/* 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 * * libaudiofile (wave) support added 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 "../decoder_api.h" #ifdef HAVE_AUDIOFILE #include "../log.h" #include <sys/stat.h> #include <audiofile.h> /* pick 1020 since its devisible for 8,16,24, and 32-bit audio */ #define CHUNK_SIZE 1020 static int getAudiofileTotalTime(char *file) { int total_time; AFfilehandle af_fp = afOpenFile(file, "r", NULL); if (af_fp == AF_NULL_FILEHANDLE) { return -1; } total_time = (int) ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK) / afGetRate(af_fp, AF_DEFAULT_TRACK)); afCloseFile(af_fp); return total_time; } static int audiofile_decode(struct decoder * decoder, char *path) { int fs, frame_count; AFfilehandle af_fp; int bits; struct audio_format audio_format; float total_time; uint16_t bitRate; struct stat st; int ret, current = 0; char chunk[CHUNK_SIZE]; if (stat(path, &st) < 0) { ERROR("failed to stat: %s\n", path); return -1; } af_fp = afOpenFile(path, "r", NULL); if (af_fp == AF_NULL_FILEHANDLE) { ERROR("failed to open: %s\n", path); return -1; } afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); audio_format.bits = (uint8_t)bits; audio_format.sampleRate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); audio_format.channels = (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); total_time = ((float)frame_count / (float)audio_format.sampleRate); bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5); if (audio_format.bits != 8 && audio_format.bits != 16) { ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n", path, audio_format.bits); afCloseFile(af_fp); return -1; } fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); decoder_initialized(decoder, &audio_format, total_time); do { if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) { decoder_clear(decoder); current = decoder_seek_where(decoder) * audio_format.sampleRate; afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); decoder_command_finished(decoder); } ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, CHUNK_SIZE / fs); if (ret <= 0) break; current += ret; decoder_data(decoder, NULL, 1, chunk, ret * fs, (float)current / (float)audio_format.sampleRate, bitRate, NULL); } while (decoder_get_command(decoder) != DECODE_COMMAND_STOP); decoder_flush(decoder); afCloseFile(af_fp); return 0; } static struct tag *audiofileTagDup(char *file) { struct tag *ret = NULL; int total_time = getAudiofileTotalTime(file); if (total_time >= 0) { if (!ret) ret = tag_new(); ret->time = total_time; } else { DEBUG ("audiofileTagDup: Failed to get total song time from: %s\n", file); } return ret; } static const char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL }; struct decoder_plugin audiofilePlugin = { .name = "audiofile", .file_decode = audiofile_decode, .tag_dup = audiofileTagDup, .stream_types = INPUT_PLUGIN_STREAM_FILE, .suffixes = audiofileSuffixes, }; #else struct decoder_plugin audiofilePlugin; #endif /* HAVE_AUDIOFILE */