/* 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 * * Common data structures and functions used by FLAC and OggFLAC * (c) 2005 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 "_flac_common.h" #include "../log.h" #include <FLAC/format.h> #include <FLAC/metadata.h> void init_FlacData(FlacData * data, struct decoder * decoder, InputStream * inStream) { data->time = 0; data->position = 0; data->bitRate = 0; data->decoder = decoder; data->inStream = inStream; data->replayGainInfo = NULL; data->tag = NULL; } static int flacFindVorbisCommentFloat(const FLAC__StreamMetadata * block, const char *cmnt, float *fl) { int offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, cmnt); if (offset >= 0) { size_t pos = strlen(cmnt) + 1; /* 1 is for '=' */ int len = block->data.vorbis_comment.comments[offset].length - pos; if (len > 0) { unsigned char tmp; unsigned char *p = &(block->data.vorbis_comment. comments[offset].entry[pos]); tmp = p[len]; p[len] = '\0'; *fl = (float)atof((char *)p); p[len] = tmp; return 1; } } return 0; } /* replaygain stuff by AliasMrJones */ static void flacParseReplayGain(const FLAC__StreamMetadata * block, FlacData * data) { int found = 0; if (data->replayGainInfo) freeReplayGainInfo(data->replayGainInfo); data->replayGainInfo = newReplayGainInfo(); found |= flacFindVorbisCommentFloat(block, "replaygain_album_gain", &data->replayGainInfo->albumGain); found |= flacFindVorbisCommentFloat(block, "replaygain_album_peak", &data->replayGainInfo->albumPeak); found |= flacFindVorbisCommentFloat(block, "replaygain_track_gain", &data->replayGainInfo->trackGain); found |= flacFindVorbisCommentFloat(block, "replaygain_track_peak", &data->replayGainInfo->trackPeak); if (!found) { freeReplayGainInfo(data->replayGainInfo); data->replayGainInfo = NULL; } } /* tracknumber is used in VCs, MPD uses "track" ..., all the other * tag names match */ static const char *VORBIS_COMMENT_TRACK_KEY = "tracknumber"; static const char *VORBIS_COMMENT_DISC_KEY = "discnumber"; static unsigned int commentMatchesAddToTag(const FLAC__StreamMetadata_VorbisComment_Entry * entry, unsigned int itemType, struct tag ** tag) { const char *str; size_t slen; int vlen; switch (itemType) { case TAG_ITEM_TRACK: str = VORBIS_COMMENT_TRACK_KEY; break; case TAG_ITEM_DISC: str = VORBIS_COMMENT_DISC_KEY; break; default: str = mpdTagItemKeys[itemType]; } slen = strlen(str); vlen = entry->length - slen - 1; if ((vlen > 0) && (0 == strncasecmp(str, (char *)entry->entry, slen)) && (*(entry->entry + slen) == '=')) { if (!*tag) *tag = tag_new(); tag_add_item_n(*tag, itemType, (char *)(entry->entry + slen + 1), vlen); return 1; } return 0; } struct tag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block, struct tag * tag) { unsigned int i, j; FLAC__StreamMetadata_VorbisComment_Entry *comments; comments = block->data.vorbis_comment.comments; for (i = block->data.vorbis_comment.num_comments; i != 0; --i) { for (j = TAG_NUM_OF_ITEM_TYPES; j--;) { if (commentMatchesAddToTag(comments, j, &tag)) break; } comments++; } return tag; } void flac_metadata_common_cb(const FLAC__StreamMetadata * block, FlacData * data) { const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info); switch (block->type) { case FLAC__METADATA_TYPE_STREAMINFO: data->audio_format.bits = (int8_t)si->bits_per_sample; data->audio_format.sample_rate = si->sample_rate; data->audio_format.channels = (int8_t)si->channels; data->total_time = ((float)si->total_samples) / (si->sample_rate); break; case FLAC__METADATA_TYPE_VORBIS_COMMENT: flacParseReplayGain(block, data); default: break; } } void flac_error_common_cb(const char *plugin, const FLAC__StreamDecoderErrorStatus status, mpd_unused FlacData * data) { if (decoder_get_command(data->decoder) == DECODE_COMMAND_STOP) return; switch (status) { case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC: ERROR("%s lost sync\n", plugin); break; case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER: ERROR("bad %s header\n", plugin); break; case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH: ERROR("%s crc mismatch\n", plugin); break; default: ERROR("unknown %s error\n", plugin); } } static void flac_convert_stereo16(int16_t *dest, const FLAC__int32 * const buf[], unsigned int position, unsigned int end) { for (; position < end; ++position) { *dest++ = buf[0][position]; *dest++ = buf[1][position]; } } static void flac_convert_16(int16_t *dest, unsigned int num_channels, const FLAC__int32 * const buf[], unsigned int position, unsigned int end) { unsigned int c_chan; for (; position < end; ++position) for (c_chan = 0; c_chan < num_channels; c_chan++) *dest++ = buf[c_chan][position]; } /** * Note: this function also handles 24 bit files! */ static void flac_convert_32(int32_t *dest, unsigned int num_channels, const FLAC__int32 * const buf[], unsigned int position, unsigned int end) { unsigned int c_chan; for (; position < end; ++position) for (c_chan = 0; c_chan < num_channels; c_chan++) *dest++ = buf[c_chan][position]; } static void flac_convert_8(int8_t *dest, unsigned int num_channels, const FLAC__int32 * const buf[], unsigned int position, unsigned int end) { unsigned int c_chan; for (; position < end; ++position) for (c_chan = 0; c_chan < num_channels; c_chan++) *dest++ = buf[c_chan][position]; } static void flac_convert(unsigned char *dest, unsigned int num_channels, unsigned int bytes_per_sample, const FLAC__int32 * const buf[], unsigned int position, unsigned int end) { switch (bytes_per_sample) { case 2: if (num_channels == 2) flac_convert_stereo16((int16_t*)dest, buf, position, end); else flac_convert_16((int16_t*)dest, num_channels, buf, position, end); break; case 4: flac_convert_32((int32_t*)dest, num_channels, buf, position, end); break; case 1: flac_convert_8((int8_t*)dest, num_channels, buf, position, end); break; } } FLAC__StreamDecoderWriteStatus flac_common_write(FlacData *data, const FLAC__Frame * frame, const FLAC__int32 *const buf[]) { unsigned int c_samp; const unsigned int num_channels = frame->header.channels; const unsigned int bytes_per_sample = audio_format_sample_size(&data->audio_format); const unsigned int bytes_per_channel = bytes_per_sample * frame->header.channels; const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel; unsigned int num_samples; enum decoder_command cmd; if (bytes_per_sample != 1 && bytes_per_sample != 2 && bytes_per_sample != 4) /* exotic unsupported bit rate */ return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; for (c_samp = 0; c_samp < frame->header.blocksize; c_samp += num_samples) { num_samples = frame->header.blocksize - c_samp; if (num_samples > max_samples) num_samples = max_samples; flac_convert(data->chunk, num_channels, bytes_per_sample, buf, c_samp, c_samp + num_samples); cmd = decoder_data(data->decoder, data->inStream, 1, data->chunk, num_samples * bytes_per_channel, data->time, data->bitRate, data->replayGainInfo); switch (cmd) { case DECODE_COMMAND_NONE: case DECODE_COMMAND_START: break; case DECODE_COMMAND_STOP: return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; case DECODE_COMMAND_SEEK: return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } } return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; }