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-rw-r--r--gst/rtp/gstrtpqdmdepay.c419
1 files changed, 419 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpqdmdepay.c b/gst/rtp/gstrtpqdmdepay.c
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+++ b/gst/rtp/gstrtpqdmdepay.c
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+/* GStreamer
+ * Copyright (C) <2009> Edward Hervey <bilboed@bilboed.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpqdmdepay.h"
+
+GST_DEBUG_CATEGORY (rtpqdm2depay_debug);
+#define GST_CAT_DEFAULT rtpqdm2depay_debug
+
+static GstStaticPadTemplate gst_rtp_qdm2_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-qdm2")
+ );
+
+static GstStaticPadTemplate gst_rtp_qdm2_depay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "encoding-name = (string)\"X-QDM\"")
+ );
+
+GST_BOILERPLATE (GstRtpQDM2Depay, gst_rtp_qdm2_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static const guint8 headheader[20] = {
+ 0x0, 0x0, 0x0, 0xc, 0x66, 0x72, 0x6d, 0x61,
+ 0x51, 0x44, 0x4d, 0x32, 0x0, 0x0, 0x0, 0x24,
+ 0x51, 0x44, 0x43, 0x41
+};
+
+static void gst_rtp_qdm2_depay_finalize (GObject * object);
+
+static GstStateChangeReturn gst_rtp_qdm2_depay_change_state (GstElement *
+ element, GstStateChange transition);
+
+static GstBuffer *gst_rtp_qdm2_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+gboolean gst_rtp_qdm2_depay_setcaps (GstBaseRTPDepayload * filter,
+ GstCaps * caps);
+
+static void
+gst_rtp_qdm2_depay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_static_pad_template (element_class,
+ &gst_rtp_qdm2_depay_src_template);
+ gst_element_class_add_static_pad_template (element_class,
+ &gst_rtp_qdm2_depay_sink_template);
+
+
+ gst_element_class_set_details_simple (element_class, "RTP QDM2 depayloader",
+ "Codec/Depayloader/Network/RTP",
+ "Extracts QDM2 audio from RTP packets (no RFC)",
+ "Edward Hervey <bilboed@bilboed.com>");
+}
+
+static void
+gst_rtp_qdm2_depay_class_init (GstRtpQDM2DepayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ gstbasertpdepayload_class->process = gst_rtp_qdm2_depay_process;
+ gstbasertpdepayload_class->set_caps = gst_rtp_qdm2_depay_setcaps;
+
+ gobject_class->finalize = gst_rtp_qdm2_depay_finalize;
+
+ gstelement_class->change_state = gst_rtp_qdm2_depay_change_state;
+}
+
+static void
+gst_rtp_qdm2_depay_init (GstRtpQDM2Depay * rtpqdm2depay,
+ GstRtpQDM2DepayClass * klass)
+{
+ rtpqdm2depay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_qdm2_depay_finalize (GObject * object)
+{
+ GstRtpQDM2Depay *rtpqdm2depay;
+
+ rtpqdm2depay = GST_RTP_QDM2_DEPAY (object);
+
+ g_object_unref (rtpqdm2depay->adapter);
+ rtpqdm2depay->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+// only on the sink
+gboolean
+gst_rtp_qdm2_depay_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps)
+{
+ GstStructure *structure = gst_caps_get_structure (caps, 0);
+ gint clock_rate;
+
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 44100; // default
+ filter->clock_rate = clock_rate;
+
+ /* will set caps later */
+
+ return TRUE;
+}
+
+static void
+flush_data (GstRtpQDM2Depay * depay)
+{
+ guint i;
+ guint avail;
+
+ if ((avail = gst_adapter_available (depay->adapter)))
+ gst_adapter_flush (depay->adapter, avail);
+
+ GST_DEBUG ("Flushing %d packets", depay->nbpackets);
+
+ for (i = 0; depay->packets[i]; i++) {
+ QDM2Packet *pack = depay->packets[i];
+ guint32 crc = 0;
+ int i = 0;
+ GstBuffer *buf;
