mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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12bac12e5c
Original commit message from CVS: * added plugin_desc structures to libs, which makes their locations cached in the registry. this speeds plugin loading considerably, especially on uninstalled versions. * put the lib path before all others, for speed reasons. * some fixes to adder's caps. * added linefeeds (\n) to GST_DEBUG strings to match GST_INFO behavior. this is more sane. all code will need to be converted. i think some perl can do this.
166 lines
4.1 KiB
C
166 lines
4.1 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <gst/audio/audio.h>
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int
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gst_audio_frame_byte_size (GstPad* pad)
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{
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/* calculate byte size of an audio frame
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* this should be moved closer to the gstreamer core
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* and be implemented for every mime type IMO
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* returns 0 if there's an error, or the byte size if everything's ok
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*/
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int width = 0;
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int channels = 0;
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GstCaps *caps = NULL;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL)
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/* ERROR: could not get caps of pad */
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return 0;
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width = gst_caps_get_int (caps, "width");
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channels = gst_caps_get_int (caps, "channels");
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return (width / 8) * channels;
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}
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long
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gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
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/* calculate length of buffer in frames
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* this should be moved closer to the gstreamer core
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* and be implemented for every mime type IMO
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* returns 0 if there's an error, or the number of frames if everything's ok
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*/
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{
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int frame_byte_size = 0;
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frame_byte_size = gst_audio_frame_byte_size (pad);
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if (frame_byte_size == 0)
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/* error */
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return 0;
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/* FIXME: this function assumes the buffer size to be a whole multiple
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* of the frame byte size
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*/
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return GST_BUFFER_SIZE (buf) / frame_byte_size;
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}
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long
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gst_audio_frame_rate (GstPad *pad)
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/*
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* calculate frame rate (based on caps of pad)
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* returns 0 if failed, rate if success
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*/
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{
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GstCaps *caps = NULL;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL)
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/* ERROR: could not get caps of pad */
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return 0;
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else
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return gst_caps_get_int (caps, "rate");
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}
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double
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gst_audio_length (GstPad* pad, GstBuffer* buf)
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{
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/* calculate length in seconds
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* of audio buffer buf
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* based on capabilities of pad
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*/
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long bytes = 0;
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int width = 0;
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int channels = 0;
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long rate = 0L;
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double length;
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GstCaps *caps = NULL;
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/* get caps of pad */
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caps = GST_PAD_CAPS (pad);
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if (caps == NULL)
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{
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/* ERROR: could not get caps of pad */
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length = 0.0;
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}
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else
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{
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bytes = GST_BUFFER_SIZE (buf);
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width = gst_caps_get_int (caps, "width");
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channels = gst_caps_get_int (caps, "channels");
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rate = gst_caps_get_int (caps, "rate");
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length = (bytes * 8.0) / (double) (rate * channels * width);
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}
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return length;
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}
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long
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gst_audio_highest_sample_value (GstPad* pad)
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/* calculate highest possible sample value
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* based on capabilities of pad
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*/
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{
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gboolean is_signed = FALSE;
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gint width = 0;
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GstCaps *caps = NULL;
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caps = GST_PAD_CAPS (pad);
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/* FIXME : Please change this to a better warning method ! */
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if (caps == NULL)
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printf ("WARNING: gstaudio: could not get caps of pad !\n");
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width = gst_caps_get_int (caps, "width");
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is_signed = gst_caps_get_boolean (caps, "signed");
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if (is_signed) --width;
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/* example : 16 bit, signed : samples between -32768 and 32767 */
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return ((long) (1 << width));
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}
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gboolean
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gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
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/* check if the buffer size is a whole multiple of the frame size */
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{
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if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
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return TRUE;
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else
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return FALSE;
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}
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static gboolean
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plugin_init (GModule *module, GstPlugin *plugin)
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{
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gst_plugin_set_longname (plugin, "Convenience routines for audio plugins");
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return TRUE;
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}
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GstPluginDesc plugin_desc = {
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GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"gstaudio",
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plugin_init
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};
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