gstreamer/gst-libs/gst/audio/audio.c
Andy Wingo 2ff63e563b actually recurse into sndfile if we are able big ladspa cleanups, mainly to comply with the buffer-frames caps proper...
Original commit message from CVS:
* actually recurse into sndfile if we are able
* big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general
cleanups
- the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins)
you need to use a filtered connection, just like with buffer-frames
* big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit
simpler
* make the ossclock general, add it to gstaudio, and use it in sndfile as well

i need to update mimetypes, but that's coming soon. there are some other plugins that don't
support buffer-frames, i guess i need to get around to fixing them as well.
2003-07-16 16:08:13 +00:00

189 lines
4.8 KiB
C

/* 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.
*/
#include "audio.h"
int
gst_audio_frame_byte_size (GstPad* pad)
{
/* calculate byte size of an audio frame
* this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
* returns -1 if there's an error (to avoid division by zero),
* or the byte size if everything's ok
*/
int width = 0;
int channels = 0;
GstCaps *caps = NULL;
/* 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_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
return 0;
}
gst_caps_get_int (caps, "width", &width);
gst_caps_get_int (caps, "channels", &channels);
return (width / 8) * channels;
}
long
gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
/* calculate length of buffer in frames
* this should be moved closer to the gstreamer core
* and be implemented for every mime type IMO
* returns 0 if there's an error, or the number of frames if everything's ok
*/
{
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;
}
long
gst_audio_frame_rate (GstPad *pad)
/*
* calculate frame rate (based on caps of pad)
* returns 0 if failed, rate if success
*/
{
GstCaps *caps = NULL;
gint rate;
/* 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_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
return 0;
}
else {
gst_caps_get_int (caps, "rate", &rate);
return rate;
}
}
double
gst_audio_length (GstPad* pad, GstBuffer* buf)
{
/* calculate length in seconds
* of audio buffer buf
* based on capabilities of pad
*/
long bytes = 0;
int width = 0;
int channels = 0;
int rate = 0;
double length;
GstCaps *caps = NULL;
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_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
length = 0.0;
}
else
{
bytes = GST_BUFFER_SIZE (buf);
gst_caps_get_int (caps, "width", &width);
gst_caps_get_int (caps, "channels", &channels);
gst_caps_get_int (caps, "rate", &rate);
g_assert (bytes != 0);
g_assert (width != 0);
g_assert (channels != 0);
g_assert (rate != 0);
length = (bytes * 8.0) / (double) (rate * channels * width);
}
/* g_print ("DEBUG: audio: returning length of %f\n", length); */
return length;
}
long
gst_audio_highest_sample_value (GstPad* pad)
/* calculate highest possible sample value
* based on capabilities of pad
*/
{
gboolean is_signed = FALSE;
gint width = 0;
GstCaps *caps = NULL;
caps = GST_PAD_CAPS (pad);
if (caps == NULL)
{
g_warning ("gstaudio: could not get caps of pad %s:%s\n",
GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
}
gst_caps_get_int (caps, "width", &width);
gst_caps_get_boolean (caps, "signed", &is_signed);
if (is_signed) --width;
/* example : 16 bit, signed : samples between -32768 and 32767 */
return ((long) (1 << width));
}
gboolean
gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
/* check if the buffer size is a whole multiple of the frame size */
{
if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
return TRUE;
else
return FALSE;
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
gst_plugin_set_longname (plugin, "Support services for audio plugins");
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstaudio",
plugin_init
};