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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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95011fd7e8
Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
213 lines
5.8 KiB
C
213 lines
5.8 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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstgsmenc.h"
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extern GstPadTemplate *gsmenc_src_template, *gsmenc_sink_template;
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/* elementfactory information */
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GstElementDetails gst_gsmenc_details = {
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"gsm audio encoder",
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"Codec/Audio/Encoder",
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"LGPL",
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".gsm",
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VERSION,
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"Wim Taymans <wim.taymans@chello.be>",
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"(C) 2000",
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};
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/* GSMEnc signals and args */
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enum {
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FRAME_ENCODED,
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/* FILL ME */
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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/* FILL ME */
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};
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static void gst_gsmenc_class_init (GstGSMEnc *klass);
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static void gst_gsmenc_init (GstGSMEnc *gsmenc);
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static void gst_gsmenc_chain (GstPad *pad,GstBuffer *buf);
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static GstPadLinkReturn gst_gsmenc_sinkconnect (GstPad *pad, GstCaps *caps);
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static GstElementClass *parent_class = NULL;
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static guint gst_gsmenc_signals[LAST_SIGNAL] = { 0 };
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GType
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gst_gsmenc_get_type (void)
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{
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static GType gsmenc_type = 0;
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if (!gsmenc_type) {
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static const GTypeInfo gsmenc_info = {
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sizeof (GstGSMEncClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_gsmenc_class_init,
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NULL,
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NULL,
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sizeof (GstGSMEnc),
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0,
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(GInstanceInitFunc) gst_gsmenc_init,
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};
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gsmenc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
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}
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return gsmenc_type;
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}
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static void
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gst_gsmenc_class_init (GstGSMEnc *klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass*) klass;
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gstelement_class = (GstElementClass*) klass;
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gst_gsmenc_signals[FRAME_ENCODED] =
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g_signal_new ("frame_encoded", G_TYPE_FROM_CLASS(klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstGSMEncClass, frame_encoded), NULL, NULL,
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g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
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}
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static void
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gst_gsmenc_init (GstGSMEnc *gsmenc)
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{
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/* create the sink and src pads */
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gsmenc->sinkpad = gst_pad_new_from_template (gsmenc_sink_template, "sink");
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gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
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gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
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gst_pad_set_link_function (gsmenc->sinkpad, gst_gsmenc_sinkconnect);
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gsmenc->srcpad = gst_pad_new_from_template (gsmenc_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
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gsmenc->state = gsm_create ();
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gsmenc->bufsize = 0;
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gsmenc->next_ts = 0;
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gsmenc->rate = 8000;
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}
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static GstPadLinkReturn
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gst_gsmenc_sinkconnect (GstPad *pad, GstCaps *caps)
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{
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GstGSMEnc *gsmenc;
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gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
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if (!GST_CAPS_IS_FIXED (caps))
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return GST_PAD_LINK_DELAYED;
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gst_caps_get_int (caps, "rate", &gsmenc->rate);
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if (gst_pad_try_set_caps (gsmenc->srcpad, GST_CAPS_NEW (
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"gsm_gsm",
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"audio/x-gsm",
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"rate", GST_PROPS_INT (gsmenc->rate),
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"channels", GST_PROPS_INT (1)
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)) > 0)
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{
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return GST_PAD_LINK_OK;
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}
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return GST_PAD_LINK_REFUSED;
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}
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static void
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gst_gsmenc_chain (GstPad *pad, GstBuffer *buf)
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{
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GstGSMEnc *gsmenc;
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gsm_signal *data;
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guint size;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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gsmenc = GST_GSMENC (GST_OBJECT_PARENT (pad));
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if (!GST_PAD_CAPS (gsmenc->srcpad)) {
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gst_pad_try_set_caps (gsmenc->srcpad,
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GST_CAPS_NEW (
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"gsm_enc",
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"audio/x-gsm",
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"rate", GST_PROPS_INT (gsmenc->rate),
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"channels", GST_PROPS_INT (1)
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));
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}
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data = (gsm_signal*) GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf) / sizeof (gsm_signal);
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if (gsmenc->bufsize && (gsmenc->bufsize + size >= 160)) {
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GstBuffer *outbuf;
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memcpy (gsmenc->buffer + gsmenc->bufsize, data, (160 - gsmenc->bufsize) * sizeof (gsm_signal));
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outbuf = gst_buffer_new ();
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GST_BUFFER_DATA (outbuf) = g_malloc (33 * sizeof (gsm_byte));
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GST_BUFFER_SIZE (outbuf) = 33 * sizeof (gsm_byte);
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gsm_encode (gsmenc->state, gsmenc->buffer, (gsm_byte *) GST_BUFFER_DATA (outbuf));
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GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
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gst_pad_push (gsmenc->srcpad, outbuf);
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gsmenc->next_ts += (160.0 / gsmenc->rate) * 1000000;
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size -= (160 - gsmenc->bufsize);
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data += (160 - gsmenc->bufsize);
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gsmenc->bufsize = 0;
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}
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while (size >= 160) {
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GstBuffer *outbuf;
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outbuf = gst_buffer_new ();
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GST_BUFFER_DATA (outbuf) = g_malloc (33 * sizeof (gsm_byte));
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GST_BUFFER_SIZE (outbuf) = 33 * sizeof (gsm_byte);
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gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
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GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
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gst_pad_push (gsmenc->srcpad, outbuf);
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gsmenc->next_ts += (160 / gsmenc->rate) * GST_SECOND;
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size -= 160;
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data += 160;
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}
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if (size) {
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memcpy (gsmenc->buffer + gsmenc->bufsize, data, size * sizeof (gsm_signal));
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gsmenc->bufsize += size;
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}
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gst_buffer_unref(buf);
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}
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