X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpdec.c;h=7e8b52adad50394e46986504492778329c0f23ad;hb=4e8d6218c3cb8b9feffb70f8a53859540b975b36;hp=2c262d9bd3190d2a14586d4d5623ea0f8194a368;hpb=0ebcdf5cdad6bf20a5170735a7f77b23ecc081ac;p=ffmpeg

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 2c262d9bd31..7e8b52adad5 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -20,13 +20,13 @@
  */
 
 #include "libavutil/mathematics.h"
+#include "libavutil/avstring.h"
 #include "libavcodec/get_bits.h"
 #include "avformat.h"
 #include "mpegts.h"
 #include "url.h"
 
 #include <unistd.h>
-#include <strings.h>
 #include "network.h"
 
 #include "rtpdec.h"
@@ -83,6 +83,11 @@ void av_register_rtp_dynamic_payload_handlers(void)
     ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
     ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
     ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
+
+    ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
 }
 
 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
@@ -91,7 +96,7 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
     RTPDynamicProtocolHandler *handler;
     for (handler = RTPFirstDynamicPayloadHandler;
          handler; handler = handler->next)
-        if (!strcasecmp(name, handler->enc_name) &&
+        if (!av_strcasecmp(name, handler->enc_name) &&
             codec_type == handler->codec_type)
             return handler;
     return NULL;
@@ -112,14 +117,15 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
 {
     int payload_len;
-    while (len >= 2) {
+    while (len >= 4) {
+        payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
+
         switch (buf[1]) {
         case RTCP_SR:
-            if (len < 16) {
+            if (payload_len < 20) {
                 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
                 return AVERROR_INVALIDDATA;
             }
-            payload_len = (AV_RB16(buf + 2) + 1) * 4;
 
             s->last_rtcp_ntp_time = AV_RB64(buf + 8);
             s->last_rtcp_timestamp = AV_RB32(buf + 16);
@@ -130,14 +136,13 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
                 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
             }
 
-            buf += payload_len;
-            len -= payload_len;
             break;
         case RTCP_BYE:
             return -RTCP_BYE;
-        default:
-            return -1;
         }
+
+        buf += payload_len;
+        len -= payload_len;
     }
     return -1;
 }
@@ -218,23 +223,7 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
     return 1;
 }
 
-#if 0
-/**
-* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
-* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
-* never change.  I left this in in case someone else can see a way. (rdm)
-*/
-static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
-{
-    uint32_t transit= arrival_timestamp - sent_timestamp;
-    int d;
-    s->transit= transit;
-    d= FFABS(transit - s->transit);
-    s->jitter += d - ((s->jitter + 8)>>4);
-}
-#endif
-
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -331,7 +320,7 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     return 0;
 }
 
-void rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext* rtp_handle)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -375,7 +364,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
  * MPEG2TS streams to indicate that they should be demuxed inside the
  * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
  */
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
 {
     RTPDemuxContext *s;
 
@@ -423,8 +412,8 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
 }
 
 void
-rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
-                               RTPDynamicProtocolHandler *handler)
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+                                  RTPDynamicProtocolHandler *handler)
 {
     s->dynamic_protocol_context = ctx;
     s->parse_packet = handler->parse_packet;
@@ -437,7 +426,10 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
 {
     if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
         return; /* Timestamp already set by depacketizer */
-    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
+    if (timestamp == RTP_NOTS_VALUE)
+        return;
+
+    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
         int64_t addend;
         int delta_timestamp;
 
@@ -449,11 +441,16 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
                    delta_timestamp;
         return;
     }
-    if (timestamp == RTP_NOTS_VALUE)
-        return;
+
     if (!s->base_timestamp)
         s->base_timestamp = timestamp;
-    pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
+    /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
+    if (!s->timestamp)
+        s->unwrapped_timestamp += timestamp;
+    else
+        s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
+    s->timestamp = timestamp;
+    pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
 }
 
 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
@@ -738,8 +735,8 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
  * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
  * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
  */
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     uint8_t **bufptr, int len)
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+                        uint8_t **bufptr, int len)
 {
     int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
     s->prev_ret = rv;
@@ -748,7 +745,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
     return rv ? rv : has_next_packet(s);
 }
 
-void rtp_parse_close(RTPDemuxContext *s)
+void ff_rtp_parse_close(RTPDemuxContext *s)
 {
     ff_rtp_reset_packet_queue(s);
     if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {