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authorJ. Alexander Treuman <jat@spatialrift.net>2007-05-28 15:50:45 +0000
committerJ. Alexander Treuman <jat@spatialrift.net>2007-05-28 15:50:45 +0000
commit4e05a161e54fe05902b99eff521aa0759b102f05 (patch)
tree9014ab8ce74e4fb2f0115c5518c62879bc269324 /trunk/src/inputPlugins/audiofile_plugin.c
parente45bc035931b2c9ef13b85f98a3d4833a8dec8a9 (diff)
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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.c188
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 */