--- /dev/null
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.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 <gst/rtp/gstrtpbuffer.h>
+
+#include <stdlib.h>
+#include <string.h>
+#include "gstrtpqcelpdepay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpqcelpdepay_debug);
+#define GST_CAT_DEFAULT (rtpqcelpdepay_debug)
+
+/* references:
+ *
+ * RFC 2658 - RTP Payload Format for PureVoice(tm) Audio
+ */
+#define FRAME_DURATION (20 * GST_MSECOND)
+
+/* RtpQCELPDepay signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0
+};
+
+static GstStaticPadTemplate gst_rtp_qcelp_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 ", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) \"QCELP\"; "
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_QCELP_STRING ", "
+ "clock-rate = (int) 8000")
+ );
+
+static GstStaticPadTemplate gst_rtp_qcelp_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/qcelp, " "channels = (int) 1," "rate = (int) 8000")
+ );
+
+static void gst_rtp_qcelp_depay_finalize (GObject * object);
+
+static gboolean gst_rtp_qcelp_depay_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+static GstBuffer *gst_rtp_qcelp_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+
+GST_BOILERPLATE (GstRtpQCELPDepay, gst_rtp_qcelp_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtp_qcelp_depay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_qcelp_depay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_qcelp_depay_sink_template));
+
+ gst_element_class_set_details_simple (element_class, "RTP QCELP depayloader",
+ "Codec/Depayloader/Network",
+ "Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)",
+ "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_qcelp_depay_class_init (GstRtpQCELPDepayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ gobject_class->finalize = gst_rtp_qcelp_depay_finalize;
+
+ gstbasertpdepayload_class->process = gst_rtp_qcelp_depay_process;
+ gstbasertpdepayload_class->set_caps = gst_rtp_qcelp_depay_setcaps;
+
+ GST_DEBUG_CATEGORY_INIT (rtpqcelpdepay_debug, "rtpqcelpdepay", 0,
+ "QCELP RTP Depayloader");
+}
+
+static void
+gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay,
+ GstRtpQCELPDepayClass * klass)
+{
+ GstBaseRTPDepayload *depayload;
+
+ depayload = GST_BASE_RTP_DEPAYLOAD (rtpqcelpdepay);
+}
+
+static void
+gst_rtp_qcelp_depay_finalize (GObject * object)
+{
+ GstRtpQCELPDepay *depay;
+
+ depay = GST_RTP_QCELP_DEPAY (object);
+
+ if (depay->packets != NULL) {
+ g_ptr_array_foreach (depay->packets, (GFunc) gst_buffer_unref, NULL);
+ g_ptr_array_free (depay->packets, TRUE);
+ depay->packets = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+
+static gboolean
+gst_rtp_qcelp_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+ GstCaps *srccaps;
+ GstRtpQCELPDepay *rtpqcelpdepay;
+ gboolean res;
+
+ rtpqcelpdepay = GST_RTP_QCELP_DEPAY (depayload);
+
+ srccaps = gst_caps_new_simple ("audio/qcelp",
+ "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ gst_caps_unref (srccaps);
+
+ return res;
+}
+
+static gint frame_size[16] = {
+ 1, 4, 8, 17, 35, -8, 0, 0,
+ 0, 0, 0, 0, 0, 0, 1, 0
+};
+
+/* get the frame length, 0 is invalid, negative values are invalid but can be
+ * recovered from. */
+static gint
+get_frame_len (GstRtpQCELPDepay * depay, guint8 frame_type)
+{
+ if (frame_type > 16)
+ return 0;
+
+ return frame_size[frame_type];
+}
+
+static guint
+count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size)
+{
+ guint count = 0;
+
+ while (size > 0) {
+ gint frame_len;
+
+ frame_len = get_frame_len (depay, data[0]);
+
+ /* 0 is invalid and we throw away the remainder of the frames */
+ if (frame_len == 0)
+ break;
+
+ if (frame_len < 0)
+ frame_len = -frame_len;
+
+ if (frame_len > size)
+ break;
+
+ size -= frame_len;
+ data += frame_len;
+ count++;
+ }
+ return count;
+}
+
+static void
+flush_packets (GstRtpQCELPDepay * depay)
+{
+ guint i, size;
+
+ GST_DEBUG_OBJECT (depay, "flushing packets");
+
+ size = depay->packets->len;
+
+ for (i = 0; i < size; i++) {
+ GstBuffer *outbuf;
+
+ outbuf = g_ptr_array_index (depay->packets, i);
+ g_ptr_array_index (depay->packets, i) = NULL;
+
+ gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (depay), outbuf);
+ }
+
+ /* and reset interleaving state */
+ depay->interleaved = FALSE;
+ depay->bundling = 0;
+}
+
+static void
+add_packet (GstRtpQCELPDepay * depay, guint LLL, guint NNN, guint index,
+ GstBuffer * outbuf)
+{
+ guint idx;
+ GstBuffer *old;
+
+ /* figure out the position in the array, note that index is never 0 because we
+ * push those packets immediately. */
+ idx = NNN + ((LLL + 1) * (index - 1));
+
+ GST_DEBUG_OBJECT (depay, "adding packet at index %u", idx);
+ /* free old buffer (should not happen) */
+ old = g_ptr_array_index (depay->packets, idx);
+ if (old)
+ gst_buffer_unref (old);
+
