gstreamer/gst-libs/gst/audio/audio.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "audio.h"
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#include "multichannel-enumtypes.h"
#include <gst/gststructure.h>
/**
* SECTION:gstaudio
* @short_description: Support library for audio elements
*
* This library contains some helper functions for audio elements.
*/
/**
* gst_audio_frame_byte_size:
* @pad: the #GstPad to get the caps from
*
* Calculate byte size of an audio frame.
*
* Returns: the byte size, or 0 if there was an error
*/
int
gst_audio_frame_byte_size (GstPad * pad)
{
/* FIXME: this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
*/
int width = 0;
int channels = 0;
const GstCaps *caps = NULL;
GstStructure *structure;
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL) {
/* ERROR: could not get caps of pad */
g_warning ("gstaudio: could not get caps of pad %s:%s\n",
GST_DEBUG_PAD_NAME (pad));
return 0;
}
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "width", &width);
gst_structure_get_int (structure, "channels", &channels);
return (width / 8) * channels;
}
/**
* gst_audio_frame_length:
* @pad: the #GstPad to get the caps from
* @buf: the #GstBuffer
*
* Calculate length of buffer in frames.
*
* Returns: 0 if there's an error, or the number of frames if everything's ok
*/
long
gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
{
/* FIXME: this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
*/
int frame_byte_size = 0;
frame_byte_size = gst_audio_frame_byte_size (pad);
if (frame_byte_size == 0)
/* error */
return 0;
/* FIXME: this function assumes the buffer size to be a whole multiple
* of the frame byte size
*/
return GST_BUFFER_SIZE (buf) / frame_byte_size;
}
/**
* gst_audio_duration_from_pad_buffer:
* @pad: the #GstPad to get the caps from
* @buf: the #GstBuffer
*
* Calculate length in nanoseconds of audio buffer @buf based on capabilities of
* @pad.
*
* Return: the length.
*/
GstClockTime
gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf)
{
long bytes = 0;
int width = 0;
int channels = 0;
int rate = 0;
GstClockTime length;
const GstCaps *caps = NULL;
GstStructure *structure;
g_assert (GST_IS_BUFFER (buf));
/* get caps of pad */
caps = GST_PAD_CAPS (pad);
if (caps == NULL) {
/* ERROR: could not get caps of pad */
g_warning ("gstaudio: could not get caps of pad %s:%s\n",
GST_DEBUG_PAD_NAME (pad));
length = GST_CLOCK_TIME_NONE;
} else {
structure = gst_caps_get_structure (caps, 0);
bytes = GST_BUFFER_SIZE (buf);
gst_structure_get_int (structure, "width", &width);
gst_structure_get_int (structure, "channels", &channels);
gst_structure_get_int (structure, "rate", &rate);
g_assert (bytes != 0);
g_assert (width != 0);
g_assert (channels != 0);
g_assert (rate != 0);
length = (bytes * 8 * GST_SECOND) / (rate * channels * width);
}
return length;
}
/**
* gst_audio_is_buffer_framed:
* @pad: the #GstPad to get the caps from
* @buf: the #GstBuffer
*
* Check if the buffer size is a whole multiple of the frame size.
*
* Returns: %TRUE if buffer size is multiple.
*/
gboolean
gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
{
if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
return TRUE;
else
return FALSE;
}
/* _getcaps helper functions
* sets structure fields to default for audio type
* flag determines which structure fields to set to default
* keep these functions in sync with the templates in audio.h
*/
/* private helper function
* sets a list on the structure
* pass in structure, fieldname for the list, type of the list values,
* number of list values, and each of the values, terminating with NULL
*/
static void
_gst_audio_structure_set_list (GstStructure * structure,
const gchar * fieldname, GType type, int number, ...)
{
va_list varargs;
GValue value = { 0 };
GArray *array;
int j;
g_return_if_fail (structure != NULL);
g_value_init (&value, GST_TYPE_LIST);
array = g_value_peek_pointer (&value);
va_start (varargs, number);
for (j = 0; j < number; ++j) {
int i;
gboolean b;
GValue list_value = { 0 };
switch (type) {
case G_TYPE_INT:
i = va_arg (varargs, int);
g_value_init (&list_value, G_TYPE_INT);
g_value_set_int (&list_value, i);
break;
case G_TYPE_BOOLEAN:
b = va_arg (varargs, gboolean);
g_value_init (&list_value, G_TYPE_BOOLEAN);
g_value_set_boolean (&list_value, b);
break;
default:
g_warning
("_gst_audio_structure_set_list: LIST of given type not implemented.");
}
g_array_append_val (array, list_value);
}
gst_structure_set_value (structure, fieldname, &value);
va_end (varargs);
}
/**
* gst_audio_structure_set_int:
* @structure: a #GstStructure
* @flag: a set of #GstAudioFieldFlag
*
* Do not use anymore.
* @Deprecated: use gst_structure_set()
*/
void
gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
{
/* was added here:
* http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/audio/audio.c?r1=1.16&r2=1.17
* but it is not used
*/
if (flag & GST_AUDIO_FIELD_RATE)
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
NULL);
if (flag & GST_AUDIO_FIELD_CHANNELS)
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
NULL);
if (flag & GST_AUDIO_FIELD_ENDIANNESS)
_gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
if (flag & GST_AUDIO_FIELD_WIDTH)
_gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
NULL);
if (flag & GST_AUDIO_FIELD_DEPTH)
gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
if (flag & GST_AUDIO_FIELD_SIGNED)
_gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
FALSE, NULL);
}