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authorLeif Johnson <leif@ambient.2y.net>2003-07-19 23:47:42 +0000
committerLeif Johnson <leif@ambient.2y.net>2003-07-19 23:47:42 +0000
commit6d6150c0529d8f5af1857543c5d94902eab7237c (patch)
tree11813ead7a82bbf5dc1ab36ce6c4d3ea1303bd74 /gst-libs/gst/audio/audio.h
parentf6830d4ad15f488c6a61ebccf4f09529fdf5ebd3 (diff)
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+ the last of the float caps changes ... these are a bit more pervasive
Original commit message from CVS: + the last of the float caps changes ... these are a bit more pervasive
Diffstat (limited to 'gst-libs/gst/audio/audio.h')
-rw-r--r--gst-libs/gst/audio/audio.h152
1 files changed, 86 insertions, 66 deletions
diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
index a737e468..c22052f5 100644
--- a/gst-libs/gst/audio/audio.h
+++ b/gst-libs/gst/audio/audio.h
@@ -22,78 +22,97 @@
#include <gst/audio/audioclock.h>
+G_BEGIN_DECLS
+
/* For people that are looking at this source: the purpose of these defines is
* to make GstCaps a bit easier, in that you don't have to know all of the
* properties that need to be defined. you can just use these macros. currently
* (8/01) the only plugins that use these are the passthrough, speed, volume,
- * adder, and [de]interleave plugins.
- * These are for convenience only, and do not specify the 'limits' of
- * GStreamer. you might also use these definitions as a
+ * adder, and [de]interleave plugins. These are for convenience only, and do not
+ * specify the 'limits' of GStreamer. you might also use these definitions as a
* base for making your own caps, if need be.
*
- * For example, to make a source pad that can output mono streams of either
- * float or int:
-
- template = gst_pad_template_new
- ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
- gst_caps_append(gst_caps_new ("sink_int", "audio/raw",
- GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
- gst_caps_new ("sink_float", "audio/raw",
- GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS)),
- NULL);
-
- srcpad = gst_pad_new_from_template(template,"src");
-
- * Andy Wingo, 18 August 2001
+ * For example, to make a source pad that can output streams of either mono
+ * float or any channel int:
+ *
+ * template = gst_pad_template_new
+ * ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+ * gst_caps_append(gst_caps_new ("sink_int", "audio/x-raw-int",
+ * GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
+ * gst_caps_new ("sink_float", "audio/x-raw-float",
+ * GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
+ * NULL);
+ *
+ * sinkpad = gst_pad_new_from_template(template, "sink");
+ *
+ * Andy Wingo, 18 August 2001
* Thomas, 6 September 2002 */
-/* a few useful defines for arbitrary limits */
-#define GST_AUDIO_MIN_RATE 4000
-#define GST_AUDIO_MAX_RATE 96000
-#define GST_AUDIO_DEF_RATE 44100
+#define GST_AUDIO_DEF_RATE 44100
#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "endianness", GST_PROPS_INT (G_BYTE_ORDER),\
- "signed", GST_PROPS_LIST (\
- GST_PROPS_BOOLEAN (TRUE),\
- GST_PROPS_BOOLEAN (FALSE)\
- ),\
- "width", GST_PROPS_LIST (GST_PROPS_INT (8), \
- GST_PROPS_INT (16)), \
- "depth", GST_PROPS_LIST (GST_PROPS_INT (8), \
- GST_PROPS_INT (16)),\
- "rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \
- GST_AUDIO_MAX_RATE),\
- "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- NULL)
+ gst_props_new (\