+ /* store new buffer */
+ g_ptr_array_index (depay->packets, idx) = outbuf;
+}
+
+static GstBuffer *
+create_erasure_buffer (GstRtpQCELPDepay * depay)
+{
+ GstBuffer *outbuf;
+
+ outbuf = gst_buffer_new_and_alloc (1);
+ GST_BUFFER_DATA (outbuf)[0] = 14;
+
+ return outbuf;
+}
+
+static GstBuffer *
+gst_rtp_qcelp_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstRtpQCELPDepay *depay;
+ GstBuffer *outbuf;
+ GstClockTime timestamp;
+ guint payload_len, offset, index;
+ guint8 *payload;
+ guint RR, LLL, NNN;
+
+ depay = GST_RTP_QCELP_DEPAY (depayload);
+
+ payload_len = gst_rtp_buffer_get_payload_len (buf);
+
+ if (payload_len < 2)
+ goto too_small;
+
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+
+ payload = gst_rtp_buffer_get_payload (buf);
+
+ /* 0 1 2 3 4 5 6 7
+ * +-+-+-+-+-+-+-+-+
+ * |RR | LLL | NNN |
+ * +-+-+-+-+-+-+-+-+
+ */
+ RR = payload[0] >> 6;
+ LLL = (payload[0] & 0x38) >> 3;
+ NNN = (payload[0] & 0x07);
+
+ payload_len--;
+ payload++;
+
+ GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN);
+
+ if (LLL > 5)
+ goto invalid_lll;
+
+ if (NNN > LLL)
+ goto invalid_nnn;
+
+ if (LLL != 0) {
+ /* we are interleaved */
+ if (!depay->interleaved) {
+ guint size;
+
+ GST_DEBUG_OBJECT (depay, "starting interleaving group");
+ /* bundling is not allowed to change in one interleave group */
+ depay->bundling = count_packets (depay, payload, payload_len);
+ GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling);
+ /* we have one bundle where NNN goes from 0 to L, we don't store the index
+ * 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */
+ size = (depay->bundling - 1) * (LLL + 1);
+ /* create the array to hold the packets */
+ if (depay->packets == NULL)
+ depay->packets = g_ptr_array_sized_new (size);
+ GST_DEBUG_OBJECT (depay, "created packet array of size %u", size);
+ g_ptr_array_set_size (depay->packets, size);
+ /* we were previously not interleaved, figure out how much space we
+ * need to deinterleave */
+ depay->interleaved = TRUE;
+ }
+ } else {
+ /* we are not interleaved */
+ if (depay->interleaved) {
+ GST_DEBUG_OBJECT (depay, "stopping interleaving");
+ /* flush packets if we were previously interleaved */
+ flush_packets (depay);
+ }
+ depay->bundling = 0;
+ }
+
+ index = 0;
+ offset = 1;
+
+ while (payload_len > 0) {
+ gint frame_len;
+ gboolean do_erasure;
+
+ frame_len = get_frame_len (depay, payload[0]);
+ GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len);
+
+ if (frame_len == 0)
+ goto invalid_frame;
+
+ if (frame_len < 0) {
+ /* need to add an erasure frame but we can recover */
+ frame_len = -frame_len;
+ do_erasure = TRUE;
+ } else {
+ do_erasure = FALSE;
+ }
+
+ if (frame_len > payload_len)
+ goto invalid_frame;
+
+ if (do_erasure) {
+ /* create erasure frame */
+ outbuf = create_erasure_buffer (depay);
+ } else {
+ /* each frame goes into its buffer */
+ outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, offset, frame_len);
+ }
+
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ GST_BUFFER_DURATION (outbuf) = FRAME_DURATION;
+
+ if (!depay->interleaved || index == 0) {
+ /* not interleaved or first frame in packet, just push */
+ gst_base_rtp_depayload_push (depayload, outbuf);
+
+ if (timestamp != -1)
+ timestamp += FRAME_DURATION;
+ } else {
+ /* put in interleave buffer */
+ add_packet (depay, LLL, NNN, index, outbuf);
+
+ if (timestamp != -1)
+ timestamp += (FRAME_DURATION * (LLL + 1));
+ }
+
+ payload_len -= frame_len;
+ payload += frame_len;
+ offset += frame_len;
+ index++;
+
+ /* discard excess packets */
+ if (depay->bundling > 0 && depay->bundling <= index)
+ break;
+ }
+ while (index < depay->bundling) {
+ GST_DEBUG_OBJECT (depay, "filling with erasure buffer");
+ /* fill remainder with erasure packets */
+ outbuf = create_erasure_buffer (depay);
+ add_packet (depay, LLL, NNN, index, outbuf);
+ index++;
+ }
+ if (depay->interleaved && LLL == NNN) {
+ GST_DEBUG_OBJECT (depay, "interleave group ended, flushing");
+ /* we have the complete interleave group, flush */
+ flush_packets (depay);
+ }
+
+ return NULL;
+
+ /* ERRORS */
+too_small:
+ {
+ GST_ELEMENT_WARNING (depay, STREAM, DECODE,
+ (NULL), ("QCELP RTP payload too small (%d)", payload_len));
+ return NULL;
+ }
+invalid_lll:
+ {
+ GST_ELEMENT_WARNING (depay, STREAM, DECODE,
+ (NULL), ("QCELP RTP invalid LLL received (%d)", LLL));
+ return NULL;
+ }
+invalid_nnn:
+ {
+ GST_ELEMENT_WARNING (depay, STREAM, DECODE,
+ (NULL), ("QCELP RTP invalid NNN received (%d)", NNN));
+ return NULL;
+ }
+invalid_frame:
+ {
+ GST_ELEMENT_WARNING (depay, STREAM, DECODE,
+ (NULL), ("QCELP RTP invalid frame received"));
+ return NULL;
+ }
+}
+
+gboolean
+gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpqcelpdepay",
+ GST_RANK_MARGINAL, GST_TYPE_RTP_QCELP_DEPAY);
+}
projects / platform / upstream / gst-plugins-good.git / commitdiff