rtpac3pay: add AC3 payloader

gst/rtp/Makefile.am

@@ -6,6 +6,7 @@ libgstrtp_la_SOURCES = \
 	gstrtpchannels.c \
 	gstrtpdepay.c \
 	gstrtpac3depay.c \
+	gstrtpac3pay.c \
 	gstrtpbvdepay.c \
 	gstrtpbvpay.c \
 	gstrtpceltdepay.c \
@@ -94,6 +95,7 @@ noinst_HEADERS =			\
 		 gstrtpL16depay.h 	\
 		 gstrtpL16pay.h 	\
 		 gstrtpac3depay.h 	\
+		 gstrtpac3pay.h 	\
 		 gstrtpbvdepay.h 	\
 		 gstrtpbvpay.h 	        \
 		 gstrtpceltpay.h	\

gst/rtp/gstrtp.c

@@ -23,6 +23,7 @@
 
 #include "gstrtpdepay.h"
 #include "gstrtpac3depay.h"
+#include "gstrtpac3pay.h"
 #include "gstrtpbvdepay.h"
 #include "gstrtpbvpay.h"
 #include "gstrtpceltdepay.h"
@@ -97,6 +98,9 @@ plugin_init (GstPlugin * plugin)
   if (!gst_rtp_ac3_depay_plugin_init (plugin))
     return FALSE;
 
+  if (!gst_rtp_ac3_pay_plugin_init (plugin))
+    return FALSE;
+
   if (!gst_rtp_bv_depay_plugin_init (plugin))
     return FALSE;
 

