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authorMax Kellermann <max@duempel.org>2008-10-26 11:29:25 +0100
committerMax Kellermann <max@duempel.org>2008-10-26 11:29:25 +0100
commite11355f47d545fe523b019481415b1347aecd4bd (patch)
treef178cd838be280d0517dc0e5910c36cb96a2a80e /src/decoder/audiofile_plugin.c
parentcbc71191f0ed75c5fafad5c387f009c2139a7bed (diff)
downloadmpd-e11355f47d545fe523b019481415b1347aecd4bd.tar.gz
mpd-e11355f47d545fe523b019481415b1347aecd4bd.tar.xz
mpd-e11355f47d545fe523b019481415b1347aecd4bd.zip
renamed src/inputPlugins/ to src/decoder/
These plugins are not input plugins, they are decoder plugins. No CamelCase in the directory name.
Diffstat (limited to 'src/decoder/audiofile_plugin.c')
-rw-r--r--src/decoder/audiofile_plugin.c147
1 files changed, 147 insertions, 0 deletions
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
new file mode 100644
index 000000000..99846e853
--- /dev/null
+++ b/src/decoder/audiofile_plugin.c
@@ -0,0 +1,147 @@
+/* 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"
+#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.sample_rate =
+ (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.sample_rate);
+
+ 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.sample_rate;
+ 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.sample_rate,
+ 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,
+};