rtpdec: Check the return value from av_new_packet
[ffmpeg.git] / libavformat / rtpdec.c
index 61653f7..a818887 100644 (file)
 
 #include "libavutil/mathematics.h"
 #include "libavutil/avstring.h"
+#include "libavutil/time.h"
 #include "libavcodec/get_bits.h"
 #include "avformat.h"
-#include "mpegts.h"
-#include "url.h"
-
-#include <unistd.h>
 #include "network.h"
-
+#include "srtp.h"
+#include "url.h"
 #include "rtpdec.h"
 #include "rtpdec_formats.h"
 
-//#define DEBUG
+#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
 
-/* TODO: - add RTCP statistics reporting (should be optional).
+static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
+    .enc_name   = "X-MP3-draft-00",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = AV_CODEC_ID_MP3ADU,
+};
 
-         - add support for h263/mpeg4 packetized output : IDEA: send a
-         buffer to 'rtp_write_packet' contains all the packets for ONE
-         frame. Each packet should have a four byte header containing
-         the length in big endian format (same trick as
-         'ffio_open_dyn_packet_buf')
-*/
+static RTPDynamicProtocolHandler speex_dynamic_handler = {
+    .enc_name   = "speex",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = AV_CODEC_ID_SPEEX,
+};
 
-static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
-    .enc_name           = "X-MP3-draft-00",
-    .codec_type         = AVMEDIA_TYPE_AUDIO,
-    .codec_id           = CODEC_ID_MP3ADU,
+static RTPDynamicProtocolHandler opus_dynamic_handler = {
+    .enc_name   = "opus",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = AV_CODEC_ID_OPUS,
 };
 
-/* statistics functions */
-static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
+static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
 
 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
 {
-    handler->next= RTPFirstDynamicPayloadHandler;
-    RTPFirstDynamicPayloadHandler= handler;
+    handler->next = rtp_first_dynamic_payload_handler;
+    rtp_first_dynamic_payload_handler = handler;
 }
 
 void av_register_rtp_dynamic_payload_handlers(void)
 {
-    ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
-    ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_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_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);
     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_h263_rfc2190_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_qdm2_dynamic_handler);
-    ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
-    ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
-    ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
-    ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
-
+    ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
     ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
-
+    ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
     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);
+    ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
+    ff_register_dynamic_payload_handler(&opus_dynamic_handler);
+    ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
+    ff_register_dynamic_payload_handler(&speex_dynamic_handler);
 }
 
 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
-                                                  enum AVMediaType codec_type)
+                                                       enum AVMediaType codec_type)
 {
     RTPDynamicProtocolHandler *handler;
-    for (handler = RTPFirstDynamicPayloadHandler;
+    for (handler = rtp_first_dynamic_payload_handler;
          handler; handler = handler->next)
         if (!av_strcasecmp(name, handler->enc_name) &&
             codec_type == handler->codec_type)
@@ -104,10 +108,10 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
 }
 
 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
-                                                enum AVMediaType codec_type)
+                                                     enum AVMediaType codec_type)
 {
     RTPDynamicProtocolHandler *handler;
-    for (handler = RTPFirstDynamicPayloadHandler;
+    for (handler = rtp_first_dynamic_payload_handler;
          handler; handler = handler->next)
         if (handler->static_payload_id && handler->static_payload_id == id &&
             codec_type == handler->codec_type)
@@ -115,7 +119,8 @@ RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
     return NULL;
 }
 
-static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
+static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
+                             int len)
 {
     int payload_len;
     while (len >= 4) {
@@ -124,11 +129,13 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
         switch (buf[1]) {
         case RTCP_SR:
             if (payload_len < 20) {
-                av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
+                av_log(NULL, AV_LOG_ERROR,
+                       "Invalid length for RTCP SR packet\n");
                 return AVERROR_INVALIDDATA;
             }
 
-            s->last_rtcp_ntp_time = AV_RB64(buf + 8);
+            s->last_rtcp_reception_time = av_gettime();
+            s->last_rtcp_ntp_time  = AV_RB64(buf + 8);
             s->last_rtcp_timestamp = AV_RB32(buf + 16);
             if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
                 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
@@ -148,73 +155,70 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
     return -1;
 }
 
-#define RTP_SEQ_MOD (1<<16)
+#define RTP_SEQ_MOD (1 << 16)
 
-/**
-* called on parse open packet
-*/
-static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
+static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
 {
     memset(s, 0, sizeof(RTPStatistics));
-    s->max_seq= base_sequence;
-    s->probation= 1;
+    s->max_seq   = base_sequence;
+    s->probation = 1;
 }
 
