned long)(dur * channels);
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if (sampleCount > 0) {
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initial = 0;
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bitRate = frameInfo.bytesconsumed * 8.0 *
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frameInfo.channels * scale /
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frameInfo.samples / 1000 + 0.5;
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}
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sampleBufferLen = sampleCount * 2;
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sampleBuffer += offset * channels * 2;
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decoder_data(mpd_decoder, inStream, 1, sampleBuffer,
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sampleBufferLen, file_time,
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bitRate, NULL);
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if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP)
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break;
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}
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free(seekTable);
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faacDecClose(decoder);
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mp4ff_close(mp4fh);
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free(mp4cb);
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if (!initialized)
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return -1;
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if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK && seeking) {
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decoder_clear(mpd_decoder);
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decoder_command_finished(mpd_decoder);
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}
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decoder_flush(mpd_decoder);
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return 0;
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}
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static struct tag *mp4DataDup(char *file, int *mp4MetadataFound)
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{
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struct tag *ret = NULL;
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InputStream inStream;
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mp4ff_t *mp4fh;
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mp4ff_callback_t *callback;
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int32_t track;
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int32_t file_time;
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int32_t scale;
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int i;
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*mp4MetadataFound = 0;
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if (openInputStream(&inStream, file) < 0) {
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DEBUG("mp4DataDup: Failed to open file: %s\n", file);
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return NULL;
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}
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callback = xmalloc(sizeof(mp4ff_callback_t));
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callback->read = mp4_inputStreamReadCallback;
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callback->seek = mp4_inputStreamSeekCallback;
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callback->user_data = &inStream;
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mp4fh = mp4ff_open_read(callback);
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if (!mp4fh) {
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free(callback);
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closeInputStream(&inStream);
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return NULL;
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}
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track = mp4_getAACTrack(mp4fh);
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if (track < 0) {
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mp4ff_close(mp4fh);
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closeInputStream(&inStream);
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free(callback);
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return NULL;
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}
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ret = tag_new();
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file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
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scale = mp4ff_time_scale(mp4fh, track);
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if (scale < 0) {
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mp4ff_close(mp4fh);
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closeInputStream(&inStream);
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free(callback);
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tag_free(ret);
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return NULL;
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}
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ret->time = ((float)file_time) / scale + 0.5;
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for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
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char *item;
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char *value;
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mp4ff_meta_get_by_index(mp4fh, i, &item, &value);
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if (0 == strcasecmp("artist", item)) {
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tag_add_item(ret, TAG_ITEM_ARTIST, value);
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*mp4MetadataFound = 1;
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} else if (0 == strcasecmp("title", item)) {
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tag_add_item(ret, TAG_ITEM_TITLE, value);
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*mp4MetadataFound = 1;
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} else if (0 == strcasecmp("album", item)) {
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tag_add_item(ret, TAG_ITEM_ALBUM, value);
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*mp4MetadataFound = 1;
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} else if (0 == strcasecmp("track", item)) {
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tag_add_item(ret, TAG_ITEM_TRACK, value);
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*mp4MetadataFound = 1;
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} else if (0 == strcasecmp("disc", item)) { /* Is that the correct id? */
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tag_add_item(ret, TAG_ITEM_DISC, value);
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*mp4MetadataFound = 1;
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} else if (0 == strcasecmp("genre", item)) {
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tag_add_item(ret, TAG_ITEM_GENRE, value);
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*mp4MetadataFound = 1;
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} else if (0 == strcasecmp("date", item)) {
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tag_add_item(ret, TAG_ITEM_DATE, value);
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*mp4MetadataFound = 1;
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}
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free(item);
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free(value);
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}
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mp4ff_close(mp4fh);
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closeInputStream(&inStream);
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return ret;
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}
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static struct tag *mp4TagDup(char *file)
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{
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struct tag *ret = NULL;
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int mp4MetadataFound = 0;
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ret = mp4DataDup(file, &mp4MetadataFound);
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if (!ret)
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return NULL;
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if (!mp4MetadataFound) {
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struct tag *temp = tag_id3_load(file);
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if (temp) {
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temp->time = ret->time;
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tag_free(ret);
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ret = temp;
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}
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}
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return ret;
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}
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static const char *mp4_suffixes[] = { "m4a", "mp4", NULL };
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static const char *mp4_mimeTypes[] = { "audio/mp4", "audio/m4a", NULL };
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struct decoder_plugin mp4Plugin = {
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.name = "mp4",
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.stream_decode = mp4_decode,
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.tag_dup = mp4TagDup,
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.stream_types = INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
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.suffixes = mp4_suffixes,
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.mime_types = mp4_mimeTypes,
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};
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Block a user