audio: move function to convert

diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c
index 0f647c6..2753a62 100644 (file)
--- a/gst-libs/gst/audio/audio.c
+++ b/gst-libs/gst/audio/audio.c
@@ -411,6 +411,99 @@ gst_audio_info_to_caps (GstAudioInfo * info)
 }
 
 /**
+ * gst_audio_format_convert:
+ * @info: a #GstAudioInfo
+ * @src_format: #GstFormat of the @src_value
+ * @src_value: value to convert
+ * @dest_format: #GstFormat of the @dest_value
+ * @dest_value: pointer to destination value
+ *
+ * Converts among various #GstFormat types.  This function handles
+ * GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT.  For
+ * raw audio, GST_FORMAT_DEFAULT corresponds to audio frames.  This
+ * function can be used to handle pad queries of the type GST_QUERY_CONVERT.
+ *
+ * Returns: TRUE if the conversion was successful.
+ */
+gboolean
+gst_audio_info_convert (GstAudioInfo * info,
+    GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
+{
+  gboolean res = TRUE;
+  gint bpf, rate;
+
+  GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
+      src_val, gst_format_get_name (src_fmt), src_fmt,
+      gst_format_get_name (dest_fmt), dest_fmt);
+
+  if (src_fmt == dest_fmt || src_val == -1) {
+    *dest_val = src_val;
+    goto done;
+  }
+
+  /* get important info */
+  bpf = GST_AUDIO_INFO_BPF (info);
+  rate = GST_AUDIO_INFO_RATE (info);
+
+  if (bpf == 0 || rate == 0) {
+    GST_DEBUG ("no rate or bpf configured");
+    res = FALSE;
+    goto done;
+  }
+
+  switch (src_fmt) {
+    case GST_FORMAT_BYTES:
+      switch (dest_fmt) {
+        case GST_FORMAT_TIME:
+          *dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val / bpf, rate);
+          break;
+        case GST_FORMAT_DEFAULT:
+          *dest_val = src_val / bpf;
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    case GST_FORMAT_DEFAULT:
+      switch (dest_fmt) {
+        case GST_FORMAT_TIME:
+          *dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val, rate);
+          break;
+        case GST_FORMAT_BYTES:
+          *dest_val = src_val * bpf;
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    case GST_FORMAT_TIME:
+      switch (dest_fmt) {
+        case GST_FORMAT_DEFAULT:
+          *dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
+          break;
+        case GST_FORMAT_BYTES:
+          *dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
+          *dest_val *= bpf;
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    default:
+      res = FALSE;
+      break;
+  }
+done:
+  GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, *dest_val);
+
+  return res;
+}
+
+
+/**
  * gst_audio_buffer_clip:
  * @buffer: The buffer to clip.
  * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which

diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
index e60312d..f5a3d1e 100644 (file)
--- a/gst-libs/gst/audio/audio.h
+++ b/gst-libs/gst/audio/audio.h
@@ -297,6 +297,11 @@ void         gst_audio_info_set_format  (GstAudioInfo *info, GstAudioFormat form
 gboolean     gst_audio_info_from_caps   (GstAudioInfo *info, const GstCaps *caps);
 GstCaps *    gst_audio_info_to_caps     (GstAudioInfo *info);
 
+gboolean     gst_audio_info_convert     (GstAudioInfo * info,
+                                         GstFormat src_fmt, gint64 src_val,
+                                         GstFormat dest_fmt, gint64 * dest_val);
+
+
 
 #define GST_AUDIO_RATE_RANGE "(int) [ 1, max ]"
 #define GST_AUDIO_CHANNELS_RANGE "(int) [ 1, max ]"

diff --git a/gst-libs/gst/audio/gstringbuffer.c b/gst-libs/gst/audio/gstringbuffer.c
index cb18f76..53dfa71 100644 (file)
--- a/gst-libs/gst/audio/gstringbuffer.c
+++ b/gst-libs/gst/audio/gstringbuffer.c
@@ -316,79 +316,14 @@ gboolean
 gst_ring_buffer_convert (GstRingBuffer * buf,
     GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
 {
-  gboolean res = TRUE;
-  gint bpf, rate;
-
-  GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
-      src_val, gst_format_get_name (src_fmt), src_fmt,
-      gst_format_get_name (dest_fmt), dest_fmt);
-
-  if (src_fmt == dest_fmt || src_val == -1) {
-    *dest_val = src_val;
-    goto done;
-  }
+  gboolean res;
 
-  /* get important info */
   GST_OBJECT_LOCK (buf);
-  bpf = GST_AUDIO_INFO_BPF (&buf->spec.info);
-  rate = GST_AUDIO_INFO_RATE (&buf->spec.info);
+  res =
+      gst_audio_info_convert (&buf->spec.info, src_fmt, src_val, dest_fmt,
+      dest_val);
   GST_OBJECT_UNLOCK (buf);
 
-  if (bpf == 0 || rate == 0) {
-    GST_DEBUG ("no rate or bpf configured");
-    res = FALSE;
-    goto done;
-  }
-
-  switch (src_fmt) {
-    case GST_FORMAT_BYTES:
-      switch (dest_fmt) {
-        case GST_FORMAT_TIME:
-          *dest_val = gst_util_uint64_scale_int (src_val / bpf, GST_SECOND,
-              rate);
-          break;
-        case GST_FORMAT_DEFAULT:
-          *dest_val = src_val / bpf;
-          break;
-        default:
-          res = FALSE;
-          break;
-      }
-      break;
-    case GST_FORMAT_DEFAULT:
-      switch (dest_fmt) {
-        case GST_FORMAT_TIME:
-          *dest_val = gst_util_uint64_scale_int (src_val, GST_SECOND, rate);
-          break;
-        case GST_FORMAT_BYTES:
-          *dest_val = src_val * bpf;
-          break;
-        default:
-          res = FALSE;
-          break;
-      }
-      break;
-    case GST_FORMAT_TIME:
-      switch (dest_fmt) {
-        case GST_FORMAT_DEFAULT:
-          *dest_val = gst_util_uint64_scale_int (src_val, rate, GST_SECOND);
-          break;
-        case GST_FORMAT_BYTES:
-          *dest_val = gst_util_uint64_scale_int (src_val, rate, GST_SECOND);
-          *dest_val *= bpf;
-          break;
-        default:
-          res = FALSE;
-          break;
-      }
-      break;
-    default:
-      res = FALSE;
-      break;
-  }
-done:
-  GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, *dest_val);
-
   return res;
 }