reverting rtp patches to fix freeze break on -base as explained on the list

Original commit message from CVS:
reverting rtp patches to fix freeze break on -base as explained on the list

1 files changed, 121 insertions, 9 deletions

diff --git a/gst/rtp/gstrtppcmupay.c b/gst/rtp/gstrtppcmupay.c
index 02ba1f84..ffb220b5 100644
--- a/gst/rtp/gstrtppcmupay.c
+++ b/gst/rtp/gstrtppcmupay.c
@@ -50,9 +50,17 @@ GST_STATIC_PAD_TEMPLATE ("src",
 
 static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
     GstCaps * caps);
+static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
+    payload, GstBuffer * buffer);
+static void gst_rtp_pcmu_pay_finalize (GObject * object);
 
-GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
-    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD);
+
+/* The lower limit for number of octet to put in one packet
+ * (clock-rate=8000, octet-per-sample=1). The default 80 is equal
+ * to to 10msec (see RFC3551) */
+#define GST_RTP_PCMU_MIN_PTIME_OCTETS   80
 
 static void
 gst_rtp_pcmu_pay_base_init (gpointer klass)
@@ -78,24 +86,30 @@ gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
   gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   parent_class = g_type_class_peek_parent (klass);
+  gobject_class->finalize = gst_rtp_pcmu_pay_finalize;
 
   gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
 }
 
 static void
 gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
 {
-  GstBaseRTPAudioPayload *basertpaudiopayload;
+  rtppcmupay->adapter = gst_adapter_new ();
+  GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
+}
 
-  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
+static void
+gst_rtp_pcmu_pay_finalize (GObject * object)
+{
+  GstRtpPcmuPay *rtppcmupay;
 
-  GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
+  rtppcmupay = GST_RTP_PCMU_PAY (object);
 
-  /* tell basertpaudiopayload that this is a sample based codec */
-  gst_basertpaudiopayload_set_sample_based (basertpaudiopayload);
+  g_object_unref (rtppcmupay->adapter);
+  rtppcmupay->adapter = NULL;
 
-  /* octet-per-sample is 1 for PCM */
-  gst_basertpaudiopayload_set_sample_options (basertpaudiopayload, 1);
+  G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
 static gboolean
@@ -109,6 +123,104 @@ gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   return TRUE;
 }
 
+static GstFlowReturn
+gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay)
+{
+  guint avail;
+  GstBuffer *outbuf;
+  GstFlowReturn ret;
+  guint maxptime_octets = G_MAXUINT;
+  guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
+
+  if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
+    /* calculate octet count with:
+       maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
+    maxptime_octets =
+        GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime *
+        GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate / GST_SECOND;
+  }
+
+  /* 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.  */
+  avail = gst_adapter_available (rtppcmupay->adapter);
+
+  ret = GST_FLOW_OK;
+
+  while (avail >= minptime_octets) {
+    guint8 *payload;
+    guint8 *data;
+    guint payload_len;
+    guint packet_len;
+
+    /* fill one MTU or all available bytes */
+    payload_len =
+        MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
+        avail);
+
+    /* this will be the total lenght of the packet */
+    packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
+
+    /* create buffer to hold the payload */
+    outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+    /* copy payload */
+    gst_rtp_buffer_set_payload_type (outbuf,
+        GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
+    payload = gst_rtp_buffer_get_payload (outbuf);
+    data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
+    memcpy (payload, data, payload_len);
+    gst_adapter_flush (rtppcmupay->adapter, payload_len);
+
+    avail -= payload_len;
+
+    GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
+    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRtpPcmuPay *rtppcmupay;
+  guint size, packet_len, avail;
+  GstFlowReturn ret;
+  GstClockTime duration;
+
+  rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
+
+  size = GST_BUFFER_SIZE (buffer);
+  duration = GST_BUFFER_TIMESTAMP (buffer);
+
+  avail = gst_adapter_available (rtppcmupay->adapter);
+  if (avail == 0) {
+    rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+    rtppcmupay->duration = 0;
+  }
+
+  /* get packet length of data and see if we exceeded MTU. */
+  packet_len = gst_rtp_buffer_calc_packet_len (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, rtppcmupay->duration + duration)) {
+    ret = gst_rtp_pcmu_pay_flush (rtppcmupay);
+    rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+    rtppcmupay->duration = 0;
+  } else {
+    ret = GST_FLOW_OK;
+  }
+
+  gst_adapter_push (rtppcmupay->adapter, buffer);
+  rtppcmupay->duration += duration;
+
+  return ret;
+}
+
 gboolean
 gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
 {