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author | J. Alexander Treuman <jat@spatialrift.net> | 2007-05-28 15:50:45 +0000 |
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committer | J. Alexander Treuman <jat@spatialrift.net> | 2007-05-28 15:50:45 +0000 |
commit | 4e05a161e54fe05902b99eff521aa0759b102f05 (patch) | |
tree | 9014ab8ce74e4fb2f0115c5518c62879bc269324 /trunk/src/inputPlugins/audiofile_plugin.c | |
parent | e45bc035931b2c9ef13b85f98a3d4833a8dec8a9 (diff) | |
download | mpd-4e05a161e54fe05902b99eff521aa0759b102f05.tar.gz mpd-4e05a161e54fe05902b99eff521aa0759b102f05.tar.xz mpd-4e05a161e54fe05902b99eff521aa0759b102f05.zip |
Making branch for 0.13.0 fixes.
git-svn-id: https://svn.musicpd.org/mpd/branches/branch-0.13.0-fixes@6330 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'trunk/src/inputPlugins/audiofile_plugin.c')
-rw-r--r-- | trunk/src/inputPlugins/audiofile_plugin.c | 188 |
1 files changed, 0 insertions, 188 deletions
diff --git a/trunk/src/inputPlugins/audiofile_plugin.c b/trunk/src/inputPlugins/audiofile_plugin.c deleted file mode 100644 index 35fb48b8a..000000000 --- a/trunk/src/inputPlugins/audiofile_plugin.c +++ /dev/null @@ -1,188 +0,0 @@ -/* 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 */ |