tizen 2.4 releasetizen_2.4_mobile_releasesubmit/tizen_2.4/20151028.064339accepted/tizen/2.4/mobile/20151029.032704accepted/tizen_2.4_mobile

1 files changed, 287 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpsbcdepay.c b/gst/rtp/gstrtpsbcdepay.c
new file mode 100755
index 0000000..28c00fc
--- /dev/null
+++ b/gst/rtp/gstrtpsbcdepay.c
@@ -0,0 +1,287 @@
+/*
+ * GStreamer RTP SBC depayloader
+ *
+ * Copyright (C) 2012  Collabora Ltd.
+ *   @author: Arun Raghavan <arun.raghavan@collabora.co.uk>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpsbcdepay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpsbcdepay_debug);
+#define GST_CAT_DEFAULT (rtpsbcdepay_debug)
+
+static GstStaticPadTemplate gst_rtp_sbc_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-sbc, "
+        "rate = (int) { 16000, 32000, 44100, 48000 }, "
+        "channels = (int) [ 1, 2 ], "
+        "mode = (string) { mono, dual, stereo, joint }, "
+        "blocks = (int) { 4, 8, 12, 16 }, "
+        "subbands = (int) { 4, 8 }, "
+        "allocation-method = (string) { snr, loudness }, "
+        "bitpool = (int) [ 2, 64 ]")
+    );
+
+static GstStaticPadTemplate gst_rtp_sbc_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) { 16000, 32000, 44100, 48000 },"
+        "encoding-name = (string) SBC")
+    );
+
+#define gst_rtp_sbc_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
+
+static void gst_rtp_sbc_depay_finalize (GObject * object);
+
+static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
+    GstCaps * caps);
+static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
+    GstBuffer * in);
+
+static void
+gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
+{
+  GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
+      GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->finalize = gst_rtp_sbc_depay_finalize;
+
+  gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
+  gstbasertpdepayload_class->process = gst_rtp_sbc_depay_process;
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_sbc_depay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_sbc_depay_sink_template));
+
+  GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
+      "SBC Audio RTP Depayloader");
+
+  gst_element_class_set_static_metadata (element_class,
+      "RTP SBC audio depayloader",
+      "Codec/Depayloader/Network/RTP",
+      "Extracts SBC audio from RTP packets",
+      "Arun Raghavan <arun.raghavan@collabora.co.uk>");
+}
+
+static void
+gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
+{
+  rtpsbcdepay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_sbc_depay_finalize (GObject * object)
+{
+  GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
+
+  gst_object_unref (depay->adapter);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
+ * simple way to consolidate the two. This is best done by moving the function
+ * to the codec-utils library in gst-plugins-base when these elements move to
+ * GStreamer. */
+static int
+gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
+    gint size, int *framelen, int *samples)
+{
+  int blocks, channel_mode, channels, subbands, bitpool;
+  int length;
+
+  if (size < 3) {
+    /* Not enough data for the header */
+    return -1;
+  }
+
+  /* Sanity check */
+  if (data[0] != 0x9c) {
+    GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
+    return -2;
+  }
+
+  blocks = (data[1] >> 4) & 0x3;
+  blocks = (blocks + 1) * 4;
+  channel_mode = (data[1] >> 2) & 0x3;
+  channels = channel_mode ? 2 : 1;
+  subbands = (data[1] & 0x1);
+  subbands = (subbands + 1) * 4;
+  bitpool = data[2];
+
+  length = 4 + ((4 * subbands * channels) / 8);
+
+  if (channel_mode == 0 || channel_mode == 1) {
+    /* Mono || Dual channel */
+    length += ((blocks * channels * bitpool)
+        + 4 /* round up */ ) / 8;
+  } else {
+    /* Stereo || Joint stereo */
+    gboolean joint = (channel_mode == 3);
+
+    length += ((joint * subbands) + (blocks * bitpool)
+        + 4 /* round up */ ) / 8;
+  }
+
+  *framelen = length;
+  *samples = blocks * subbands;
+
+  return 0;
+}
+
+static gboolean
+gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
+{
+  GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
+  GstStructure *structure;
+  GstCaps *outcaps, *oldcaps;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
+    goto bad_caps;
+
+  outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
+      depay->rate, NULL);
+
+  gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);
+
+  oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
+  if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
+    /* Caps have changed, flush old data */
+    gst_adapter_clear (depay->adapter);
+  }
+
+  gst_caps_unref (outcaps);
+
+  return TRUE;
+
+bad_caps:
+  GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
+      GST_PTR_FORMAT, caps);
+  return FALSE;
+}
+
+static GstBuffer *
+gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstBuffer * in)
+{
+  GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
+  GstBuffer *data = NULL;
+  GstRTPBuffer rtp = { NULL };
+
+  gboolean fragment, start, last;
+  guint8 nframes;
+  guint8 *payload;
+  guint payload_len;
+
+  gst_rtp_buffer_map (in, GST_MAP_READ, &rtp);
+
+  GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
+      gst_buffer_get_size (in));
+
+  if (gst_rtp_buffer_get_marker (&rtp)) {
+    /* Marker isn't supposed to be set */
+    GST_WARNING_OBJECT (depay, "Marker bit was set");
+    goto bad_packet;
+  }
+
+  payload = gst_rtp_buffer_get_payload (&rtp);
+  payload_len = gst_rtp_buffer_get_payload_len (&rtp);
+
+  fragment = payload[0] & 0x80;
+  start = payload[0] & 0x40;
+  last = payload[0] & 0x20;
+  nframes = payload[0] & 0x0f;
+
+  payload += 1;
+  payload_len -= 1;
+
+  data = gst_rtp_buffer_get_payload_subbuffer (&rtp, 1, -1);
+
+  if (fragment) {
+    /* Got a packet with a fragment */
+    GST_LOG_OBJECT (depay, "Got fragment");
+
+    if (start && gst_adapter_available (depay->adapter)) {
+      GST_WARNING_OBJECT (depay, "Missing last fragment");
+      gst_adapter_clear (depay->adapter);
+
+    } else if (!start && !gst_adapter_available (depay->adapter)) {
+      GST_WARNING_OBJECT (depay, "Missing start fragment");
+      gst_buffer_unref (data);
+      data = NULL;
+      goto out;
+    }
+
+    gst_adapter_push (depay->adapter, data);
+
+    if (last) {
+      data = gst_adapter_take_buffer (depay->adapter,
+          gst_adapter_available (depay->adapter));
+    } else
+      data = NULL;
+
+  } else {
+    /* !fragment */
+    gint framelen, samples;
+
+    GST_LOG_OBJECT (depay, "Got %d frames", nframes);
+
+    if (gst_rtp_sbc_depay_get_params (depay, payload,
+            payload_len, &framelen, &samples) < 0) {
+      gst_adapter_clear (depay->adapter);
+      goto bad_packet;
+    }
+
+    GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);
+
+    if (nframes * framelen > (gint) payload_len) {
+      GST_WARNING_OBJECT (depay, "Short packet");
+      goto bad_packet;
+    } else if (nframes * framelen < (gint) payload_len) {
+      GST_WARNING_OBJECT (depay, "Junk at end of packet");
+    }
+  }
+
+out:
+  gst_rtp_buffer_unmap (&rtp);
+  return data;
+
+bad_packet:
+  GST_ELEMENT_WARNING (depay, STREAM, DECODE,
+      ("Received invalid RTP payload, dropping"), (NULL));
+  goto out;
+}
+
+gboolean
+gst_rtp_sbc_depay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpsbcdepay", GST_RANK_SECONDARY,
+      GST_TYPE_RTP_SBC_DEPAY);
+}