-/**
-* called whenever there is a large jump in sequence numbers, or when they get out of probation...
-*/
+/*
+ * Called whenever there is a large jump in sequence numbers,
+ * or when they get out of probation...
+ */
 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
 {
-    s->max_seq= seq;
-    s->cycles= 0;
-    s->base_seq= seq -1;
-    s->bad_seq= RTP_SEQ_MOD + 1;
-    s->received= 0;
-    s->expected_prior= 0;
-    s->received_prior= 0;
-    s->jitter= 0;
-    s->transit= 0;
+    s->max_seq        = seq;
+    s->cycles         = 0;
+    s->base_seq       = seq - 1;
+    s->bad_seq        = RTP_SEQ_MOD + 1;
+    s->received       = 0;
+    s->expected_prior = 0;
+    s->received_prior = 0;
+    s->jitter         = 0;
+    s->transit        = 0;
 }
 
-/**
-* returns 1 if we should handle this packet.
-*/
+/* Returns 1 if we should handle this packet. */
 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
 {
-    uint16_t udelta= seq - s->max_seq;
-    const int MAX_DROPOUT= 3000;
-    const int MAX_MISORDER = 100;
+    uint16_t udelta = seq - s->max_seq;
+    const int MAX_DROPOUT    = 3000;
+    const int MAX_MISORDER   = 100;
     const int MIN_SEQUENTIAL = 2;
 
-    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
-    if(s->probation)
-    {
-        if(seq==s->max_seq + 1) {
+    /* source not valid until MIN_SEQUENTIAL packets with sequence
+     * seq. numbers have been received */
+    if (s->probation) {
+        if (seq == s->max_seq + 1) {
             s->probation--;
-            s->max_seq= seq;
-            if(s->probation==0) {
+            s->max_seq = seq;
+            if (s->probation == 0) {
                 rtp_init_sequence(s, seq);
                 s->received++;
                 return 1;
             }
         } else {
-            s->probation= MIN_SEQUENTIAL - 1;
-            s->max_seq = seq;
+            s->probation = MIN_SEQUENTIAL - 1;
+            s->max_seq   = seq;
         }
     } else if (udelta < MAX_DROPOUT) {
         // in order, with permissible gap
-        if(seq < s->max_seq) {
-            //sequence number wrapped; count antother 64k cycles
+        if (seq < s->max_seq) {
+            // sequence number wrapped; count another 64k cycles
             s->cycles += RTP_SEQ_MOD;
         }
-        s->max_seq= seq;
+        s->max_seq = seq;
     } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
         // sequence made a large jump...
-        if(seq==s->bad_seq) {
-            // two sequential packets-- assume that the other side restarted without telling us; just resync.
+        if (seq == s->bad_seq) {
+            /* two sequential packets -- assume that the other side
+             * restarted without telling us; just resync. */
             rtp_init_sequence(s, seq);
         } else {
-            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
+            s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
             return 0;
         }
     } else {
@@ -224,27 +228,45 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
     return 1;
 }
 
-int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
+static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
+                               uint32_t arrival_timestamp)
+{
+    // Most of this is pretty straight from RFC 3550 appendix A.8
+    uint32_t transit = arrival_timestamp - sent_timestamp;
+    uint32_t prev_transit = s->transit;
+    int32_t d = transit - prev_transit;
+    // Doing the FFABS() call directly on the "transit - prev_transit"
+    // expression doesn't work, since it's an unsigned expression. Doing the
+    // transit calculation in unsigned is desired though, since it most
+    // probably will need to wrap around.
+    d = FFABS(d);
+    s->transit = transit;
+    if (!prev_transit)
+        return;
+    s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
+}
+
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
+                                  AVIOContext *avio, int count)
 {
     AVIOContext *pb;
     uint8_t *buf;
     int len;
     int rtcp_bytes;
-    RTPStatistics *stats= &s->statistics;
+    RTPStatistics *stats = &s->statistics;
     uint32_t lost;
     uint32_t extended_max;
     uint32_t expected_interval;
     uint32_t received_interval;
-    uint32_t lost_interval;
+    int32_t  lost_interval;
     uint32_t expected;
     uint32_t fraction;
-    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
 
-    if (!s->rtp_ctx || (count < 1))
+    if ((!fd && !avio) || (count < 1))
         return -1;
 
