gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.

Original commit message from CVS:
Based on patch by: Stefan Kost  <ensonic@users.sf.net>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
(gst_rtp_mp4a_depay_get_property),
(gst_rtp_mp4a_depay_change_state),
(gst_rtp_mp4a_depay_plugin_init):
* gst/rtp/gstrtpmp4adepay.h:
Added MP4A-LATM depayloader. Fixes #417792.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Fixup depayloader, setting codec_data, using more efficient adaptor and
rtpbuffer handling.
* gst/rtsp/URLS:
Add url to test above.

1 files changed, 367 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c
new file mode 100644
index 000000000..395db196e
--- /dev/null
+++ b/gst/rtp/gstrtpmp4adepay.c
@@ -0,0 +1,367 @@
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
+ *               <2007> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License version 2 as published by the Free Software Foundation.
+ *
+ * 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 <string.h>
+#include "gstrtpmp4adepay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
+#define GST_CAT_DEFAULT (rtpmp4adepay_debug)
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_mp4adepay_details =
+GST_ELEMENT_DETAILS ("RTP packet parser",
+    "Codec/Depayloader/Network",
+    "Extracts MPEG4 audio from RTP packets (RFC 3016)",
+    "Nokia Corporation (contact <stefan.kost@nokia.com>), "
+    "Wim Taymans <wim@fluendo.com>");
+
+/* RtpMP4ADepay signals and args */
+enum
+{
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+};
+
+static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg,"
+        "mpegversion = (int) 4," "framed = (boolean) false")
+    );
+
+static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "clock-rate = (int) [1, MAX ], "
+        "encoding-name = (string) \"MP4A-LATM\""
+        /* All optional parameters
+         *
+         * "profile-level-id=[1,MAX]"
+         * "config=" 
+         */
+    )
+    );
+
+GST_BOILERPLATE (GstRtpMP4ADepay, gst_rtp_mp4a_depay, GstBaseRTPDepayload,
+    GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void gst_rtp_mp4a_depay_finalize (GObject * object);
+
+static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload,
+    GstCaps * caps);
+static GstBuffer *gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload,
+    GstBuffer * buf);
+
+static void gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
+    element, GstStateChange transition);
+
+
+static void
+gst_rtp_mp4a_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_mp4a_depay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
+
+  gst_element_class_set_details (element_class, &gst_rtp_mp4adepay_details);
+}
+
+static void
+gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
+  gobject_class->set_property = gst_rtp_mp4a_depay_set_property;
+  gobject_class->get_property = gst_rtp_mp4a_depay_get_property;
+
+  gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
+
+  gstbasertpdepayload_class->process = gst_rtp_mp4a_depay_process;
+  gstbasertpdepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
+      "MPEG4 audio RTP Depayloader");
+}
+
+static void
+gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay,
+    GstRtpMP4ADepayClass * klass)
+{
+  rtpmp4adepay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_mp4a_depay_finalize (GObject * object)
+{
+  GstRtpMP4ADepay *rtpmp4adepay;
+
+  rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
+
+  g_object_unref (rtpmp4adepay->adapter);
+  rtpmp4adepay->adapter = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+
+static gboolean
+gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+  GstStructure *structure;
+  GstRtpMP4ADepay *rtpmp4adepay;
+  GstCaps *srccaps;
+  const gchar *str;
+  gint clock_rate = 90000;      /* default */
+  gint object_type = 2;         /* AAC LC default */
+  gint channels = 2;            /* default */
+
+  rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (gst_structure_has_field (structure, "clock-rate"))
+    gst_structure_get_int (structure, "clock-rate", &clock_rate);
+  depayload->clock_rate = clock_rate;
+
+  if (gst_structure_has_field (structure, "object"))
+    gst_structure_get_int (structure, "object", &object_type);
+
+  srccaps = gst_caps_new_simple ("audio/mpeg",
+      "mpegversion", G_TYPE_INT, 4,
+      "framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels, NULL);
+
+  if ((str = gst_structure_get_string (structure, "config"))) {
+    GValue v = { 0 };
+
+    g_value_init (&v, GST_TYPE_BUFFER);
+    if (gst_value_deserialize (&v, str)) {
+      GstBuffer *buffer;
+      guint8 *data;
+      guint size;
+      gint i;
+
+      buffer = gst_value_get_buffer (&v);
+      gst_buffer_ref (buffer);
+      g_value_unset (&v);
+
+      data = GST_BUFFER_DATA (buffer);
+      size = GST_BUFFER_SIZE (buffer);
+      if (size < 2) {
+        GST_WARNING_OBJECT (depayload, "config too short (%d < 2)", size);
+        goto bad_config;
+      }
+
+      /* Parse StreamMuxConfig according to ISO/IEC 14496-3:
+       *
+       * audioMuxVersion           == 0 (1 bit)
+       * allStreamsSameTimeFraming == 1 (1 bit)
+       * numSubFrames              == 0 (6 bits)
+       * numProgram                == 0 (4 bits)
+       * numLayer                  == 0 (3 bits)
+       *
+       * We only require audioMuxVersion == 0;
+       *
+       * The remaining bit of the second byte and the rest of the bits are used
+       * for audioSpecificConfig which we need to set in codec_info.