+ "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ "endianness", GST_PROPS_LIST (\
+ GST_PROPS_INT (G_LITTLE_ENDIAN),\
+ GST_PROPS_INT (G_BIG_ENDIAN)\
+ ),\
+ "width", GST_PROPS_LIST (\
+ GST_PROPS_INT (8),\
+ GST_PROPS_INT (16),\
+ GST_PROPS_INT (32)\
+ ),\
+ "depth", GST_PROPS_INT_RANGE (1, 32),\
+ "signed", GST_PROPS_LIST (\
+ GST_PROPS_BOOLEAN (TRUE),\
+ GST_PROPS_BOOLEAN (FALSE)\
+ ),\
+ NULL)
#define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "endianness", GST_PROPS_INT (G_BYTE_ORDER),\
- "signed", GST_PROPS_LIST (\
- GST_PROPS_BOOLEAN (TRUE),\
- GST_PROPS_BOOLEAN (FALSE)\
- ),\
- "width", GST_PROPS_LIST (GST_PROPS_INT (8), \
- GST_PROPS_INT (16)),\
- "depth", GST_PROPS_LIST (GST_PROPS_INT (8), \
- GST_PROPS_INT (16)),\
- "rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \
- GST_AUDIO_MAX_RATE),\
- "channels", GST_PROPS_INT (1),\
- NULL)
-
-#define GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "depth", GST_PROPS_INT (32),\
- "endianness", GST_PROPS_INT (G_BYTE_ORDER),\
- "intercept", GST_PROPS_FLOAT (0.0),\
- "slope", GST_PROPS_FLOAT (1.0),\
- "rate", GST_PROPS_INT_RANGE (GST_AUDIO_MIN_RATE, \
- GST_AUDIO_MAX_RATE),\
- "channels", GST_PROPS_INT (1),\
- NULL)
+ gst_props_new (\
+ "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ "channels", GST_PROPS_INT (1),\
+ "endianness", GST_PROPS_LIST (\
+ GST_PROPS_INT (G_LITTLE_ENDIAN),\
+ GST_PROPS_INT (G_BIG_ENDIAN)\
+ ),\
+ "width", GST_PROPS_LIST (\
+ GST_PROPS_INT (8),\
+ GST_PROPS_INT (16),\
+ GST_PROPS_INT (32)\
+ ),\
+ "depth", GST_PROPS_INT_RANGE (1, 32),\
+ "signed", GST_PROPS_LIST (\
+ GST_PROPS_BOOLEAN (TRUE),\
+ GST_PROPS_BOOLEAN (FALSE)\
+ ),\
+ NULL)
+
+#define GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS \
+ gst_props_new (\
+ "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ "endianness", GST_PROPS_LIST (\
+ GST_PROPS_INT (G_LITTLE_ENDIAN),\
+ GST_PROPS_INT (G_BIG_ENDIAN)\
+ ),\
+ "width", GST_PROPS_LIST (\
+ GST_PROPS_INT (32),\
+ GST_PROPS_INT (64)\
+ ),\
+ "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ NULL)
+
+#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS \
+ gst_props_new (\
+ "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ "channels", GST_PROPS_INT (1),\
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),\
+ "width", GST_PROPS_INT (32),\
+ "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
+ NULL)
/*
* this library defines and implements some helper functions for audio
@@ -101,21 +120,22 @@
*/
/* get byte size of audio frame (based on caps of pad */
-int gst_audio_frame_byte_size (GstPad* pad);
+int gst_audio_frame_byte_size (GstPad* pad);
/* get length in frames of buffer */
-long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
+long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
/* get frame rate based on caps */
-long gst_audio_frame_rate (GstPad *pad);
+long gst_audio_frame_rate (GstPad *pad);
/* calculate length in seconds of audio buffer buf based on caps of pad */
-double gst_audio_length (GstPad* pad, GstBuffer* buf);
+double gst_audio_length (GstPad* pad, GstBuffer* buf);
/* calculate highest possible sample value based on capabilities of pad */
-long gst_audio_highest_sample_value (GstPad* pad);
+long gst_audio_highest_sample_value (GstPad* pad);
/* check if the buffer size is a whole multiple of the frame size */
-gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
+gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
+G_END_DECLS