gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added...

gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...

Original commit message from CVS:
Patch by: Zeeshan Ali  <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).

ChangeLog

+2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>
+
+	Patch by: Zeeshan Ali  <zeenix gmail com>
+
+	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+	(gst_base_rtp_audio_payload_handle_frame_based_buffer),
+	(gst_base_rtp_audio_payload_handle_sample_based_buffer),
+	(gst_base_rtp_audio_payload_push):
+	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+	  The recently-added gst_base_rtp_audio_payload_push() should take an
+	  object of type GstBaseRTPAudioPayload as first argument (#431672).
+
 2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>
 	* gst/audioresample/gstaudioresample.c:

gst-libs/gst/rtp/gstbasertpaudiopayload.c

@@ -414,7 +414,7 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
     /* this will check against max_ptime and max_mtu */
     if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
         GST_BUFFER_SIZE (buffer) <= max_payload_len) {
-      ret = gst_base_rtp_audio_payload_push (basepayload,
+      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
           GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
           GST_BUFFER_TIMESTAMP (buffer));
       gst_buffer_unref (buffer);
@@ -437,7 +437,8 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
       data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
     }
 
-    ret = gst_base_rtp_audio_payload_push (basepayload, data, payload_len,
+    ret =
+        gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
         basertpaudiopayload->base_ts);
 
     ts_inc = (payload_len * frame_duration) / frame_size;
@@ -540,7 +541,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
     /* this will check against max_ptime and max_mtu */
     if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
         GST_BUFFER_SIZE (buffer) <= max_payload_len) {
-      ret = gst_base_rtp_audio_payload_push (basepayload,
+      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
           GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
           GST_BUFFER_TIMESTAMP (buffer));
       gst_buffer_unref (buffer);
@@ -562,7 +563,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
       data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
     }
 
-    ret = gst_base_rtp_audio_payload_push (basepayload, data, payload_len,
+    ret =
+        gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
         basertpaudiopayload->base_ts);
 
     num = payload_len;
@@ -612,14 +614,17 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
  * Returns: a #GstFlowReturn
  */
 GstFlowReturn
-gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload,
+gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
     const guint8 * data, guint payload_len, GstClockTime timestamp)
 {
+  GstBaseRTPPayload *basepayload;
   GstBuffer *outbuf;
   guint8 *payload;
   GstFlowReturn ret;
 
-  GST_DEBUG_OBJECT (basepayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
+  basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+
+  GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
       payload_len, GST_TIME_ARGS (timestamp));
 
   /* create buffer to hold the payload */

gst-libs/gst/rtp/gstbasertpaudiopayload.h

@@ -85,7 +85,7 @@ gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
     *basertpaudiopayload, gint sample_size);
 
 GstFlowReturn
-gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload, 
+gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, 
     const guint8 * data, guint payload_len, GstClockTime timestamp);
 
 GstAdapter*