mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
420 lines
12 KiB
C
420 lines
12 KiB
C
/* GStreamer
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* Copyright (C) <2009> Edward Hervey <bilboed@bilboed.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 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 <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpqdmdepay.h"
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GST_DEBUG_CATEGORY (rtpqdm2depay_debug);
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#define GST_CAT_DEFAULT rtpqdm2depay_debug
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static GstStaticPadTemplate gst_rtp_qdm2_depay_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-qdm2")
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);
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static GstStaticPadTemplate gst_rtp_qdm2_depay_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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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"encoding-name = (string)\"X-QDM\"")
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);
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#define gst_rtp_qdm2_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpQDM2Depay, gst_rtp_qdm2_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static const guint8 headheader[20] = {
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0x0, 0x0, 0x0, 0xc, 0x66, 0x72, 0x6d, 0x61,
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0x51, 0x44, 0x4d, 0x32, 0x0, 0x0, 0x0, 0x24,
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0x51, 0x44, 0x43, 0x41
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};
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static void gst_rtp_qdm2_depay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_qdm2_depay_change_state (GstElement *
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element, GstStateChange transition);
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static GstBuffer *gst_rtp_qdm2_depay_process (GstRTPBaseDepayload * depayload,
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GstBuffer * buf);
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gboolean gst_rtp_qdm2_depay_setcaps (GstRTPBaseDepayload * filter,
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GstCaps * caps);
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static void
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gst_rtp_qdm2_depay_class_init (GstRtpQDM2DepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gstrtpbasedepayload_class->process = gst_rtp_qdm2_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_qdm2_depay_setcaps;
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gobject_class->finalize = gst_rtp_qdm2_depay_finalize;
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gstelement_class->change_state = gst_rtp_qdm2_depay_change_state;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_qdm2_depay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_qdm2_depay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP QDM2 depayloader",
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"Codec/Depayloader/Network/RTP",
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"Extracts QDM2 audio from RTP packets (no RFC)",
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"Edward Hervey <bilboed@bilboed.com>");
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}
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static void
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gst_rtp_qdm2_depay_init (GstRtpQDM2Depay * rtpqdm2depay)
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{
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rtpqdm2depay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_qdm2_depay_finalize (GObject * object)
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{
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GstRtpQDM2Depay *rtpqdm2depay;
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rtpqdm2depay = GST_RTP_QDM2_DEPAY (object);
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g_object_unref (rtpqdm2depay->adapter);
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rtpqdm2depay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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// only on the sink
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gboolean
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gst_rtp_qdm2_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
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{
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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gint clock_rate;
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 44100; // default
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filter->clock_rate = clock_rate;
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/* will set caps later */
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return TRUE;
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}
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static void
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flush_data (GstRtpQDM2Depay * depay)
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{
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guint i;
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guint avail;
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if ((avail = gst_adapter_available (depay->adapter)))
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gst_adapter_flush (depay->adapter, avail);
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GST_DEBUG ("Flushing %d packets", depay->nbpackets);
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for (i = 0; depay->packets[i]; i++) {
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QDM2Packet *pack = depay->packets[i];
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guint32 crc = 0;
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int i = 0;
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GstBuffer *buf;
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guint8 *data;
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/* CRC is the sum of everything (including first bytes) */
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data = pack->data;
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if (G_UNLIKELY (data == NULL))
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continue;
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/* If the packet size is bigger than 0xff, we need 2 bytes to store the size */
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if (depay->packetsize > 0xff) {
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/* Expanded size 0x02 | 0x80 */
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data[0] = 0x82;
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GST_WRITE_UINT16_BE (data + 1, depay->packetsize - 3);
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} else {
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data[0] = 0x2;
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data[1] = depay->packetsize - 2;
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}
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/* Calculate CRC */
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for (; i < depay->packetsize; i++)
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crc += data[i];
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GST_DEBUG ("CRC is 0x%x", crc);
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/* Write CRC */
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if (depay->packetsize > 0xff)
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GST_WRITE_UINT16_BE (data + 3, crc);
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else