+ guint8 *data;
+
+ /* CRC is the sum of everything (including first bytes) */
+
+ data = pack->data;
+
+ if (G_UNLIKELY (data == NULL))
+ continue;
+
+ /* If the packet size is bigger than 0xff, we need 2 bytes to store the size */
+ if (depay->packetsize > 0xff) {
+ /* Expanded size 0x02 | 0x80 */
+ data[0] = 0x82;
+ GST_WRITE_UINT16_BE (data + 1, depay->packetsize - 3);
+ } else {
+ data[0] = 0x2;
+ data[1] = depay->packetsize - 2;
+ }
+
+ /* Calculate CRC */
+ for (; i < depay->packetsize; i++)
+ crc += data[i];
+
+ GST_DEBUG ("CRC is 0x%x", crc);
+
+ /* Write CRC */
+ if (depay->packetsize > 0xff)
+ GST_WRITE_UINT16_BE (data + 3, crc);
+ else
+ GST_WRITE_UINT16_BE (data + 2, crc);
+
+ GST_MEMDUMP ("Extracted packet", data, depay->packetsize);
+
+ buf = gst_buffer_new ();
+ GST_BUFFER_DATA (buf) = data;
+ GST_BUFFER_MALLOCDATA (buf) = data;
+ GST_BUFFER_SIZE (buf) = depay->packetsize;
+
+ gst_adapter_push (depay->adapter, buf);
+
+ if (pack->data) {
+ pack->data = NULL;
+ }
+ }
+}
+
+static void
+add_packet (GstRtpQDM2Depay * depay, guint32 pid, guint32 len, guint8 * data)
+{
+ QDM2Packet *packet;
+
+ if (G_UNLIKELY (!depay->configured))
+ return;
+
+ GST_DEBUG ("pid:%d, len:%d, data:%p", pid, len, data);
+
+ if (G_UNLIKELY (depay->packets[pid] == NULL)) {
+ depay->packets[pid] = g_malloc0 (sizeof (QDM2Packet));
+ depay->nbpackets = MAX (depay->nbpackets, pid + 1);
+ }
+ packet = depay->packets[pid];
+
+ GST_DEBUG ("packet:%p", packet);
+ GST_DEBUG ("packet->data:%p", packet->data);
+
+ if (G_UNLIKELY (packet->data == NULL)) {
+ packet->data = g_malloc0 (depay->packetsize);
+ /* We leave space for the header/crc */
+ if (depay->packetsize > 0xff)
+ packet->offs = 5;
+ else
+ packet->offs = 4;
+ }
+
+ /* Finally copy the data over */
+ memcpy (packet->data + packet->offs, data, len);
+ packet->offs += len;
+}
+
+static GstBuffer *
+gst_rtp_qdm2_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstRtpQDM2Depay *rtpqdm2depay;
+ GstBuffer *outbuf;
+ guint16 seq;
+
+ rtpqdm2depay = GST_RTP_QDM2_DEPAY (depayload);
+
+ {
+ gint payload_len;
+ guint8 *payload;
+ guint avail;
+ guint pos = 0;
+
+ payload_len = gst_rtp_buffer_get_payload_len (buf);
+ if (payload_len < 3)
+ goto bad_packet;
+
+ payload = gst_rtp_buffer_get_payload (buf);
+ seq = gst_rtp_buffer_get_seq (buf);
+ if (G_UNLIKELY (seq != rtpqdm2depay->nextseq)) {
+ GST_DEBUG ("GAP in sequence number, Resetting data !");
+ /* Flush previous data */
+ flush_data (rtpqdm2depay);
+ /* And store new timestamp */
+ rtpqdm2depay->ptimestamp = rtpqdm2depay->timestamp;
+ rtpqdm2depay->timestamp = GST_BUFFER_TIMESTAMP (buf);
+ /* And that previous data will be pushed at the bottom */
+ }
+ rtpqdm2depay->nextseq = seq + 1;
+
+ GST_DEBUG ("Payload size %d 0x%x sequence:%d", payload_len, payload_len,
+ seq);
+
+ GST_MEMDUMP ("Incoming payload", payload, payload_len);
+
+ while (pos < payload_len) {
+ switch (payload[pos]) {
+ case 0x80:{
+ GST_DEBUG ("Unrecognized 0x80 marker, skipping 12 bytes");
+ pos += 12;
+ }
+ break;
+ case 0xff:
+ /* HEADERS */
+ GST_DEBUG ("Headers");
+ /* Store the incoming timestamp */
+ rtpqdm2depay->ptimestamp = rtpqdm2depay->timestamp;
+ rtpqdm2depay->timestamp = GST_BUFFER_TIMESTAMP (buf);
+ /* flush the internal data if needed */
+ flush_data (rtpqdm2depay);
+ if (G_UNLIKELY (!rtpqdm2depay->configured)) {
+ guint8 *ourdata;
+ GstBuffer *codecdata;
+ GstCaps *caps;
+
+ /* First bytes are unknown */
