audiortppay: use offsets for RTP timestamps

Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.

gst-libs/gst/rtp/gstbasertpaudiopayload.c

@@ -76,14 +76,20 @@ typedef gboolean (*GetLengthsFunc) (GstBaseRTPPayload * basepayload,
 /* function to convert bytes to a duration */
 typedef GstClockTime (*GetDurationFunc) (GstBaseRTPAudioPayload * payload,
     guint64 bytes);
+/* function to convert bytes to RTP timestamp */
+typedef guint32 (*GetRTPTimeFunc) (GstBaseRTPAudioPayload * payload,
+    guint64 bytes);
 
 struct _GstBaseRTPAudioPayloadPrivate
 {
   GetLengthsFunc get_lengths;
   GetDurationFunc get_duration;
+  GetRTPTimeFunc get_rtptime;
 
   GstAdapter *adapter;
   guint fragment_size;
+  gboolean discont;
+  guint64 offset;
 };
 
 
@@ -109,6 +115,14 @@ static GstClockTime
 gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
     payload, guint64 bytes);
 
+/* rtptime functions */
+static guint32
+gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload,
+    guint64 bytes);
+static guint32
+gst_base_rtp_audio_payload_get_sample_rtptime (GstBaseRTPAudioPayload *
+    payload, guint64 bytes);
+
 static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
     * payload, GstBuffer * buffer);
 
@@ -202,6 +216,8 @@ gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
       gst_base_rtp_audio_payload_get_frame_lengths;
   basertpaudiopayload->priv->get_duration =
       gst_base_rtp_audio_payload_get_frame_duration;
+  basertpaudiopayload->priv->get_rtptime =
+      gst_base_rtp_audio_payload_get_frame_rtptime;
 }
 
 /**
@@ -223,6 +239,8 @@ gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
       gst_base_rtp_audio_payload_get_sample_lengths;
   basertpaudiopayload->priv->get_duration =
       gst_base_rtp_audio_payload_get_sample_duration;
+  basertpaudiopayload->priv->get_rtptime =
+      gst_base_rtp_audio_payload_get_sample_rtptime;
 }
 
 /**
@@ -292,6 +310,32 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
   gst_adapter_clear (basertpaudiopayload->priv->adapter);
 }
 
+static void
+gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload,
+    GstBuffer * buffer, guint payload_len, GstClockTime timestamp)
+{
+  GstBaseRTPPayload *basepayload;
+
+  basepayload = GST_BASE_RTP_PAYLOAD_CAST (payload);
+
+  /* set payload type */
+  gst_rtp_buffer_set_payload_type (buffer, basepayload->pt);
+  /* set marker bit for disconts */
+  if (payload->priv->discont) {
+    GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT");
+    gst_rtp_buffer_set_marker (buffer, TRUE);
+    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
+    payload->priv->discont = FALSE;
+  }
+  GST_BUFFER_TIMESTAMP (buffer) = timestamp;
+
+  /* get the offset in bytes */
+  GST_BUFFER_OFFSET (buffer) =
+      payload->priv->get_rtptime (payload, payload->priv->offset);
+
+  payload->priv->offset += payload_len;
+}
+
 /**
  * gst_base_rtp_audio_payload_push:
  * @baseaudiopayload: a #GstBaseRTPPayload
@@ -324,12 +368,14 @@ gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
   /* create buffer to hold the payload */
   outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
 
+  /* set metadata */
+  gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
+      timestamp);
+
   /* copy payload */
-  gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
   payload = gst_rtp_buffer_get_payload (outbuf);
   memcpy (payload, data, payload_len);
 
-  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
   ret = gst_basertppayload_push (basepayload, outbuf);
 
   return ret;
@@ -345,6 +391,9 @@ gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
  * payload. Set the timestamp on the new buffer to @timestamp before pushing
  * the buffer downstream.
  *
+ * If @payload_len is -1, all pending bytes will be flushed. If @timestamp is
+ * -1, the timestamp will be calculated automatically.
+ *
  * Returns: a #GstFlowReturn
  *
  * Since: 0.10.25
@@ -357,22 +406,50 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
   GstBuffer *outbuf;
   guint8 *payload;
   GstFlowReturn ret;
+  GstAdapter *adapter;
+  guint64 distance;
+
+  adapter = baseaudiopayload->priv->adapter;
 
   basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
 
+  if (payload_len == -1)
+    payload_len = gst_adapter_available (adapter);
+
+  /* nothing to do, just return */
+  if (payload_len == 0)
+    return GST_FLOW_OK;
+
+  if (timestamp == -1) {
+    /* calculate the timestamp */
+    timestamp = gst_adapter_prev_timestamp (adapter, &distance);
+
+    GST_LOG_OBJECT (baseaudiopayload,
+        "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
+        GST_TIME_ARGS (timestamp), distance);
+
+    if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
+      /* convert the number of bytes since the last timestamp to time and add to
+       * the last seen timestamp */
+      timestamp +=
+          baseaudiopayload->priv->get_duration (baseaudiopayload, distance);
+    }
+  }
+
   GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
       payload_len, GST_TIME_ARGS (timestamp));
 