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GST_WRITE_UINT16_BE (data + 2, crc);
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GST_MEMDUMP ("Extracted packet", data, depay->packetsize);
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buf = gst_buffer_new ();
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gst_buffer_append_memory (buf,
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gst_memory_new_wrapped (0, data, depay->packetsize, 0,
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depay->packetsize, data, g_free));
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gst_adapter_push (depay->adapter, buf);
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if (pack->data) {
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pack->data = NULL;
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}
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}
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}
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static void
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add_packet (GstRtpQDM2Depay * depay, guint32 pid, guint32 len, guint8 * data)
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{
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QDM2Packet *packet;
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if (G_UNLIKELY (!depay->configured))
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return;
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GST_DEBUG ("pid:%d, len:%d, data:%p", pid, len, data);
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if (G_UNLIKELY (depay->packets[pid] == NULL)) {
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depay->packets[pid] = g_malloc0 (sizeof (QDM2Packet));
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depay->nbpackets = MAX (depay->nbpackets, pid + 1);
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}
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packet = depay->packets[pid];
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GST_DEBUG ("packet:%p", packet);
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GST_DEBUG ("packet->data:%p", packet->data);
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if (G_UNLIKELY (packet->data == NULL)) {
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packet->data = g_malloc0 (depay->packetsize);
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/* We leave space for the header/crc */
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if (depay->packetsize > 0xff)
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packet->offs = 5;
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else
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packet->offs = 4;
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}
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/* Finally copy the data over */
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memcpy (packet->data + packet->offs, data, len);
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packet->offs += len;
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}
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static GstBuffer *
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gst_rtp_qdm2_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
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{
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GstRtpQDM2Depay *rtpqdm2depay;
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GstBuffer *outbuf = NULL;
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guint16 seq;
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GstRTPBuffer rtp = { NULL };
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rtpqdm2depay = GST_RTP_QDM2_DEPAY (depayload);
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{
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gint payload_len;
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guint8 *payload;
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guint avail;
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guint pos = 0;
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gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
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payload_len = gst_rtp_buffer_get_payload_len (&rtp);
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if (payload_len < 3)
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goto bad_packet;
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payload = gst_rtp_buffer_get_payload (&rtp);
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seq = gst_rtp_buffer_get_seq (&rtp);
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if (G_UNLIKELY (seq != rtpqdm2depay->nextseq)) {
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GST_DEBUG ("GAP in sequence number, Resetting data !");
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/* Flush previous data */
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flush_data (rtpqdm2depay);
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/* And store new timestamp */
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rtpqdm2depay->ptimestamp = rtpqdm2depay->timestamp;
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rtpqdm2depay->timestamp = GST_BUFFER_TIMESTAMP (buf);
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/* And that previous data will be pushed at the bottom */
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}
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rtpqdm2depay->nextseq = seq + 1;
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GST_DEBUG ("Payload size %d 0x%x sequence:%d", payload_len, payload_len,
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seq);
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GST_MEMDUMP ("Incoming payload", payload, payload_len);
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while (pos < payload_len) {
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switch (payload[pos]) {
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case 0x80:{
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GST_DEBUG ("Unrecognized 0x80 marker, skipping 12 bytes");
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pos += 12;
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}
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break;
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case 0xff:
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/* HEADERS */
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GST_DEBUG ("Headers");
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/* Store the incoming timestamp */
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rtpqdm2depay->ptimestamp = rtpqdm2depay->timestamp;
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rtpqdm2depay->timestamp = GST_BUFFER_TIMESTAMP (buf);
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/* flush the internal data if needed */
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flush_data (rtpqdm2depay);
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if (G_UNLIKELY (!rtpqdm2depay->configured)) {
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guint8 *ourdata;
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GstBuffer *codecdata;
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GstMapInfo cmap;
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GstCaps *caps;
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/* First bytes are unknown */
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GST_MEMDUMP ("Header", payload + pos, 32);
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ourdata = payload + pos + 10;
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pos += 10;
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rtpqdm2depay->channs = GST_READ_UINT32_BE (payload + pos + 4);
