Speex RTP payload format output - refs #1291

modules/stream_out/rtp.c

@@ -782,6 +782,7 @@ static int rtp_packetize_mp4a_latm ( sout_stream_t *, sout_stream_id_t *, block_
 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 int rtp_packetize_t140 ( sout_stream_t *, sout_stream_id_t *, block_t * );
 
 static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
@@ -1114,6 +1115,14 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
             id->i_clock_rate = p_fmt->audio.i_rate;
             id->pf_packetize = rtp_packetize_amr;
             break;
+        case VLC_FOURCC( 's', 'p', 'x', ' ' ):
+            id->i_payload_type = p_sys->i_payload_type++;
+            if( asprintf( &id->psz_rtpmap, "SPEEX/%d",
+                          p_fmt->audio.i_rate ) == -1)
+                id->psz_rtpmap = NULL;
+            id->i_clock_rate = p_fmt->audio.i_rate;
+            id->pf_packetize = rtp_packetize_spx;
+            break;
         case VLC_FOURCC( 't', '1', '4', '0' ):
             id->psz_rtpmap = strdup( "t140/1000" );
             id->i_clock_rate = 1000;
@@ -2257,3 +2266,80 @@ static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
     p_grab->pf_write    = AccessOutGrabberWrite;
     return p_grab;
 }
+
+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;
+}