gst/rtp/gstrtpac3pay.c

+/* 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 <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpac3pay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug);
+#define GST_CAT_DEFAULT (rtpac3pay_debug)
+
+static GstStaticPadTemplate gst_rtp_ac3_pay_sink_template =
+    GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/ac3; " "audio/x-ac3; ")
+    );
+
+static GstStaticPadTemplate gst_rtp_ac3_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) { 32000, 44100, 48000 }, "
+        "encoding-name = (string) \"AC3\"")
+    );
+
+static void gst_rtp_ac3_pay_finalize (GObject * object);
+
+static GstStateChangeReturn gst_rtp_ac3_pay_change_state (GstElement * element,
+    GstStateChange transition);
+
+static gboolean gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+static gboolean gst_rtp_ac3_pay_handle_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay);
+static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * payload,
+    GstBuffer * buffer);
+
+GST_BOILERPLATE (GstRtpAC3Pay, gst_rtp_ac3_pay, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD)
+
+     static void gst_rtp_ac3_pay_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_ac3_pay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_ac3_pay_sink_template));
+
+  gst_element_class_set_details_simple (element_class,
+      "RTP AC3 audio payloader", "Codec/Payloader/Network/RTP",
+      "Payload AC3 audio as RTP packets (RFC 4184)",
+      "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gobject_class->finalize = gst_rtp_ac3_pay_finalize;
+
+  gstelement_class->change_state = gst_rtp_ac3_pay_change_state;
+
+  gstbasertppayload_class->set_caps = gst_rtp_ac3_pay_setcaps;
+  gstbasertppayload_class->handle_event = gst_rtp_ac3_pay_handle_event;
+  gstbasertppayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer;
+
+  GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0,
+      "AC3 Audio RTP Depayloader");
+}
+
+static void
+gst_rtp_ac3_pay_init (GstRtpAC3Pay * rtpac3pay, GstRtpAC3PayClass * klass)
+{
+  rtpac3pay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_ac3_pay_finalize (GObject * object)
+{
+  GstRtpAC3Pay *rtpac3pay;
+
+  rtpac3pay = GST_RTP_AC3_PAY (object);
+
+  g_object_unref (rtpac3pay->adapter);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_rtp_ac3_pay_reset (GstRtpAC3Pay * pay)
+{
+  pay->first_ts = -1;
+  pay->duration = 0;
+  gst_adapter_clear (pay->adapter);
+  GST_DEBUG_OBJECT (pay, "reset depayloader");
+}
+
+static gboolean
+gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  gboolean res;
+  gint rate;
+  GstStructure *structure;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (!gst_structure_get_int (structure, "rate", &rate))
+    rate = 90000;               /* default */
+
+  gst_basertppayload_set_options (payload, "audio", TRUE, "AC3", rate);
+  res = gst_basertppayload_set_outcaps (payload, NULL);
+
+  return res;
+}
+
+static gboolean
+gst_rtp_ac3_pay_handle_event (GstPad * pad, GstEvent * event)
+{
+  GstRtpAC3Pay *rtpac3pay;
+
+  rtpac3pay = GST_RTP_AC3_PAY (gst_pad_get_parent (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+      /* make sure we push the last packets in the adapter on EOS */
+      gst_rtp_ac3_pay_flush (rtpac3pay);
+      break;
+    case GST_EVENT_FLUSH_STOP:
+      gst_rtp_ac3_pay_reset (rtpac3pay);
+      break;
+    default:
+      break;
+  }
+
+  gst_object_unref (rtpac3pay);
+
+  /* FALSE to let the parent handle the event as well */
+  return FALSE;
+}
+
+struct frmsize_s
+{
+  guint16 bit_rate;
+  guint16 frm_size[3];
+};
+
+static const struct frmsize_s frmsizecod_tbl[] = {
+  {32, {64, 69, 96}},
+  {32, {64, 70, 96}},
+  {40, {80, 87, 120}},
+  {40, {80, 88, 120}},
+  {48, {96, 104, 144}},
+  {48, {96, 105, 144}},
+  {56, {112, 121, 168}},
+  {56, {112, 122, 168}},
+  {64, {128, 139, 192}},
+  {64, {128, 140, 192}},
+  {80, {160, 174, 240}},
+  {80, {160, 175, 240}},
+  {96, {192, 208, 288}},
+  {96, {192, 209, 288}},
+  {112, {224, 243, 336}},
+  {112, {224, 244, 336}},
+  {128, {256, 278, 384}},
+  {128, {256, 279, 384}},
+  {160, {320, 348, 480}},
+  {160, {320, 349, 480}},
+  {192, {384, 417, 576}},
+  {192, {384, 418, 576}},
+  {224, {448, 487, 672}},
+  {224, {448, 488, 672}},
+  {256, {512, 557, 768}},
+  {256, {512, 558, 768}},
+  {320, {640, 696, 960}},
+  {320, {640, 697, 960}},
+  {384, {768, 835, 1152}},
+  {384, {768, 836, 1152}},
+  {448, {896, 975, 1344}},
+  {448, {896, 976, 1344}},
+  {512, {1024, 1114, 1536}},
+  {512, {1024, 1115, 1536}},
+  {576, {1152, 1253, 1728}},
+  {576, {1152, 1254, 1728}},
+  {640, {1280, 1393, 1920}},
+  {640, {1280, 1394, 1920}}
+};
+
+static GstFlowReturn
+gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
+{
+  guint avail, FT, NF, mtu;
+  GstBuffer *outbuf;
+  GstFlowReturn ret;
+
+  /* the data available in the adapter is either smaller
+   * than the MTU or bigger. In the case it is smaller, the complete
+   * adapter contents can be put in one packet. In the case the
+   * adapter has more than one MTU, we need to split the AC3 data
+   * over multiple packets. */
+  avail = gst_adapter_available (rtpac3pay->adapter);
+
+  ret = GST_FLOW_OK;
+
+  FT = 0;
+  /* number of frames */
+  NF = rtpac3pay->NF;
+
+  mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpac3pay);
+
+  GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
+
+  while (avail > 0) {
+    guint towrite;
+    guint8 *payload;
+    guint payload_len;
+    guint packet_len;
+
+    /* this will be the total length of the packet */
+    packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0);
+
+    /* fill one MTU or all available bytes */
+    towrite = MIN (packet_len, mtu);
+
+    /* this is the payload length */
+    payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
+
+    /* create buffer to hold the payload */