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rtpqdm2depay->samplerate = GST_READ_UINT32_BE (payload + pos + 8);
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rtpqdm2depay->bitrate = GST_READ_UINT32_BE (payload + pos + 12);
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rtpqdm2depay->blocksize = GST_READ_UINT32_BE (payload + pos + 16);
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rtpqdm2depay->framesize = GST_READ_UINT32_BE (payload + pos + 20);
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rtpqdm2depay->packetsize = GST_READ_UINT32_BE (payload + pos + 24);
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/* 16 bit empty block (0x02 0x00) */
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pos += 30;
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GST_DEBUG
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("channs:%d, samplerate:%d, bitrate:%d, blocksize:%d, framesize:%d, packetsize:%d",
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rtpqdm2depay->channs, rtpqdm2depay->samplerate,
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rtpqdm2depay->bitrate, rtpqdm2depay->blocksize,
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rtpqdm2depay->framesize, rtpqdm2depay->packetsize);
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/* Caps */
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codecdata = gst_buffer_new_and_alloc (48);
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gst_buffer_map (codecdata, &cmap, GST_MAP_WRITE);
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memcpy (cmap.data, headheader, 20);
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memcpy (cmap.data + 20, ourdata, 28);
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gst_buffer_unmap (codecdata, &cmap);
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caps = gst_caps_new_simple ("audio/x-qdm2",
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"samplesize", G_TYPE_INT, 16,
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"rate", G_TYPE_INT, rtpqdm2depay->samplerate,
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"channels", G_TYPE_INT, rtpqdm2depay->channs,
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"codec_data", GST_TYPE_BUFFER, codecdata, NULL);
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gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), caps);
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gst_caps_unref (caps);
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rtpqdm2depay->configured = TRUE;
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} else {
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GST_DEBUG ("Already configured, skipping headers");
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pos += 40;
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}
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break;
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default:{
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/* Shuffled packet contents */
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guint packetid = payload[pos++];
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guint packettype = payload[pos++];
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guint packlen = payload[pos++];
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guint hsize = 2;
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GST_DEBUG ("Packet id:%d, type:0x%x, len:%d",
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packetid, packettype, packlen);
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/* Packets bigger than 0xff bytes have a type with the high bit set */
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if (G_UNLIKELY (packettype & 0x80)) {
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packettype &= 0x7f;
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packlen <<= 8;
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packlen |= payload[pos++];
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hsize = 3;
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GST_DEBUG ("Packet id:%d, type:0x%x, len:%d",
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packetid, packettype, packlen);
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}
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if (packettype > 0x7f) {
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GST_ERROR ("HOUSTON WE HAVE A PROBLEM !!!!");
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}
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add_packet (rtpqdm2depay, packetid, packlen + hsize,
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payload + pos - hsize);
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pos += packlen;
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}
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}
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}
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GST_DEBUG ("final pos %d", pos);
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avail = gst_adapter_available (rtpqdm2depay->adapter);
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if (G_UNLIKELY (avail)) {
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GST_DEBUG ("Pushing out %d bytes of collected data", avail);
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outbuf = gst_adapter_take_buffer (rtpqdm2depay->adapter, avail);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpqdm2depay->ptimestamp;
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GST_DEBUG ("Outgoing buffer timestamp %" GST_TIME_FORMAT,
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GST_TIME_ARGS (rtpqdm2depay->ptimestamp));
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}
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}
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gst_rtp_buffer_unmap (&rtp);
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return outbuf;
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/* ERRORS */
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bad_packet:
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{
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GST_ELEMENT_WARNING (rtpqdm2depay, STREAM, DECODE,
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(NULL), ("Packet was too short"));
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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}
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}
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static GstStateChangeReturn
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gst_rtp_qdm2_depay_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstRtpQDM2Depay *rtpqdm2depay;
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GstStateChangeReturn ret;
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rtpqdm2depay = GST_RTP_QDM2_DEPAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (rtpqdm2depay->adapter);
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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, transition);
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switch (transition) {
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case GST_STATE_CHANGE_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_rtp_qdm2_depay_plugin_init (GstPlugin * plugin)
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{
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GST_DEBUG_CATEGORY_INIT (rtpqdm2depay_debug, "rtpqdm2depay", 0,
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"RTP QDM2 depayloader");
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return gst_element_register (plugin, "rtpqdm2depay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_QDM2_DEPAY);
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}
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