     /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
-    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
+    /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
     s->octet_count += count;
     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
         RTCP_TX_RATIO_DEN;
@@ -253,7 +275,9 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
         return -1;
     s->last_octet_count = s->octet_count;
 
-    if (avio_open_dyn_buf(&pb) < 0)
+    if (!fd)
+        pb = avio;
+    else if (avio_open_dyn_buf(&pb) < 0)
         return -1;
 
     // Receiver Report
@@ -265,31 +289,33 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     avio_wb32(pb, s->ssrc); // server SSRC
     // some placeholders we should really fill...
     // RFC 1889/p64
-    extended_max= stats->cycles + stats->max_seq;
-    expected= extended_max - stats->base_seq + 1;
-    lost= expected - stats->received;
-    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
-    expected_interval= expected - stats->expected_prior;
-    stats->expected_prior= expected;
-    received_interval= stats->received - stats->received_prior;
-    stats->received_prior= stats->received;
-    lost_interval= expected_interval - received_interval;
-    if (expected_interval==0 || lost_interval<=0) fraction= 0;
-    else fraction = (lost_interval<<8)/expected_interval;
-
-    fraction= (fraction<<24) | lost;
+    extended_max          = stats->cycles + stats->max_seq;
+    expected              = extended_max - stats->base_seq;
+    lost                  = expected - stats->received;
+    lost                  = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
+    expected_interval     = expected - stats->expected_prior;
+    stats->expected_prior = expected;
+    received_interval     = stats->received - stats->received_prior;
+    stats->received_prior = stats->received;
+    lost_interval         = expected_interval - received_interval;
+    if (expected_interval == 0 || lost_interval <= 0)
+        fraction = 0;
+    else
+        fraction = (lost_interval << 8) / expected_interval;
+
+    fraction = (fraction << 24) | lost;
 
     avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
     avio_wb32(pb, extended_max); /* max sequence received */
-    avio_wb32(pb, stats->jitter>>4); /* jitter */
+    avio_wb32(pb, stats->jitter >> 4); /* jitter */
 
-    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
-    {
+    if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
         avio_wb32(pb, 0); /* last SR timestamp */
         avio_wb32(pb, 0); /* delay since last SR */
     } else {
-        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
-        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
+        uint32_t middle_32_bits   = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
+        uint32_t delay_since_last = av_rescale(av_gettime() - s->last_rtcp_reception_time,
+                                               65536, AV_TIME_BASE);
 
         avio_wb32(pb, middle_32_bits); /* last SR timestamp */
         avio_wb32(pb, delay_since_last); /* delay since last SR */
@@ -299,29 +325,31 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
     avio_w8(pb, RTCP_SDES);
     len = strlen(s->hostname);
-    avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
+    avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
     avio_wb32(pb, s->ssrc + 1);
     avio_w8(pb, 0x01);
     avio_w8(pb, len);
     avio_write(pb, s->hostname, len);
+    avio_w8(pb, 0); /* END */
     // padding
-    for (len = (6 + len) % 4; len % 4; len++) {
+    for (len = (7 + len) % 4; len % 4; len++)
         avio_w8(pb, 0);
-    }
 
     avio_flush(pb);
+    if (!fd)
+        return 0;
     len = avio_close_dyn_buf(pb, &buf);
     if ((len > 0) && buf) {
         int av_unused result;
         av_dlog(s->ic, "sending %d bytes of RR\n", len);
-        result= ffurl_write(s->rtp_ctx, buf, len);
+        result = ffurl_write(fd, buf, len);
         av_dlog(s->ic, "result from ffurl_write: %d\n", result);
         av_free(buf);
     }
     return 0;
 }
 
-void ff_rtp_send_punch_packets(URLContextrtp_handle)
+void ff_rtp_send_punch_packets(URLContext *rtp_handle)
 {
     AVIOContext *pb;
     uint8_t *buf;
@@ -359,47 +387,122 @@ void ff_rtp_send_punch_packets(URLContext* rtp_handle)
     av_free(buf);
 }
 