+ GST_MEMDUMP ("Header", payload + pos, 32);
+ ourdata = payload + pos + 10;
+ pos += 10;
+ rtpqdm2depay->channs = GST_READ_UINT32_BE (payload + pos + 4);
+ rtpqdm2depay->samplerate = GST_READ_UINT32_BE (payload + pos + 8);
+ rtpqdm2depay->bitrate = GST_READ_UINT32_BE (payload + pos + 12);
+ rtpqdm2depay->blocksize = GST_READ_UINT32_BE (payload + pos + 16);
+ rtpqdm2depay->framesize = GST_READ_UINT32_BE (payload + pos + 20);
+ rtpqdm2depay->packetsize = GST_READ_UINT32_BE (payload + pos + 24);
+ /* 16 bit empty block (0x02 0x00) */
+ pos += 30;
+ GST_DEBUG
+ ("channs:%d, samplerate:%d, bitrate:%d, blocksize:%d, framesize:%d, packetsize:%d",
+ rtpqdm2depay->channs, rtpqdm2depay->samplerate,
+ rtpqdm2depay->bitrate, rtpqdm2depay->blocksize,
+ rtpqdm2depay->framesize, rtpqdm2depay->packetsize);
+
+ /* Caps */
+ codecdata = gst_buffer_new_and_alloc (48);
+ memcpy (GST_BUFFER_DATA (codecdata), headheader, 20);
+ memcpy (GST_BUFFER_DATA (codecdata) + 20, ourdata, 28);
+
+ caps = gst_caps_new_simple ("audio/x-qdm2",
+ "samplesize", G_TYPE_INT, 16,
+ "rate", G_TYPE_INT, rtpqdm2depay->samplerate,
+ "channels", G_TYPE_INT, rtpqdm2depay->channs,
+ "codec_data", GST_TYPE_BUFFER, codecdata, NULL);
+ gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), caps);
+ gst_caps_unref (caps);
+ rtpqdm2depay->configured = TRUE;
+ } else {
+ GST_DEBUG ("Already configured, skipping headers");
+ pos += 40;
+ }
+ break;
+ default:{
+ /* Shuffled packet contents */
+ guint packetid = payload[pos++];
+ guint packettype = payload[pos++];
+ guint packlen = payload[pos++];
+ guint hsize = 2;
+
+ GST_DEBUG ("Packet id:%d, type:0x%x, len:%d",
+ packetid, packettype, packlen);
+
+ /* Packets bigger than 0xff bytes have a type with the high bit set */
+ if (G_UNLIKELY (packettype & 0x80)) {
+ packettype &= 0x7f;
+ packlen <<= 8;
+ packlen |= payload[pos++];
+ hsize = 3;
+ GST_DEBUG ("Packet id:%d, type:0x%x, len:%d",
+ packetid, packettype, packlen);
+ }
+
+ if (packettype > 0x7f) {
+ GST_ERROR ("HOUSTON WE HAVE A PROBLEM !!!!");
+ }
+ add_packet (rtpqdm2depay, packetid, packlen + hsize,
+ payload + pos - hsize);
+ pos += packlen;
+ }
+ }
+ }
+
+ GST_DEBUG ("final pos %d", pos);
+
+ avail = gst_adapter_available (rtpqdm2depay->adapter);
+ if (G_UNLIKELY (avail)) {
+ GST_DEBUG ("Pushing out %d bytes of collected data", avail);
+ outbuf = gst_adapter_take_buffer (rtpqdm2depay->adapter, avail);
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpqdm2depay->ptimestamp;
+ GST_DEBUG ("Outgoing buffer timestamp %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (rtpqdm2depay->ptimestamp));
+ return outbuf;
+ }
+ }
+ return NULL;
+
+ /* ERRORS */
+bad_packet:
+ {
+ GST_ELEMENT_WARNING (rtpqdm2depay, STREAM, DECODE,
+ (NULL), ("Packet was too short"));
+ return NULL;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtp_qdm2_depay_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstRtpQDM2Depay *rtpqdm2depay;
+ GstStateChangeReturn ret;
+
+ rtpqdm2depay = GST_RTP_QDM2_DEPAY (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_adapter_clear (rtpqdm2depay->adapter);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+ return ret;
+}
+
+gboolean
+gst_rtp_qdm2_depay_plugin_init (GstPlugin * plugin)
+{
+ GST_DEBUG_CATEGORY_INIT (rtpqdm2depay_debug, "rtpqdm2depay", 0,
+ "RTP QDM2 depayloader");
+
+ return gst_element_register (plugin, "rtpqdm2depay",
+ GST_RANK_SECONDARY, GST_TYPE_RTP_QDM2_DEPAY);
+}