gst/rtp/: Added MP4A-LATM payloader to match the depayloader.

Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type),
(gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init),
(gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize),
(gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps),
(gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer),
(gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init):
* gst/rtp/gstrtpmp4apay.h:
Added MP4A-LATM payloader to match the depayloader.

1 files changed, 467 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c
new file mode 100644
index 00000000..b7a70722
--- /dev/null
+++ b/gst/rtp/gstrtpmp4apay.c
@@ -0,0 +1,467 @@
+/* GStreamer
+ * Copyright (C) <2008> 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 "gstrtpmp4apay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
+#define GST_CAT_DEFAULT (rtpmp4apay_debug)
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_mp4apay_details =
+GST_ELEMENT_DETAILS ("RTP packet payloader",
+    "Codec/Payloader/Network",
+    "Payload MPEG4 audio as RTP packets (RFC 3016)",
+    "Wim Taymans <wim.taymans@gmail.com>");
+
+static GstStaticPadTemplate gst_rtp_mp4a_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg," "mpegversion=(int) 4")
+    );
+
+static GstStaticPadTemplate gst_rtp_mp4a_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) [1, MAX ], "
+        "encoding-name = (string) \"MP4A-LATM\""
+        /* All optional parameters
+         *
+         * "cpresent = (string) \"0\""
+         * "config=" 
+         */
+    )
+    );
+
+
+static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass);
+static void gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass);
+static void gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay);
+static void gst_rtp_mp4a_pay_finalize (GObject * object);
+
+static gboolean gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+static GstStateChangeReturn gst_rtp_mp4a_pay_change_state (GstElement * element,
+    GstStateChange transition);
+static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload *
+    payload, GstBuffer * buffer);
+
+static GstBaseRTPPayloadClass *parent_class = NULL;
+
+static GType
+gst_rtp_mp4a_pay_get_type (void)
+{
+  static GType rtpmp4apay_type = 0;
+
+  if (!rtpmp4apay_type) {
+    static const GTypeInfo rtpmp4apay_info = {
+      sizeof (GstRtpMP4APayClass),
+      (GBaseInitFunc) gst_rtp_mp4a_pay_base_init,
+      NULL,
+      (GClassInitFunc) gst_rtp_mp4a_pay_class_init,
+      NULL,
+      NULL,
+      sizeof (GstRtpMP4APay),
+      0,
+      (GInstanceInitFunc) gst_rtp_mp4a_pay_init,
+    };
+
+    rtpmp4apay_type =
+        g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4APay",
+        &rtpmp4apay_info, 0);
+  }
+  return rtpmp4apay_type;
+}
+
+static void
+gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_mp4a_pay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template));
+
+  gst_element_class_set_details (element_class, &gst_rtp_mp4apay_details);
+}
+
+static void
+gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
+
+  gstelement_class->change_state = gst_rtp_mp4a_pay_change_state;
+
+  gstbasertppayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
+
+  GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
+      "MP4A-LATM RTP Payloader");
+}
+
+static void
+gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
+{
+  rtpmp4apay->rate = 90000;
+  rtpmp4apay->profile = g_strdup ("1");
+}
+
+static void
+gst_rtp_mp4a_pay_finalize (GObject * object)
+{
+  GstRtpMP4APay *rtpmp4apay;
+
+  rtpmp4apay = GST_RTP_MP4A_PAY (object);
+
+  g_free (rtpmp4apay->params);
+  rtpmp4apay->params = NULL;
+
+  if (rtpmp4apay->config)
+    gst_buffer_unref (rtpmp4apay->config);
+  rtpmp4apay->config = NULL;
+
+  g_free (rtpmp4apay->profile);
+  rtpmp4apay->profile = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static unsigned sampling_table[16] = {
+  96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+  16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+static gboolean
+gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
+    GstBuffer * buffer)
+{
+  guint8 *data;
+  guint size;
+  guint8 objectType;
+  guint8 samplingIdx;
+  guint8 channelCfg;
+
+  data = GST_BUFFER_DATA (buffer);
+  size = GST_BUFFER_SIZE (buffer);
+
+  if (size < 2)
+    goto too_short;
+
+  /* any object type is fine, we need to copy it to the profile-level-id field. */
+  objectType = (data[0] & 0xf8) >> 3;
+  if (objectType == 0)
+    goto invalid_object;
+
+  samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
+  /* only fixed values for now */
+  if (samplingIdx > 12 && samplingIdx != 15)
+    goto wrong_freq;
+
+  channelCfg = ((data[1] & 0x78) >> 3);
+  if (channelCfg > 7)
+    goto wrong_channels;
+
+  /* rtp rate depends on sampling rate of the audio */
+  if (samplingIdx == 15) {
+    if (size < 5)
+      goto too_short;
+
+    /* index of 15 means we get the rate in the next 24 bits */
+    rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
+        ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
+  } else {
+    /* else use the rate from the table */
+    rtpmp4apay->rate = sampling_table[samplingIdx];
+  }
+  /* extra rtp params contain the number of channels */
+  g_free (rtpmp4apay->params);
+  rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
+  /* audio stream type */
+  rtpmp4apay->streamtype = "5";
+  /* profile */
+  g_free (rtpmp4apay->profile);
+  rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
+
+  GST_DEBUG_OBJECT (rtpmp4apay,
+      "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
+      samplingIdx, rtpmp4apay->rate, channelCfg);
+
+  return TRUE;
+
+  /* ERROR */
+too_short:
+  {
+    GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
