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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f48c4cbe42
Original commit message from CVS: * ext/amrnb/amrnbdec.c: * ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps): * ext/amrnb/amrnbparse.c: Update caps with audio/AMR. * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init), (gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain), (gst_rtpamrdec_change_state): * gst/rtp/gstrtpamrdec.h: * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init), (gst_rtpamrenc_init), (gst_rtpamrenc_chain): Dont set FT headers twice, it was already in the encoded bitstream. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play): * gst/rtsp/rtspconnection.c: (parse_line): Cleanups * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property): * gst/udp/gstudpsrc.h: Added caps property, we need this soon to type the buffers.
413 lines
11 KiB
C
413 lines
11 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
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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
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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 <gst/rtp/gstrtpbuffer.h>
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#include <string.h>
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#include "gstrtpamrdec.h"
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/* elementfactory information */
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static GstElementDetails gst_rtp_amrdec_details = {
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"RTP packet parser",
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"Codec/Parser/Network",
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"Extracts MPEG audio from RTP packets",
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"Wim Taymans <wim@fluendo.com>"
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};
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/* RtpAMRDec 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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ARG_0,
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ARG_FREQUENCY
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};
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/* input is an RTP packet
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*
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* params see RFC 3267, section 8.1
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*/
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static GstStaticPadTemplate gst_rtpamrdec_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 ("application/x-rtp, "
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"octet-align = (boolean) TRUE, "
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"crc = (boolean) FALSE, "
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"robust-sorting = (boolean) FALSE, "
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"interleaving = (boolean) FALSE, "
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"channels = (int) 1, " "rate = (int) 8000"
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/* following options are not needed for a decoder
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*
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
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"maxptime = (int) [ 20, MAX ], "
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"ptime = (int) [ 20, MAX ]"
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*/
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)
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);
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static GstStaticPadTemplate gst_rtpamrdec_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/AMR, " "channels = (int) 1," "rate = (int) 8000")
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);
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static void gst_rtpamrdec_class_init (GstRtpAMRDecClass * klass);
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static void gst_rtpamrdec_base_init (GstRtpAMRDecClass * klass);
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static void gst_rtpamrdec_init (GstRtpAMRDec * rtpamrdec);
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static gboolean gst_rtpamrdec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_rtpamrdec_chain (GstPad * pad, GstBuffer * buffer);
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static void gst_rtpamrdec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtpamrdec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementStateReturn gst_rtpamrdec_change_state (GstElement * element);
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static GstElementClass *parent_class = NULL;
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static GType
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gst_rtpamrdec_get_type (void)
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{
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static GType rtpamrdec_type = 0;
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if (!rtpamrdec_type) {
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static const GTypeInfo rtpamrdec_info = {
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sizeof (GstRtpAMRDecClass),
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(GBaseInitFunc) gst_rtpamrdec_base_init,
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NULL,
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(GClassInitFunc) gst_rtpamrdec_class_init,
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NULL,
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NULL,
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sizeof (GstRtpAMRDec),
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0,
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(GInstanceInitFunc) gst_rtpamrdec_init,
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};
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rtpamrdec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstRtpAMRDec",
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&rtpamrdec_info, 0);
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}
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return rtpamrdec_type;
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}
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static void
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gst_rtpamrdec_base_init (GstRtpAMRDecClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpamrdec_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpamrdec_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_amrdec_details);
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}
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static void
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gst_rtpamrdec_class_init (GstRtpAMRDecClass * 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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gobject_class->set_property = gst_rtpamrdec_set_property;
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gobject_class->get_property = gst_rtpamrdec_get_property;
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gstelement_class->change_state = gst_rtpamrdec_change_state;
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}
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static void
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gst_rtpamrdec_init (GstRtpAMRDec * rtpamrdec)
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{
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GstCaps *srccaps;
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rtpamrdec->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_rtpamrdec_src_template), "src");
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/* FIXME */
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srccaps = gst_caps_new_simple ("audio/AMR",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
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gst_pad_set_caps (rtpamrdec->srcpad, srccaps);
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gst_caps_unref (srccaps);
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gst_element_add_pad (GST_ELEMENT (rtpamrdec), rtpamrdec->srcpad);
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rtpamrdec->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_rtpamrdec_sink_template), "sink");
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gst_pad_set_setcaps_function (rtpamrdec->sinkpad, gst_rtpamrdec_sink_setcaps);
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gst_pad_set_chain_function (rtpamrdec->sinkpad, gst_rtpamrdec_chain);
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gst_element_add_pad (GST_ELEMENT (rtpamrdec), rtpamrdec->sinkpad);
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}
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static gboolean
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gst_rtpamrdec_sink_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstStructure *structure;
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GstCaps *srccaps;
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GstRtpAMRDec *rtpamrdec;
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rtpamrdec = GST_RTP_AMR_DEC (GST_OBJECT_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_boolean (structure, "octet-align",
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&rtpamrdec->octet_align))
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rtpamrdec->octet_align = FALSE;
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/* FIXME, force octect align for now until all elements negotiate
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* correctly*/
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rtpamrdec->octet_align = TRUE;
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if (!gst_structure_get_boolean (structure, "crc", &rtpamrdec->crc))
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rtpamrdec->crc = FALSE;
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if (rtpamrdec->crc) {
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/* crc mode implies octet aligned mode */
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rtpamrdec->octet_align = TRUE;
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}
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if (!gst_structure_get_boolean (structure, "robust-sorting",
