Ticket #1291: rtp.c.patch

vlc-0.9.0-svn-spx/modules/stream_out/rtp.c

@@ -904 +904 @@
-static int rtp_packetize_h263 ( sout_stream_t *, sout_stream_id_t *, block_t * );
-static int rtp_packetize_h264 ( sout_stream_t *, sout_stream_id_t *, block_t * );
-static int rtp_packetize_amr  ( sout_stream_t *, sout_stream_id_t *, block_t * );
+static int rtp_packetize_spx  ( sout_stream_t *, sout_stream_id_t *, block_t * );
-
-static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
-{
@@ -1208 +1209 @@
-            id->i_clock_rate = p_fmt->audio.i_rate;
-            id->pf_packetize = rtp_packetize_amr;
-            break; 
-
+
+
+
+
+
+
-        default:
-            msg_Err( p_stream, "cannot add this stream (unsupported "
-                     "codec:%4.4s)", (char*)&p_fmt->i_codec );
@@ -2545 +2551 @@
-
-    return VLC_SUCCESS;
-}
+
+static int rtp_packetize_spx( sout_stream_t *p_stream, sout_stream_id_t *id,
+                              block_t *in )
+{
+    uint8_t *p_buffer = in->p_buffer;
+    int i_data_size, i_payload_size, i_payload_padding;
+    i_data_size = i_payload_size = in->i_buffer;
+    i_payload_padding = 0;
+    block_t *p_out;
+
+    if ( in->i_buffer + 12 > id->i_mtu )
+    {
+        msg_Warn( p_stream, "Cannot send packet larger than output MTU" );
+        return VLC_SUCCESS;
+    }
+
+    /*
+      RFC for Speex in RTP says that each packet must end on an octet 
+      boundary. So, we check to see if the number of bytes % 4 is zero.
+      If not, we have to add some padding. 
+
+      This MAY be overkill since packetization is handled elsewhere and 
+      appears to ensure the octet boundary. However, better safe than
+      sorry.
+    */
+    if ( i_payload_size % 4 )
+    {
+        i_payload_padding = 4 - ( i_payload_size % 4 );
+        i_payload_size += i_payload_padding;
+    }
+
+    /* 
+      Allocate a new RTP p_output block of the appropriate size. 
+      Allow for 12 extra bytes of RTP header. 
+    */
+    p_out = block_New( p_stream, 12 + i_payload_size );
+
+    if ( i_payload_padding )
+    {
+        /* 
+          The padding is required to be a zero followed by all 1s.
+        */
+        char c_first_pad, c_remaining_pad;
+        c_first_pad = 0x7F;
+        c_remaining_pad = 0xFF;
+
+        /* 
+          Allow for 12 bytes before the i_data_size because
+          of the expected RTP header added during
+          rtp_packetize_common.
+        */
+        p_out->p_buffer[12 + i_data_size] = c_first_pad; 
+        switch (i_payload_padding)
+        {
+            case 2:
+                p_out->p_buffer[12 + i_data_size + 1] = c_remaining_pad; 
+                break;
+            case 3:
+                p_out->p_buffer[12 + i_data_size + 1] = c_remaining_pad; 
+                p_out->p_buffer[12 + i_data_size + 2] = c_remaining_pad; 
+                break;
+        }
+    } 
+
+    /* Add the RTP header to our p_output buffer. */
+    rtp_packetize_common( id, p_out, 0, (in->i_pts > 0 ? in->i_pts : in->i_dts) );
+    /* Copy the Speex payload to the p_output buffer */
+    memcpy( &p_out->p_buffer[12], p_buffer, i_data_size );
+
+    p_out->i_buffer = 12 + i_payload_size;
+    p_out->i_dts = in->i_dts;
+    p_out->i_length = in->i_length;
+
+    /* Queue the buffer for actual transmission. */
+    rtp_packetize_send( id, p_out );
+    return VLC_SUCCESS;
+}