/* 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 "../inputPlugin.h" #ifdef HAVE_AUDIOFILE #include "../utils.h" #include "../audio.h" #include "../log.h" #include "../pcm_utils.h" #include "../playerData.h" #include <stdio.h> #include <unistd.h> #include <stdlib.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <unistd.h> #include <audiofile.h> static int getAudiofileTotalTime(char *file) { int time; AFfilehandle af_fp = afOpenFile(file, "r", NULL); if (af_fp == AF_NULL_FILEHANDLE) { return -1; } time = (int) ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK) / afGetRate(af_fp, AF_DEFAULT_TRACK)); afCloseFile(af_fp); return time; } static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path) { int fs, frame_count; AFfilehandle af_fp; int bits; mpd_uint16 bitRate; struct stat st; 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); dc->audioFormat.bits = bits; dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK); dc->audioFormat.channels = afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat)); frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK); dc->totalTime = ((float)frame_count / (float)dc->audioFormat.sampleRate); bitRate = st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5; if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) { ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n", path, dc->audioFormat.bits); afCloseFile(af_fp); return -1; } fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1); dc->state = DECODE_STATE_DECODE; { int ret, eof = 0, current = 0; char chunk[CHUNK_SIZE]; while (!eof) { if (dc->seek) { clearOutputBuffer(cb); current = dc->seekWhere * dc->audioFormat.sampleRate; afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); dc->seek = 0; } ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, CHUNK_SIZE / fs); if (ret <= 0) eof = 1; else { current += ret; sendDataToOutputBuffer(cb, NULL, dc, 1, chunk, ret * fs, (float)current / (float)dc->audioFormat. sampleRate, bitRate, NULL); if (dc->stop) break; } } flushOutputBuffer(cb); /*if(dc->seek) { dc->seekError = 1; dc->seek = 0; } */ if (dc->stop) { dc->state = DECODE_STATE_STOP; dc->stop = 0; } else dc->state = DECODE_STATE_STOP; } afCloseFile(af_fp); return 0; } static MpdTag *audiofileTagDup(char *file) { MpdTag *ret = NULL; int time = getAudiofileTotalTime(file); if (time >= 0) { if (!ret) ret = newMpdTag(); ret->time = time; } else { DEBUG ("audiofileTagDup: Failed to get total song time from: %s\n", file); } return ret; } static char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL }; InputPlugin audiofilePlugin = { "audiofile", NULL, NULL, NULL, NULL, audiofile_decode, audiofileTagDup, INPUT_PLUGIN_STREAM_FILE, audiofileSuffixes, NULL }; #else InputPlugin audiofilePlugin; #endif /* HAVE_AUDIOFILE */