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&rtpamrdec->robust_sorting))
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rtpamrdec->robust_sorting = FALSE;
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if (rtpamrdec->robust_sorting) {
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/* robust_sorting mode implies octet aligned mode */
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rtpamrdec->octet_align = TRUE;
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}
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if (!gst_structure_get_boolean (structure, "interleaving",
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&rtpamrdec->interleaving))
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rtpamrdec->interleaving = FALSE;
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if (rtpamrdec->interleaving) {
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/* interleaving mode implies octet aligned mode */
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rtpamrdec->octet_align = TRUE;
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}
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if (!gst_structure_get_int (structure, "channels", &rtpamrdec->channels))
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rtpamrdec->channels = 1;
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if (!gst_structure_get_int (structure, "rate", &rtpamrdec->rate))
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rtpamrdec->rate = 8000;
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/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving for now */
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if (rtpamrdec->channels != 1)
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return FALSE;
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if (rtpamrdec->rate != 8000)
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return FALSE;
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if (rtpamrdec->octet_align != TRUE)
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return FALSE;
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if (rtpamrdec->crc != FALSE)
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return FALSE;
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if (rtpamrdec->robust_sorting != FALSE)
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return FALSE;
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if (rtpamrdec->interleaving != FALSE)
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return FALSE;
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srccaps = gst_caps_new_simple ("audio/AMR",
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"channels", G_TYPE_INT, rtpamrdec->channels,
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"rate", G_TYPE_INT, rtpamrdec->rate, NULL);
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gst_pad_set_caps (rtpamrdec->srcpad, srccaps);
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gst_caps_unref (srccaps);
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rtpamrdec->negotiated = TRUE;
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return TRUE;
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}
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static GstFlowReturn
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gst_rtpamrdec_chain (GstPad * pad, GstBuffer * buf)
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{
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GstRtpAMRDec *rtpamrdec;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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rtpamrdec = GST_RTP_AMR_DEC (GST_OBJECT_PARENT (pad));
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if (!rtpamrdec->negotiated)
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goto not_negotiated;
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if (!gst_rtpbuffer_validate (buf))
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goto bad_packet;
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/* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving data is to be parsed */
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{
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gint payload_len;
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guint8 *payload;
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guint32 timestamp;
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guint8 CMR, F, FT, Q;
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payload_len = gst_rtpbuffer_get_payload_len (buf);
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/* need at least 2 bytes for the header */
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if (payload_len < 2)
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goto bad_packet;
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payload = gst_rtpbuffer_get_payload (buf);
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/* parse header
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+..
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* | CMR=6 |R|R|R|R|0|FT#1=5 |Q|P|P|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+..
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*/
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CMR = (payload[0] & 0xf0) >> 4;
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F = (payload[1] & 0x80) >> 7;
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/* we only support 1 packet per RTP packet for now */
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if (F != 0)
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goto one_packet_only;
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FT = (payload[1] & 0x78) >> 3;
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Q = (payload[1] & 0x04) >> 2;
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/* skip packet */
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if (FT > 9 && FT < 15) {
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ret = GST_FLOW_OK;
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goto skip;
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}
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/* strip header now, leave FT in the data for the decoder */
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payload_len -= 1;
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payload += 1;
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timestamp = gst_rtpbuffer_get_timestamp (buf);
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outbuf = gst_buffer_new_and_alloc (payload_len);
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp * GST_SECOND / 8000;
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memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (rtpamrdec->srcpad));
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GST_DEBUG ("gst_rtpamrdec_chain: pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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ret = gst_pad_push (rtpamrdec->srcpad, outbuf);
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skip:
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gst_buffer_unref (buf);
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}
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return ret;
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not_negotiated:
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{
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GST_DEBUG ("not_negotiated");
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gst_buffer_unref (buf);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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bad_packet:
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{
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GST_DEBUG ("Packet did not validate");
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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one_packet_only:
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{
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GST_DEBUG ("One packet per RTP packet only");
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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}
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static void
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gst_rtpamrdec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpAMRDec *rtpamrdec;
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rtpamrdec = GST_RTP_AMR_DEC (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtpamrdec_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstRtpAMRDec *rtpamrdec;
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rtpamrdec = GST_RTP_AMR_DEC (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstElementStateReturn
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gst_rtpamrdec_change_state (GstElement * element)
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{
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GstRtpAMRDec *rtpamrdec;
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gint transition;
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GstElementStateReturn ret;
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rtpamrdec = GST_RTP_AMR_DEC (element);
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transition = GST_STATE_TRANSITION (element);
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switch (transition) {
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case GST_STATE_NULL_TO_READY:
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break;
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case GST_STATE_READY_TO_PAUSED:
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/* FIXME, don't require negotiation until all elements
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* do */
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rtpamrdec->negotiated = TRUE;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
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switch (transition) {
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case GST_STATE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtpamrdec_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpamrdec",
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GST_RANK_NONE, GST_TYPE_RTP_AMR_DEC);
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}
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