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9ee0b03986
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.
189 lines
4.8 KiB
C
189 lines
4.8 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 "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 -1 if there's an error (to avoid division by zero),
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* 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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{
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/* ERROR: could not get caps of pad */
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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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return 0;
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}
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gst_caps_get_int (caps, "width", &width);
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gst_caps_get_int (caps, "channels", &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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gint rate;
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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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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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return 0;
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}
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else {
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gst_caps_get_int (caps, "rate", &rate);
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return rate;
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}
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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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int rate = 0;
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double length;
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GstCaps *caps = NULL;
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g_assert (GST_IS_BUFFER (buf));
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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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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (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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gst_caps_get_int (caps, "width", &width);
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gst_caps_get_int (caps, "channels", &channels);
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gst_caps_get_int (caps, "rate", &rate);
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g_assert (bytes != 0);
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g_assert (width != 0);
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g_assert (channels != 0);
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g_assert (rate != 0);
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length = (bytes * 8.0) / (double) (rate * channels * width);
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}
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/* g_print ("DEBUG: audio: returning length of %f\n", length); */
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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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if (caps == NULL)
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{
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g_warning ("gstaudio: could not get caps of pad %s:%s\n",
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GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
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}
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gst_caps_get_int (caps, "width", &width);
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gst_caps_get_boolean (caps, "signed", &is_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, "Support services 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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