+static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
+                                uint16_t *missing_mask)
+{
+    int i;
+    uint16_t next_seq = s->seq + 1;
+    RTPPacket *pkt = s->queue;
+
+    if (!pkt || pkt->seq == next_seq)
+        return 0;
+
+    *missing_mask = 0;
+    for (i = 1; i <= 16; i++) {
+        uint16_t missing_seq = next_seq + i;
+        while (pkt) {
+            int16_t diff = pkt->seq - missing_seq;
+            if (diff >= 0)
+                break;
+            pkt = pkt->next;
+        }
+        if (!pkt)
+            break;
+        if (pkt->seq == missing_seq)
+            continue;
+        *missing_mask |= 1 << (i - 1);
+    }
+
+    *first_missing = next_seq;
+    return 1;
+}
+
+int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
+                              AVIOContext *avio)
+{
+    int len, need_keyframe, missing_packets;
+    AVIOContext *pb;
+    uint8_t *buf;
+    int64_t now;
+    uint16_t first_missing, missing_mask;
+
+    if (!fd && !avio)
+        return -1;
+
+    need_keyframe = s->handler && s->handler->need_keyframe &&
+                    s->handler->need_keyframe(s->dynamic_protocol_context);
+    missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
+
+    if (!need_keyframe && !missing_packets)
+        return 0;
+
+    /* Send new feedback if enough time has elapsed since the last
+     * feedback packet. */
+
+    now = av_gettime();
+    if (s->last_feedback_time &&
+        (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
+        return 0;
+    s->last_feedback_time = now;
+
+    if (!fd)
+        pb = avio;
+    else if (avio_open_dyn_buf(&pb) < 0)
+        return -1;
+
+    if (need_keyframe) {
+        avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
+        avio_w8(pb, RTCP_PSFB);
+        avio_wb16(pb, 2); /* length in words - 1 */
+        // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
+        avio_wb32(pb, s->ssrc + 1);
+        avio_wb32(pb, s->ssrc); // server SSRC
+    }
+
+    if (missing_packets) {
+        avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
+        avio_w8(pb, RTCP_RTPFB);
+        avio_wb16(pb, 3); /* length in words - 1 */
+        avio_wb32(pb, s->ssrc + 1);
+        avio_wb32(pb, s->ssrc); // server SSRC
+
+        avio_wb16(pb, first_missing);
+        avio_wb16(pb, missing_mask);
+    }
+
+    avio_flush(pb);
+    if (!fd)
+        return 0;
+    len = avio_close_dyn_buf(pb, &buf);
+    if (len > 0 && buf) {
+        ffurl_write(fd, buf, len);
+        av_free(buf);
+    }
+    return 0;
+}
 
 /**
  * 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
- * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
+ * MPEG2-TS streams.
  */
-RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
+                                   int payload_type, int queue_size)
 {
     RTPDemuxContext *s;
 
     s = av_mallocz(sizeof(RTPDemuxContext));
     if (!s)
         return NULL;
-    s->payload_type = payload_type;
-    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
+    s->payload_type        = payload_type;
+    s->last_rtcp_ntp_time  = AV_NOPTS_VALUE;
     s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
-    s->ic = s1;
-    s->st = st;
-    s->queue_size = queue_size;
-    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
-    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
-        s->ts = ff_mpegts_parse_open(s->ic);
-        if (s->ts == NULL) {
-            av_free(s);
-            return NULL;
-        }
-    } else {
-        switch(st->codec->codec_id) {
-        case CODEC_ID_MPEG1VIDEO:
-        case CODEC_ID_MPEG2VIDEO:
-        case CODEC_ID_MP2:
-        case CODEC_ID_MP3:
-        case CODEC_ID_MPEG4:
-        case CODEC_ID_H263:
-        case CODEC_ID_H264:
-            st->need_parsing = AVSTREAM_PARSE_FULL;
-            break;
-        case CODEC_ID_VORBIS:
-            st->need_parsing = AVSTREAM_PARSE_HEADERS;
-            break;
-        case CODEC_ID_ADPCM_G722:
+    s->ic                  = s1;
+    s->st                  = st;
+    s->queue_size          = queue_size;
+    rtp_init_statistics(&s->statistics, 0);
+    if (st) {
+        switch (st->codec->codec_id) {
+        case AV_CODEC_ID_ADPCM_G722:
             /* According to RFC 3551, the stream clock rate is 8000
              * even if the sample rate is 16000. */
             if (st->codec->sample_rate == 8000)
@@ -410,21 +513,27 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext
         }
     }
     // needed to send back RTCP RR in RTSP sessions
-    s->rtp_ctx = rtpc;
     gethostname(s->hostname, sizeof(s->hostname));
     return s;
 }
 
-void
-ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
-                                  RTPDynamicProtocolHandler *handler)
+void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+                                       RTPDynamicProtocolHandler *handler)
 {
     s->dynamic_protocol_context = ctx;
-    s->parse_packet = handler->parse_packet;
+    s->handler                  = handler;
+}
+
+void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
+                             const char *params)
+{
+    if (!ff_srtp_set_crypto(&s->srtp, suite, params))
+        s->srtp_enabled = 1;
 }
 
