2 files changed, 100 insertions, 1 deletions
diff --git a/libavcodec/audioconvert.c b/libavcodec/audioconvert.c
index 4c021219f0..6f00af49f9 100644
--- a/libavcodec/audioconvert.c
+++ b/libavcodec/audioconvert.c
@@ -25,7 +25,49 @@
  * @author Michael Niedermayer <michaelni@gmx.at>
  */
 
-#include "avcodec.h"
+#include "audioconvert.h"
+
+typedef struct SampleFmtInfo {
+    const char *name;
+    int bits;
+} SampleFmtInfo;
+
+/** this table gives more information about formats */
+static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = {
+    [SAMPLE_FMT_U8]  = { .name = "u8",  .bits = 8 },
+    [SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
+    [SAMPLE_FMT_S24] = { .name = "s24", .bits = 24 },
+    [SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
+    [SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 }
+};
+
+const char *avcodec_get_sample_fmt_name(int sample_fmt)
+{
+    if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB)
+        return NULL;
+    return sample_fmt_info[sample_fmt].name;
+}
+
+enum SampleFormat avcodec_get_sample_fmt(const char* name)
+{
+    int i;
+
+    for (i=0; i < SAMPLE_FMT_NB; i++)
+        if (!strcmp(sample_fmt_info[i].name, name))
+            return i;
+    return SAMPLE_FMT_NONE;
+}
+
+void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt)
+{
+    /* print header */
+    if (sample_fmt < 0)
+        snprintf (buf, buf_size, "name  " " depth");
+    else if (sample_fmt < SAMPLE_FMT_NB) {
+        SampleFmtInfo info= sample_fmt_info[sample_fmt];
+        snprintf (buf, buf_size, "%-6s" "   %2d ", info.name, info.bits);
+    }
+}
 
 int av_audio_convert(void *maybe_dspcontext_or_something_av_convert_specific,
                      void *out[6], int out_stride[6], enum SampleFormat out_fmt,
diff --git a/libavcodec/audioconvert.h b/libavcodec/audioconvert.h
new file mode 100644
index 0000000000..210cc87716
--- /dev/null
+++ b/libavcodec/audioconvert.h
@@ -0,0 +1,57 @@
+/*
+ * audio conversion
+ * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (c) 2008 Peter Ross
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef FFMPEG_AUDIOCONVERT_H
+#define FFMPEG_AUDIOCONVERT_H
+
+/**
+ * @file audioconvert.h
+ * Audio format conversion routines
+ */
+
+
+#include "avcodec.h"
+
+
+/**
+ * Generate string corresponding to the sample format with
+ * number sample_fmt, or a header if sample_fmt is negative.
+ *
+ * @param[in] buf the buffer where to write the string
+ * @param[in] buf_size the size of buf
+ * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
+ * a negative value to print the corresponding header.
+ * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
+ */
+void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
+
+/**
+ * @return NULL on error
+ */
+const char *avcodec_get_sample_fmt_name(int sample_fmt);
+
+/**
+ * @return SAMPLE_FMT_NONE on error
+ */
+enum SampleFormat avcodec_get_sample_fmt(const char* name);
+
+#endif /* FFMPEG_AUDIOCONVERT_H */