+       */
+      if ((data[0] & 0x80) != 0x00) {
+        GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
+        goto bad_config;
+      }
+
+      /* shift rest of string 15 bits down */
+      size -= 2;
+      for (i = 0; i < size; i++) {
+        data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
+      }
+      /* last bit, this is probably not needed. */
+      data[i] = ((data[i + 1] & 1) << 7);
+      GST_BUFFER_SIZE (buffer) = size + 1;
+
+      gst_caps_set_simple (srccaps,
+          "codec_data", GST_TYPE_BUFFER, buffer, NULL);
+      gst_buffer_unref (buffer);
+    } else {
+      g_warning ("cannot convert config to buffer");
+    }
+  }
+bad_config:
+  gst_pad_set_caps (depayload->srcpad, srccaps);
+  gst_caps_unref (srccaps);
+
+  return TRUE;
+}
+
+static GstBuffer *
+gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+  GstRtpMP4ADepay *rtpmp4adepay;
+  GstBuffer *outbuf;
+
+  rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
+
+  if (!gst_rtp_buffer_validate (buf))
+    goto bad_packet;
+
+  /* flush remaining data on discont */
+  if (GST_BUFFER_IS_DISCONT (buf))
+    gst_adapter_clear (rtpmp4adepay->adapter);
+
+  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+
+  gst_adapter_push (rtpmp4adepay->adapter, outbuf);
+
+  /* RTP marker bit indicates the last packet of the AudioMuxElement => create
+   * and push a buffer */
+  if (gst_rtp_buffer_get_marker (buf)) {
+    guint avail;
+    guint latm_header_len;
+    guint data_len;
+    guint8 *data;
+
+    avail = gst_adapter_available (rtpmp4adepay->adapter);
+
+    outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
+
+    /* determine payload length and set buffer data pointer accordingly */
+    /* FIXME, check for overrun */
+    latm_header_len = 0;
+    data_len = 0;
+    data = GST_BUFFER_DATA (outbuf);
+    do {
+      data_len += data[latm_header_len];
+    } while (data[latm_header_len++] == 0xff);
+
+    /* just a check that lengths match, possibly there can be more than one
+     * audioMuxElement in the payload? */
+    if ((data_len + latm_header_len) != avail) {
+      GST_WARNING_OBJECT (depayload, "not all payload consumed");
+    }
+
+    GST_BUFFER_SIZE (outbuf) = avail - latm_header_len;
+    GST_BUFFER_DATA (outbuf) += latm_header_len;
+
+    gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
+
+    GST_DEBUG ("gst_rtp_mp4a_depay_process: pushing buffer of size %d",
+        GST_BUFFER_SIZE (outbuf));
+
+    return outbuf;
+  }
+  return NULL;
+
+bad_packet:
+  {
+    GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
+        ("Packet did not validate"), (NULL));
+    return NULL;
+  }
+}
+
+static void
+gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRtpMP4ADepay *rtpmp4adepay;
+
+  rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRtpMP4ADepay *rtpmp4adepay;
+
+  rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_rtp_mp4a_depay_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstRtpMP4ADepay *rtpmp4adepay;
+  GstStateChangeReturn ret;
+
+  rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      gst_adapter_clear (rtpmp4adepay->adapter);
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    default:
+      break;
+  }
+  return ret;
+}
+
+gboolean
+gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpmp4adepay",
+      GST_RANK_NONE, GST_TYPE_RTP_MP4A_DEPAY);
+}