 /**
- * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
+ * This was the second switch in rtp_parse packet.
+ * Normalizes time, if required, sets stream_index, etc.
  */
 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
 {
@@ -440,7 +549,9 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
         /* compute pts from timestamp with received ntp_time */
         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);
+        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 = s->range_start_offset + s->rtcp_ts_offset + addend +
                    delta_timestamp;
         return;
@@ -448,32 +559,34 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
 
     if (!s->base_timestamp)
         s->base_timestamp = timestamp;
-    /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
+    /* 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;
+    pkt->pts     = s->unwrapped_timestamp + s->range_start_offset -
+                   s->base_timestamp;
 }
 
 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
                                      const uint8_t *buf, int len)
 {
-    unsigned int ssrc, h;
-    int payload_type, seq, ret, flags = 0;
+    unsigned int ssrc;
+    int payload_type, seq, flags = 0;
     int ext;
     AVStream *st;
     uint32_t timestamp;
-    int rv= 0;
+    int rv = 0;
 
-    ext = buf[0] & 0x10;
+    ext          = buf[0] & 0x10;
     payload_type = buf[1] & 0x7f;
     if (buf[1] & 0x80)
         flags |= RTP_FLAG_MARKER;
-    seq  = AV_RB16(buf + 2);
+    seq       = AV_RB16(buf + 2);
     timestamp = AV_RB32(buf + 4);
-    ssrc = AV_RB32(buf + 8);
+    ssrc      = AV_RB32(buf + 8);
     /* store the ssrc in the RTPDemuxContext */
     s->ssrc = ssrc;
 
@@ -483,9 +596,9 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
 
     st = s->st;
     // only do something with this if all the rtp checks pass...
-    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
-    {
-        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
+    if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
+        av_log(st ? st->codec : NULL, AV_LOG_ERROR,
+               "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
                payload_type, seq, ((s->seq + 1) & 0xffff));
         return -1;
     }
@@ -497,8 +610,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
     }
 
     s->seq = seq;
-    len -= 12;
-    buf += 12;
+    len   -= 12;
+    buf   += 12;
 
     /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
     if (ext) {
@@ -515,63 +628,19 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
         buf += ext;
     }
 
-    if (!st) {
-        /* specific MPEG2TS demux support */
-        ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
-        /* The only error that can be returned from ff_mpegts_parse_packet
-         * is "no more data to return from the provided buffer", so return
-         * AVERROR(EAGAIN) for all errors */
-        if (ret < 0)
-            return AVERROR(EAGAIN);
-        if (ret < len) {
-            s->read_buf_size = len - ret;
-            memcpy(s->buf, buf + ret, s->read_buf_size);
-            s->read_buf_index = 0;
-            return 1;
-        }
-        return 0;
-    } else if (s->parse_packet) {
-        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
-                             s->st, pkt, &timestamp, buf, len, flags);
-    } else {
-        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
-        switch(st->codec->codec_id) {
-        case CODEC_ID_MP2:
-        case CODEC_ID_MP3:
-            /* better than nothing: skip mpeg audio RTP header */
-            if (len <= 4)
-                return -1;
-            h = AV_RB32(buf);
-            len -= 4;
-            buf += 4;
-            av_new_packet(pkt, len);
-            memcpy(pkt->data, buf, len);
-            break;
-        case CODEC_ID_MPEG1VIDEO:
-        case CODEC_ID_MPEG2VIDEO:
-            /* better than nothing: skip mpeg video RTP header */
-            if (len <= 4)
-                return -1;
-            h = AV_RB32(buf);
-            buf += 4;
-            len -= 4;
-            if (h & (1 << 26)) {
-                /* mpeg2 */
-                if (len <= 4)
-                    return -1;
-                buf += 4;
-                len -= 4;
-            }
-            av_new_packet(pkt, len);
-            memcpy(pkt->data, buf, len);
-            break;
-        default:
-            av_new_packet(pkt, len);
-            memcpy(pkt->data, buf, len);
-            break;
-        }
-
+    if (s->handler && s->handler->parse_packet) {
+        rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
+                                      s->st, pkt, &timestamp, buf, len, seq,
+                                      flags);
+    } else if (st) {
+        /* At this point, the RTP header has been stripped;
+         * This is ASSUMING that there is only 1 CSRC, which isn't wise. */
+        if ((rv = av_new_packet(pkt, len)) < 0)
+            return rv;
+        memcpy(pkt->data, buf, len);
         pkt->stream_index = st->index;
+    } else {
+        return AVERROR(EINVAL);
     }
 
