Framework API now complete in its first iteration. So far with support for raw, opus and amrwb.

1 files changed, 188 insertions, 0 deletions

diff --git a/src/rtp_profile_opus.cc b/src/rtp_profile_opus.cc
new file mode 100644
index 0000000..102d175
--- /dev/null
+++ b/src/rtp_profile_opus.cc
@@ -0,0 +1,188 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set et sw=2 ts=2: */
+/***************************************************************************
+ *            rtp_profile_opus.cc
+ *
+ *  Tue Sep 10 13:53:24 CEST 2013
+ *  Copyright 2013 Bent Bisballe Nyeng
+ *  deva@aasimon.org
+ ****************************************************************************/
+
+/*
+ *  This file is part of lrtp.
+ *
+ * lrtp is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * lrtp 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with lrtp; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Part of this code is an adaptation of the GStreamer RTP profile code for Celt
+ * by Wim Taymans <wim.taymans@gmail.com>
+ * The original code can be found in the gst-plugins-good package version 1.1.3:
+ * http://gstreamer.freedesktop.org/src/gst-plugins-good
+ */
+#include "rtp_profile_opus.h"
+
+/*
+         +--------------+----------------+-----------+----------+
+         | Abbreviation |      Name      | Bandwidth | Sampling |
+         +--------------+----------------+-----------+----------+
+         |      nb      |   Narrowband   |  0 - 4000 |   8000   |
+         |              |                |           |          |
+         |      mb      |   Mediumband   |  0 - 6000 |   12000  |
+         |              |                |           |          |
+         |      wb      |    Wideband    |  0 - 8000 |   16000  |
+         |              |                |           |          |
+         |      swb     | Super-wideband | 0 - 12000 |   24000  |
+         |              |                |           |          |
+         |      fb      |    Fullband    | 0 - 20000 |   48000  |
+         +--------------+----------------+-----------+----------+
+
+                          Audio bandwidth naming
+
+                                  Table 1
+*/
+/*
+Recommended Bitrate
+
+   For a frame size of 20 ms, these are the bitrate "sweet spots" for
+   Opus in various configurations:
+   o  8-12 kb/s for NB speech,
+   o  16-20 kb/s for WB speech,
+   o  28-40 kb/s for FB speech,
+   o  48-64 kb/s for FB mono music, and
+   o  64-128 kb/s for FB stereo music.
+*/
+
+/*
+Forward Error Correction (FEC)
+
+   The voice mode of Opus allows for "in-band" forward error correction
+   (FEC) data to be embedded into the bit stream of Opus.
+*/
+
+/*
+RTP Header Usage
+
+   The format of the RTP header is specified in [RFC3550].  The Opus
+   payload format uses the fields of the RTP header consistent with this
+   specification.
+
+   The payload length of Opus is a multiple number of octets and
+   therefore no padding is required.  The payload MAY be padded by an
+   integer number of octets according to [RFC3550].
+
+   The marker bit (M) of the RTP header is used in accordance with
+   Section 4.1 of [RFC3551].
+
+   The RTP payload type for Opus has not been assigned statically and is
+   expected to be assigned dynamically.
+
+   The receiving side MUST be prepared to receive duplicates of RTP
+   packets.  Only one of those payloads MUST be provided to the Opus
+   decoder for decoding and others MUST be discarded.
+
+   Opus supports 5 different audio bandwidths which may be adjusted
+   during the duration of a call.  The RTP timestamp clock frequency is
+   defined as the highest supported sampling frequency of Opus, i.e.
+   48000 Hz, for all modes and sampling rates of Opus.  The unit for the
+   timestamp is samples per single (mono) channel.  The RTP timestamp
+   corresponds to the sample time of the first encoded sample in the
+   encoded frame.  For sampling rates lower than 48000 Hz the number of
+   samples has to be multiplied with a multiplier according to Table 2
+   to determine the RTP timestamp.
+
+                         +---------+------------+
+                         | fs (Hz) | Multiplier |
+                         +---------+------------+
+                         |   8000  |      6     |
+                         |         |            |
+                         |  12000  |      4     |
+                         |         |            |
+                         |  16000  |      3     |
+                         |         |            |
+                         |  24000  |      2     |
+                         |         |            |
+                         |  48000  |      1     |
+                         +---------+------------+
+
+                       Table 2: Timestamp multiplier
+*/
+
+#include "rtp_profile.h"
+
+#include <stdio.h>
+#include <string.h>
+
+struct lrtp_profile_opus_t {
+  struct lrtp_profile_t profile; // 'Inherit' lrtp_profile_t
+};
+
+int profile_opus_pack(struct lrtp_profile_t *profile,
+                      const char *frame, size_t framesize,
+                      RTP &rtp)
+{
+  //struct lrtp_profile_opus_t *p = (struct lrtp_profile_opus_t *)profile;
+  (void)profile;
+
+  rtp.setPayload(frame, framesize);
+  rtp.setValid(true);
+
+  return framesize;
+}
+
+int profile_opus_unpack(struct lrtp_profile_t *profile,
+                        const RTP &rtp,
+                        std::list<outputframe_t *> &framelist)
+{
+  struct lrtp_profile_opus_t *p = (struct lrtp_profile_opus_t *)profile;
+
+  outputframe_t *of = new outputframe_t();
+  of->size = rtp.payloadSize();
+  char *buf = (char*)malloc(of->size);
+  of->size = rtp.payload(buf, of->size);
+  of->data = buf;
+
+  framelist.push_back(of);
+
+  return 0;
+}
+
+void profile_opus_destroy(struct lrtp_profile_t *profile)
+{
+  struct lrtp_profile_opus_t *p = (struct lrtp_profile_opus_t *)profile;
+  delete p;
+}
+
+struct lrtp_profile_t *profile_opus_create(struct lrtp_t *lrtp,
+                                          unsigned int csrc,
+                                          va_list vp)
+{
+  struct lrtp_profile_opus_t *p = new struct lrtp_profile_opus_t;
+
+  p->profile.pack = profile_opus_pack;
+  p->profile.unpack = profile_opus_unpack;
+  p->profile.destroy = profile_opus_destroy;
+
+  while(true) {
+    int type = va_arg(vp, int);
+    if(type == OPTION_END) break;
+
+    switch(type) {
+    default:
+      // TODO: Unknown arg type
+      break;
+    }
+  }
+
+  return &p->profile;
+}