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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f8783d2cc6
Original commit message from CVS: * ext/directfb/dfbvideosink.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/libmms/gstmms.c: * ext/neon/gstneonhttpsrc.c: * ext/theora/theoradec.c: * gst/freeze/gstfreeze.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * sys/glsink/glimagesink.c: Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
211 lines
5.5 KiB
C
211 lines
5.5 KiB
C
/*
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* Farsight
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* GStreamer GSM encoder
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* Copyright (C) 2005 Philippe Khalaf <burger@speedy.org>
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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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GST_DEBUG_CATEGORY_STATIC (gsmenc_debug);
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#define GST_CAT_DEFAULT (gsmenc_debug)
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/* elementfactory information */
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static const GstElementDetails gst_gsmenc_details =
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GST_ELEMENT_DETAILS ("GSM audio encoder",
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"Codec/Encoder/Audio",
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"Encodes GSM audio",
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"Philippe Khalaf <burger@speedy.org>");
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/* GSMEnc signals and args */
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enum
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{
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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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{
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/* FILL ME */
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ARG_0
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};
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static void gst_gsmenc_base_init (gpointer g_class);
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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_finalize (GObject * object);
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static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
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static GstElementClass *parent_class = NULL;
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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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gst_gsmenc_base_init,
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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 =
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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 GstStaticPadTemplate gsmenc_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
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);
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static GstStaticPadTemplate gsmenc_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, "
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"width = (int) 16, "
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"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
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);
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static void
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gst_gsmenc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gsmenc_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gsmenc_src_template));
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gst_element_class_set_details (element_class, &gst_gsmenc_details);
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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_peek_parent (klass);
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gobject_class->finalize = gst_gsmenc_finalize;
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GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
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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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gint use_wav49;
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/* create the sink and src pads */
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gsmenc->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&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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gsmenc->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&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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/* turn on WAV49 handling */
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use_wav49 = 0;
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gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
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gsmenc->adapter = gst_adapter_new ();
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gsmenc->next_ts = 0;
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}
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static void
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gst_gsmenc_finalize (GObject * object)
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{
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GstGSMEnc *gsmenc;
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gsmenc = GST_GSMENC (object);
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g_object_unref (gsmenc->adapter);
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gsm_destroy (gsmenc->state);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstFlowReturn
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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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GstFlowReturn ret = GST_FLOW_OK;
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gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
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if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (gsmenc->adapter);
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}
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gst_adapter_push (gsmenc->adapter, buf);
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while (gst_adapter_available (gsmenc->adapter) >= 320) {
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GstBuffer *outbuf;
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outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
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GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
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GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
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gsmenc->next_ts += 20 * GST_MSECOND;
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/* encode 160 16-bit samples into 33 bytes */
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data = (gsm_signal *) gst_adapter_peek (gsmenc->adapter, 320);
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gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
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gst_adapter_flush (gsmenc->adapter, 320);
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad));
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GST_DEBUG_OBJECT (gsmenc, "Pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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ret = gst_pad_push (gsmenc->srcpad, outbuf);
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}
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gst_object_unref (gsmenc);
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return ret;
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}
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