RTP sout: rework sample-based audio codecs to not exceed the MTU

Send as much data as the packetization AND the MTU allow

diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c
index 430be664f7e3c562e64bbc535e17d7da7a128a58..2ace05afc686135f699cc403cbbb95a190871949 100644 (file)
--- a/modules/stream_out/rtp.c
+++ b/modules/stream_out/rtp.c
@@ -812,6 +812,22 @@ static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
     s[2*i_data] = '\0';
 }
 
+/**
+ * Shrink the MTU down to a fixed packetization time (for audio).
+ */
+static void
+rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
+{
+    /* Samples per second */
+    size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
+    bytes *= id->i_channels;
+    spl *= bytes;
+
+    if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
+        id->i_mtu = 12 + spl;
+    else /* MTU is too small for ptime, align to a sample boundary */
+        id->i_mtu = (id->i_mtu / bytes) * bytes;
+}
 
 /** Add an ES as a new RTP stream */
 static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
@@ -968,13 +984,15 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
             if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
                 id->i_payload_type = 0;
             id->psz_enc = "PCMU";
-            id->pf_packetize = rtp_packetize_l8;
+            id->pf_packetize = rtp_packetize_split;
+            rtp_set_ptime (id, 20, 1);
             break;
         case VLC_FOURCC( 'a', 'l', 'a', 'w' ):
             if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
                 id->i_payload_type = 8;
             id->psz_enc = "PCMA";
-            id->pf_packetize = rtp_packetize_l8;
+            id->pf_packetize = rtp_packetize_split;
+            rtp_set_ptime (id, 20, 1);
             break;
         case VLC_FOURCC( 's', '1', '6', 'b' ):
             if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
@@ -987,11 +1005,13 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
                 id->i_payload_type = 10;
             }
             id->psz_enc = "L16";
-            id->pf_packetize = rtp_packetize_l16;
+            id->pf_packetize = rtp_packetize_split;
+            rtp_set_ptime (id, 20, 1);
             break;
         case VLC_FOURCC( 'u', '8', ' ', ' ' ):
             id->psz_enc = "L8";
-            id->pf_packetize = rtp_packetize_l8;
+            id->pf_packetize = rtp_packetize_split;
+            rtp_set_ptime (id, 20, 1);
             break;
         case VLC_FOURCC( 'm', 'p', 'g', 'a' ):
         case VLC_FOURCC( 'm', 'p', '3', ' ' ):
@@ -1515,16 +1535,6 @@ size_t rtp_mtu (const sout_stream_id_t *id)
     return id->i_mtu - 12;
 }
 
-/**
- * @return number of audio samples to include for a given packetization time
- * (this really only makes sense for audio formats).
- */
-size_t rtp_plen (const sout_stream_id_t * id, unsigned ptime_ms)
-{
-    return id->i_channels * (((id->i_clock_rate - 1) * ptime_ms / 1000) + 1);
-}
-
-
 /*****************************************************************************
  * Non-RTP mux
  *****************************************************************************/

diff --git a/modules/stream_out/rtp.h b/modules/stream_out/rtp.h
index 3cddd59a149717156c6de15fcbbe7e3c4ad4fd45..e8b1264373b172f1fcf72a107118332cead5a465 100644 (file)
--- a/modules/stream_out/rtp.h
+++ b/modules/stream_out/rtp.h
@@ -46,10 +46,7 @@ void rtp_packetize_common (sout_stream_id_t *id, block_t *out,
                            int b_marker, int64_t i_pts);
 void rtp_packetize_send (sout_stream_id_t *id, block_t *out);
 size_t rtp_mtu (const sout_stream_id_t *id);
-size_t rtp_plen (const sout_stream_id_t * id, unsigned ptime);
 
-int rtp_packetize_l16  (sout_stream_t *, sout_stream_id_t *, block_t *);
-int rtp_packetize_l8   (sout_stream_t *, sout_stream_id_t *, block_t *);
 int rtp_packetize_mpa  (sout_stream_t *, sout_stream_id_t *, block_t *);
 int rtp_packetize_mpv  (sout_stream_t *, sout_stream_id_t *, block_t *);
 int rtp_packetize_ac3  (sout_stream_t *, sout_stream_id_t *, block_t *);

diff --git a/modules/stream_out/rtpfmt.c b/modules/stream_out/rtpfmt.c
index bcd8ec6058f6966a0479da2609f099ef840647d7..53af124f7ee4d3622b5d734ea6d80bf92a71961d 100644 (file)
--- a/modules/stream_out/rtpfmt.c
+++ b/modules/stream_out/rtpfmt.c
@@ -303,66 +303,6 @@ int rtp_packetize_mp4a_latm( sout_stream_t *p_stream, sout_stream_id_t *id,
     return VLC_SUCCESS;
 }
 
-int rtp_packetize_l16( sout_stream_t *p_stream, sout_stream_id_t *id,
-                       block_t *in )
-{
-    const uint8_t *p_data = in->p_buffer;
-    size_t i_data  = in->i_buffer;
-    size_t i_plen = 2 * rtp_plen (id, 20);
-
-    for( unsigned i_packet = 0; i_data > 0; i_packet++ )
-    {
-        int           i_payload = __MIN( i_plen, i_data );
-        block_t *out = block_New( p_stream, 12 + i_payload );
-
-        /* rtp common header */
-        rtp_packetize_common( id, out, 0,
-                              (in->i_pts > 0 ? in->i_pts : in->i_dts) );
-        memcpy( &out->p_buffer[12], p_data, i_payload );
-
-        out->i_buffer = 12 + i_payload;
-        out->i_dts    = in->i_dts + i_packet * 20000;
-        out->i_length = i_payload * 20000 / i_plen;
-
-        rtp_packetize_send( id, out );
-
-        p_data += i_payload;
-        i_data -= i_payload;
-    }
-
-    return VLC_SUCCESS;
-}
-
-int rtp_packetize_l8( sout_stream_t *p_stream, sout_stream_id_t *id,
-                      block_t *in )
-{
-    const uint8_t *p_data = in->p_buffer;
-    size_t i_data  = in->i_buffer;
-    size_t i_plen = rtp_plen (id, 20);
-
-    for( unsigned i_packet = 0; i_data > 0; i_packet++ )
-    {
-        int           i_payload = __MIN( i_plen, i_data );
-        block_t *out = block_New( p_stream, 12 + i_payload );
-
-        /* rtp common header */
-        rtp_packetize_common( id, out, 0,
-                              (in->i_pts > 0 ? in->i_pts : in->i_dts) );
-        memcpy( &out->p_buffer[12], p_data, i_payload );
-
-        out->i_buffer = 12 + i_payload;
-        out->i_dts    = in->i_dts + i_packet * 20000;
-        out->i_length = i_payload * 20000 / i_plen;
-
-        rtp_packetize_send( id, out );
-
-        p_data += i_payload;
-        i_data -= i_payload;
-    }
-
-    return VLC_SUCCESS;
-}
-
 int rtp_packetize_mp4a( sout_stream_t *p_stream, sout_stream_id_t *id,
                         block_t *in )
 {