Fix "initialization from incompatible pointer type" warning in rv34.

[ffmpeg] / libavformat / rtpenc.c

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 640306aaec722e3de43e8cc91b53fc1c9ba1d3f4..3a541dd28384401a0b67d0b4ee031c1f2fd5d16b 100644 (file)
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -21,6 +21,8 @@
 
 #include "avformat.h"
 #include "mpegts.h"
+#include "internal.h"
+#include "libavutil/random_seed.h"
 
 #include <unistd.h>
 
@@ -29,13 +31,6 @@
 //#define DEBUG
 
 #define RTCP_SR_SIZE 28
-#define NTP_OFFSET 2208988800ULL
-#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
-
-static uint64_t ntp_time(void)
-{
-  return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
-}
 
 static int is_supported(enum CodecID id)
 {
@@ -83,15 +78,18 @@ static int rtp_write_header(AVFormatContext *s1)
 
     s->payload_type = ff_rtp_get_payload_type(st->codec);
     if (s->payload_type < 0)
-        s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == CODEC_TYPE_AUDIO);
+        s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
 
-// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
-    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
+    s->base_timestamp = av_get_random_seed();
     s->timestamp = s->base_timestamp;
     s->cur_timestamp = 0;
-    s->ssrc = 0; /* FIXME: was random(), what should this be? */
+    s->ssrc = av_get_random_seed();
     s->first_packet = 1;
-    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+    s->first_rtcp_ntp_time = ff_ntp_time();
+    if (s1->start_time_realtime)
+        /* Round the NTP time to whole milliseconds. */
+        s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
+                                 NTP_OFFSET_US;
 
     max_packet_size = url_fget_max_packet_size(s1->pb);
     if (max_packet_size <= 12)
@@ -104,14 +102,14 @@ static int rtp_write_header(AVFormatContext *s1)
 
     s->max_frames_per_packet = 0;
     if (s1->max_delay) {
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             if (st->codec->frame_size == 0) {
                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
             } else {
                 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
             }
         }
-        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
+        if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
             /* FIXME: We should round down here... */
             s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
         }
@@ -133,6 +131,12 @@ static int rtp_write_header(AVFormatContext *s1)
         s->max_payload_size = n * TS_PACKET_SIZE;
         s->buf_ptr = s->buf;
         break;
+    case CODEC_ID_H264:
+        /* check for H.264 MP4 syntax */
+        if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
+            s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
+        }
+        break;
     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
         if (!s->max_frames_per_packet)
@@ -153,7 +157,7 @@ static int rtp_write_header(AVFormatContext *s1)
     case CODEC_ID_AAC:
         s->num_frames = 0;
     default:
-        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             av_set_pts_info(st, 32, 1, st->codec->sample_rate);
         }
         s->buf_ptr = s->buf;
@@ -171,7 +175,6 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 
     dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
 
-    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
     s->last_rtcp_ntp_time = ntp_time;
     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
                           s1->streams[0]->time_base) + s->base_timestamp;
@@ -347,8 +350,8 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
         RTCP_TX_RATIO_DEN;
     if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
-                           (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
-        rtcp_send_sr(s1, ntp_time());
+                           (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
+        rtcp_send_sr(s1, ff_ntp_time());
         s->last_octet_count = s->octet_count;
         s->first_packet = 0;
     }