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[ffmpeg] / libavformat / rtpdec.c

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 1498b060dc0d5652d3e9bc5b1954a4746eae6e13..0d2df59a75701ff0fcd38242855357ffa206c27e 100644 (file)
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -30,10 +30,11 @@
 #include "network.h"
 
 #include "rtpdec.h"
-#include "rtp_asf.h"
-#include "rtp_h264.h"
-#include "rtp_vorbis.h"
+#include "rtpdec_amr.h"
+#include "rtpdec_asf.h"
 #include "rtpdec_h263.h"
+#include "rtpdec_h264.h"
+#include "rtpdec_xiph.h"
 
 //#define DEBUG
 
@@ -49,8 +50,8 @@
 /* statistics functions */
 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
 
-static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
-static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
+static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", AVMEDIA_TYPE_VIDEO, CODEC_ID_MPEG4};
+static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC};
 
 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
 {
@@ -62,10 +63,13 @@ void av_register_rtp_dynamic_payload_handlers(void)
 {
     ff_register_dynamic_payload_handler(&mp4v_es_handler);
     ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
+    ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
 
     ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
     ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
@@ -270,6 +274,45 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     return 0;
 }
 
+void rtp_send_punch_packets(URLContext* rtp_handle)
+{
+    ByteIOContext *pb;
+    uint8_t *buf;
+    int len;
+
+    /* Send a small RTP packet */
+    if (url_open_dyn_buf(&pb) < 0)
+        return;
+
+    put_byte(pb, (RTP_VERSION << 6));
+    put_byte(pb, 0); /* Payload type */
+    put_be16(pb, 0); /* Seq */
+    put_be32(pb, 0); /* Timestamp */
+    put_be32(pb, 0); /* SSRC */
+
+    put_flush_packet(pb);
+    len = url_close_dyn_buf(pb, &buf);
+    if ((len > 0) && buf)
+        url_write(rtp_handle, buf, len);
+    av_free(buf);
+
+    /* Send a minimal RTCP RR */
+    if (url_open_dyn_buf(&pb) < 0)
+        return;
+
+    put_byte(pb, (RTP_VERSION << 6));
+    put_byte(pb, 201); /* receiver report */
+    put_be16(pb, 1); /* length in words - 1 */
+    put_be32(pb, 0); /* our own SSRC */
+
+    put_flush_packet(pb);
+    len = url_close_dyn_buf(pb, &buf);
+    if ((len > 0) && buf)
+        url_write(rtp_handle, buf, len);
+    av_free(buf);
+}
+
+
 /**
  * open a new RTP parse context for stream 'st'. 'st' can be NULL for
  * MPEG2TS streams to indicate that they should be demuxed inside the
@@ -309,7 +352,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
             st->need_parsing = AVSTREAM_PARSE_FULL;
             break;
         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);
             }
             break;
@@ -360,7 +403,11 @@ static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
         return -1;
 
     infos->nb_au_headers = au_headers_length / au_header_size;
-    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
+    if (!infos->au_headers || infos->au_headers_allocated < infos->nb_au_headers) {
+        av_free(infos->au_headers);
+        infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
+        infos->au_headers_allocated = infos->nb_au_headers;
+    }
 
     /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
        In my test, the FAAD decoder does not behave correctly when sending each AU one by one
@@ -390,7 +437,7 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
         delta_timestamp = timestamp - s->last_rtcp_timestamp;
         /* convert to the PTS timebase */
         addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
-        pkt->pts = addend + delta_timestamp;
+        pkt->pts = s->range_start_offset + addend + delta_timestamp;
     }
 }
 
@@ -559,6 +606,8 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
 void rtp_parse_close(RTPDemuxContext *s)
 {
     // TODO: fold this into the protocol specific data fields.
+    av_free(s->rtp_payload_data->mode);
+    av_free(s->rtp_payload_data->au_headers);
     if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
         ff_mpegts_parse_close(s->ts);
     }