+        (NULL), ("config string too short, expected 2 bytes, got %d", size));
+    return FALSE;
+  }
+invalid_object:
+  {
+    GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
+        (NULL), ("invalid object type 0"));
+    return FALSE;
+  }
+wrong_freq:
+  {
+    GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
+        (NULL), ("unsupported frequency index %d", samplingIdx));
+    return FALSE;
+  }
+wrong_channels:
+  {
+    GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
+        (NULL), ("unsupported number of channels %d, must < 8", channelCfg));
+    return FALSE;
+  }
+}
+
+static void
+gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
+{
+  gchar *config;
+  GValue v = { 0 };
+
+  g_value_init (&v, GST_TYPE_BUFFER);
+  gst_value_set_buffer (&v, rtpmp4apay->config);
+  config = gst_value_serialize (&v);
+
+  gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
+      "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
+
+  g_value_unset (&v);
+  g_free (config);
+}
+
+static gboolean
+gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  GstRtpMP4APay *rtpmp4apay;
+  GstStructure *structure;
+  const GValue *codec_data;
+
+  rtpmp4apay = GST_RTP_MP4A_PAY (payload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  codec_data = gst_structure_get_value (structure, "codec_data");
+  if (codec_data) {
+    GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
+    if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
+      GstBuffer *buffer, *cbuffer;
+      guint8 *config;
+      guint8 *data;
+      guint size, i;
+      gboolean res;
+
+      buffer = gst_value_get_buffer (codec_data);
+      GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
+
+      /* parse buffer */
+      res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
+
+      if (!res)
+        goto config_failed;
+
+      size = GST_BUFFER_SIZE (buffer);
+      data = GST_BUFFER_DATA (buffer);
+
+      /* make the StreamMuxConfig, we need 15 bits for the header */
+      config = g_malloc0 (size + 2);
+
+      /* Create StreamMuxConfig according to ISO/IEC 14496-3:
+       *
+       * audioMuxVersion           == 0 (1 bit)
+       * allStreamsSameTimeFraming == 1 (1 bit)
+       * numSubFrames              == numSubFrames (6 bits)
+       * numProgram                == 0 (4 bits)
+       * numLayer                  == 0 (3 bits)
+       */
+      config[0] = 0x40;
+      config[1] = 0x00;
+
+      /* append the config bits, shifting them 1 bit left */
+      for (i = 0; i < size; i++) {
+        config[i + 1] |= ((data[i] & 0x80) >> 7);
+        config[i + 2] |= ((data[i] & 0x7f) << 1);
+      }
+
+      cbuffer = gst_buffer_new ();
+      GST_BUFFER_DATA (cbuffer) = config;
+      GST_BUFFER_MALLOCDATA (cbuffer) = config;
+      GST_BUFFER_SIZE (cbuffer) = size + 2;
+
+      /* now we can configure the buffer */
+      if (rtpmp4apay->config)
+        gst_buffer_unref (rtpmp4apay->config);
+      rtpmp4apay->config = cbuffer;
+    }
+  }
+
+  gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM",
+      rtpmp4apay->rate);
+
+  gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
+
+  return TRUE;
+
+  /* ERRORS */
+config_failed:
+  {
+    GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
+    return FALSE;
+  }
+}
+
+/* we expect buffers as exactly one complete AU
+ */
+static GstFlowReturn
+gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRtpMP4APay *rtpmp4apay;
+  GstFlowReturn ret;
+  GstBuffer *outbuf;
+  guint count, mtu, size;
+  guint8 *data;
+  gboolean fragmented;
+
+  ret = GST_FLOW_OK;
+
+  rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
+
+  size = GST_BUFFER_SIZE (buffer);
+  data = GST_BUFFER_DATA (buffer);
+
+  fragmented = FALSE;
+  mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4apay);
+
+  while (size > 0) {
+    guint towrite;
+    guint8 *payload;
+    guint payload_len;
+    guint packet_len;
+
+    /* this will be the total lenght of the packet */
+    packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
+
+    if (!fragmented) {
+      /* first packet calculate space for the packet including the header */
+      count = size;
+      while (count >= 0xff) {
+        packet_len++;
+        count -= 0xff;
+      }
+      packet_len++;
+    }
+
+    /* 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);
+
+    GST_DEBUG_OBJECT (rtpmp4apay,
+        "avail %d, towrite %d, packet_len %d, payload_len %d", size, towrite,
+        packet_len, payload_len);
+
+    /* create buffer to hold the payload. */
+    outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+    /* copy payload */
+    payload = gst_rtp_buffer_get_payload (outbuf);
+
+    if (!fragmented) {
+      /* first packet write the header */
+      count = size;
+      while (count >= 0xff) {
+        *payload++ = 0xff;
+        payload_len--;
+        count -= 0xff;
+      }
+      *payload++ = count;
+      payload_len--;
+    }
+
+    /* copy data to payload */
+    memcpy (payload, data, payload_len);
+    data += payload_len;
+    size -= payload_len;
+
+    /* marker only if the packet is complete */
+    gst_rtp_buffer_set_marker (outbuf, size == 0);
+
+    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
+
+    fragmented = TRUE;
+  }
+  return ret;
+}
+
+static GstStateChangeReturn
+gst_rtp_mp4a_pay_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRtpMP4APay *rtpmp4apay;
+  GstStateChangeReturn ret;
+
+  rtpmp4apay = GST_RTP_MP4A_PAY (element);
+
+  switch (transition) {
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    default:
+      break;
+  }
+  return ret;
+}
+
+
+gboolean
+gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpmp4apay",
+      GST_RANK_NONE, GST_TYPE_RTP_MP4A_PAY);
+}