+    outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+    if (FT == 0) {
+      /* check if it all fits */
+      if (towrite < packet_len) {
+        guint maxlen;
+
+        GST_LOG_OBJECT (rtpac3pay, "we need to fragment");
+        /* check if we will be able to put at least 5/8th of the total
+         * frame in this first frame. */
+        if ((avail * 5) / 8 >= (payload_len - 2))
+          FT = 1;
+        else
+          FT = 2;
+        /* check how many fragments we will need */
+        maxlen = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
+        NF = (avail + maxlen - 1) / maxlen;
+      }
+    } else if (FT != 3) {
+      /* remaining fragment */
+      FT = 3;
+    }
+
+    /*
+     *  0                   1
+     *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+     * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     * |    MBZ    | FT|       NF      |
+     * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     *
+     * FT: 0: one or more complete frames
+     *     1: initial 5/8 fragment
+     *     2: initial fragment not 5/8
+     *     3: other fragment
+     * NF: amount of frames if FT = 0, else number of fragments.
+     */
+    GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF);
+    payload = gst_rtp_buffer_get_payload (outbuf);
+    payload[0] = (FT & 3);
+    payload[1] = NF;
+    payload_len -= 2;
+
+    gst_adapter_copy (rtpac3pay->adapter, &payload[2], 0, payload_len);
+    gst_adapter_flush (rtpac3pay->adapter, payload_len);
+
+    avail -= payload_len;
+    if (avail == 0)
+      gst_rtp_buffer_set_marker (outbuf, TRUE);
+
+    GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts;
+    GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
+
+    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRtpAC3Pay *rtpac3pay;
+  GstFlowReturn ret;
+  guint size, avail, left, NF;
+  guint8 *data, *p;
+  guint packet_len;
+  GstClockTime duration, timestamp;
+
+  rtpac3pay = GST_RTP_AC3_PAY (basepayload);
+
+  size = GST_BUFFER_SIZE (buffer);
+  data = GST_BUFFER_DATA (buffer);
+  duration = GST_BUFFER_DURATION (buffer);
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+  if (GST_BUFFER_IS_DISCONT (buffer)) {
+    GST_DEBUG_OBJECT (rtpac3pay, "DISCONT");
+    gst_rtp_ac3_pay_reset (rtpac3pay);
+  }
+
+  /* count the amount of incomming packets */
+  NF = 0;
+  left = size;
+  p = data;
+  while (TRUE) {
+    guint bsid, fscod, frmsizecod, frame_size;
+
+    if (left < 6)
+      break;
+
+    if (p[0] != 0x0b || p[1] != 0x77)
+      break;
+
+    bsid = p[5] >> 3;
+    if (bsid > 8)
+      break;
+
+    frmsizecod = p[4] & 0x3f;
+    fscod = p[4] >> 6;
+
+    GST_DEBUG_OBJECT (rtpac3pay, "fscod %u, %u", fscod, frmsizecod);
+
+    if (fscod >= 3 || frmsizecod >= 38)
+      break;
+
+    frame_size = frmsizecod_tbl[frmsizecod].frm_size[fscod] * 2;
+    if (frame_size > left)
+      break;
+
+    NF++;
+    GST_DEBUG_OBJECT (rtpac3pay, "found frame %u of size %u", NF, frame_size);
+
+    p += frame_size;
+    left -= frame_size;
+  }
+  if (NF == 0)
+    goto no_frames;
+
+  avail = gst_adapter_available (rtpac3pay->adapter);
+
+  /* get packet length of previous data and this new data,
+   * payload length includes a 4 byte header */
+  packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + size, 0, 0);
+
+  /* if this buffer is going to overflow the packet, flush what we
+   * have. */
+  if (gst_basertppayload_is_filled (basepayload,
+          packet_len, rtpac3pay->duration + duration)) {
+    ret = gst_rtp_ac3_pay_flush (rtpac3pay);
+    avail = 0;
+  } else {
+    ret = GST_FLOW_OK;
+  }
+
+  if (avail == 0) {
+    GST_DEBUG_OBJECT (rtpac3pay,
+        "first packet, save timestamp %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (timestamp));
+    rtpac3pay->first_ts = timestamp;
+    rtpac3pay->duration = 0;
+    rtpac3pay->NF = 0;
+  }
+
+  gst_adapter_push (rtpac3pay->adapter, buffer);
+  rtpac3pay->duration += duration;
+  rtpac3pay->NF += NF;
+
+  return ret;
+
+  /* ERRORS */
+no_frames:
+  {
+    GST_WARNING_OBJECT (rtpac3pay, "no valid AC3 frames found");
+    return GST_FLOW_OK;
+  }
+}
+
+static GstStateChangeReturn
+gst_rtp_ac3_pay_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRtpAC3Pay *rtpac3pay;
+  GstStateChangeReturn ret;
+
+  rtpac3pay = GST_RTP_AC3_PAY (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      gst_rtp_ac3_pay_reset (rtpac3pay);
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_rtp_ac3_pay_reset (rtpac3pay);
+      break;
+    default:
+      break;
+  }
+  return ret;
+}
+
+gboolean
+gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpac3pay",
+      GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_PAY);
+}

gst/rtp/gstrtpac3pay.h

+/* 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.
+ */
+
+#ifndef __GST_RTP_AC3_PAY_H__
+#define __GST_RTP_AC3_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertppayload.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_AC3_PAY \
+  (gst_rtp_ac3_pay_get_type())
+#define GST_RTP_AC3_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_AC3_PAY,GstRtpAC3Pay))
+#define GST_RTP_AC3_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_AC3_PAY,GstRtpAC3PayClass))
+#define GST_IS_RTP_AC3_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_AC3_PAY))
+#define GST_IS_RTP_AC3_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_AC3_PAY))
+
+typedef struct _GstRtpAC3Pay GstRtpAC3Pay;
+typedef struct _GstRtpAC3PayClass GstRtpAC3PayClass;
+
+struct _GstRtpAC3Pay
+{
+  GstBaseRTPPayload payload;
+
+  GstAdapter *adapter;
+  GstClockTime first_ts;
+  GstClockTime duration;
+  guint NF;
+};
+
+struct _GstRtpAC3PayClass
+{
+  GstBaseRTPPayloadClass parent_class;
+};
+
+GType gst_rtp_ac3_pay_get_type (void);
+
+gboolean gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_AC3_PAY_H__ */