   /* create buffer to hold the payload */
   outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
 
-  /* copy payload */
+  /* set metadata */
-  gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
+  gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
-  payload = gst_rtp_buffer_get_payload (outbuf);
+      timestamp);
-  gst_adapter_copy (baseaudiopayload->priv->adapter, payload, 0, payload_len);
+
-  gst_adapter_flush (baseaudiopayload->priv->adapter, payload_len);
+  payload = gst_rtp_buffer_get_payload (outbuf);
+  gst_adapter_copy (adapter, payload, 0, payload_len);
+  gst_adapter_flush (adapter, payload_len);
 
-  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
   ret = gst_basertppayload_push (basepayload, outbuf);
 
   return ret;
@@ -441,6 +518,14 @@ gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
       payload->frame_size);
 }
 
+static guint32
+gst_base_rtp_audio_payload_get_frame_rtptime (GstBaseRTPAudioPayload * payload,
+    guint64 bytes)
+{
+  return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND,
+      payload->frame_size * GST_BASE_RTP_PAYLOAD_CAST (payload)->clock_rate);
+}
+
 static gboolean
 gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload *
     basepayload, guint * min_payload_len, guint * max_payload_len,
@@ -494,6 +579,13 @@ gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
       (GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size);
 }
 
+static guint32
+gst_base_rtp_audio_payload_get_sample_rtptime (GstBaseRTPAudioPayload *
+    payload, guint64 bytes)
+{
+  return (bytes * 8) / payload->sample_size;
+}
+
 static GstFlowReturn
 gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
     basepayload, GstBuffer * buffer)
@@ -506,6 +598,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
   guint max_payload_len;
   guint align;
   guint size;
+  gboolean discont;
 
   ret = GST_FLOW_OK;
 
@@ -514,6 +607,14 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
   if (payload->priv->get_lengths == NULL || payload->priv->get_duration == NULL)
     goto config_error;
 
+  discont = GST_BUFFER_IS_DISCONT (buffer);
+  if (discont) {
+    GST_DEBUG_OBJECT (payload, "Got DISCONT");
+    /* flush everything out of the adapter, mark DISCONT */
+    ret = gst_base_rtp_audio_payload_flush (payload, -1, -1);
+    payload->priv->discont = TRUE;
+  }
+
   if (!payload->priv->get_lengths (basepayload, &min_payload_len,
           &max_payload_len, &align))
     goto config_error;
@@ -547,28 +648,12 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
 
     /* as long as we have full frames */
     while (available >= min_payload_len) {
-      guint64 distance;
-      GstClockTime timestamp;
-
       payload_len = ALIGN_DOWN (available, align);
       payload_len = MIN (max_payload_len, payload_len);
 
-      /* calculate the timestamp */
+      /* and flush out the bytes from the adapter, automatically set the
-      timestamp =
+       * timestamp. */
-          gst_adapter_prev_timestamp (payload->priv->adapter, &distance);
+      ret = gst_base_rtp_audio_payload_flush (payload, payload_len, -1);
-
-      GST_LOG_OBJECT (payload,
-          "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
-          GST_TIME_ARGS (timestamp), distance);
-
-      if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
-        /* convert the number of bytes since the last timestamp to time and add to
-         * the last seen timestamp */
-        timestamp += payload->priv->get_duration (payload, distance);
-      }
-
-      /* and flush out the bytes from the adapter */
-      ret = gst_base_rtp_audio_payload_flush (payload, payload_len, timestamp);
 
       available -= payload_len;
       GST_DEBUG_OBJECT (payload, "available after push %u", available);
@@ -579,7 +664,8 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
   /* ERRORS */
 config_error:
   {
-    GST_DEBUG_OBJECT (payload, "Required options not set");
+    GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
+        ("subclass did not configure us properly"));
     gst_buffer_unref (buffer);
     return GST_FLOW_ERROR;
   }
@@ -617,9 +703,8 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
 
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_EOS:
-      /* FIXME. push remaining bytes? maybe not because it would violate the
+      /* flush remaining bytes in the adapter */
-       * min-ptime. */
+      gst_base_rtp_audio_payload_flush (payload, -1, -1);
-      gst_adapter_clear (payload->priv->adapter);
       break;
     case GST_EVENT_FLUSH_STOP:
       gst_adapter_clear (payload->priv->adapter);