     // now perform timestamp things....
@@ -595,30 +664,26 @@ void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
 
 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
 {
-    uint16_t seq = AV_RB16(buf + 2);
-    RTPPacket *cur = s->queue, *prev = NULL, *packet;
+    uint16_t seq   = AV_RB16(buf + 2);
+    RTPPacket **cur = &s->queue, *packet;
 
     /* Find the correct place in the queue to insert the packet */
-    while (cur) {
-        int16_t diff = seq - cur->seq;
+    while (*cur) {
+        int16_t diff = seq - (*cur)->seq;
         if (diff < 0)
             break;
-        prev = cur;
-        cur = cur->next;
+        cur = &(*cur)->next;
     }
 
     packet = av_mallocz(sizeof(*packet));
     if (!packet)
         return;
     packet->recvtime = av_gettime();
-    packet->seq = seq;
-    packet->len = len;
-    packet->buf = buf;
-    packet->next = cur;
-    if (prev)
-        prev->next = packet;
-    else
-        s->queue = packet;
+    packet->seq      = seq;
+    packet->len      = len;
+    packet->buf      = buf;
+    packet->next     = *cur;
+    *cur = packet;
     s->queue_len++;
 }
 
@@ -645,7 +710,7 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
                "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
 
     /* Parse the first packet in the queue, and dequeue it */
-    rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
+    rv   = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
     next = s->queue->next;
     av_free(s->queue->buf);
     av_free(s->queue);
@@ -655,12 +720,12 @@ static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
 }
 
 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     uint8_t **bufptr, int len)
+                                uint8_t **bufptr, int len)
 {
-    uint8_tbuf = bufptr ? *bufptr : NULL;
-    int ret, flags = 0;
+    uint8_t *buf = bufptr ? *bufptr : NULL;
+    int flags = 0;
     uint32_t timestamp;
-    int rv= 0;
+    int rv = 0;
 
     if (!buf) {
         /* If parsing of the previous packet actually returned 0 or an error,
@@ -669,27 +734,15 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
         if (s->prev_ret <= 0)
             return rtp_parse_queued_packet(s, pkt);
         /* return the next packets, if any */
-        if(s->st && s->parse_packet) {
+        if (s->handler && s->handler->parse_packet) {
             /* timestamp should be overwritten by parse_packet, if not,
              * the packet is left with pts == AV_NOPTS_VALUE */
             timestamp = RTP_NOTS_VALUE;
-            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
-                                s->st, pkt, &timestamp, NULL, 0, flags);
+            rv        = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
+                                                 s->st, pkt, &timestamp, NULL, 0, 0,
+                                                 flags);
             finalize_packet(s, pkt, timestamp);
             return rv;
-        } else {
-            // TODO: Move to a dynamic packet handler (like above)
-            if (s->read_buf_index >= s->read_buf_size)
-                return AVERROR(EAGAIN);
-            ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
-                                      s->read_buf_size - s->read_buf_index);
-            if (ret < 0)
-                return AVERROR(EAGAIN);
-            s->read_buf_index += ret;
-            if (s->read_buf_index < s->read_buf_size)
-                return 1;
-            else
-                return 0;
         }
     }
 
@@ -702,6 +755,16 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
         return rtcp_parse_packet(s, buf, len);
     }
 
+    if (s->st) {
+        int64_t received = av_gettime();
+        uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
+                                           s->st->time_base);
+        timestamp = AV_RB32(buf + 4);
+        // Calculate the jitter immediately, before queueing the packet
+        // into the reordering queue.
+        rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
+    }
+
     if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
         /* First packet, or no reordering */
         return rtp_parse_packet_internal(s, pkt, buf, len);
@@ -742,7 +805,10 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
                         uint8_t **bufptr, int len)
 {
-    int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
+    int rv;
+    if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
+        return -1;
+    rv = rtp_parse_one_packet(s, pkt, bufptr, len);
     s->prev_ret = rv;
     while (rv == AVERROR(EAGAIN) && has_next_packet(s))
         rv = rtp_parse_queued_packet(s, pkt);
@@ -752,9 +818,7 @@ int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
 void ff_rtp_parse_close(RTPDemuxContext *s)
 {
     ff_rtp_reset_packet_queue(s);
-    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
-        ff_mpegts_parse_close(s->ts);
-    }
+    ff_srtp_free(&s->srtp);
     av_free(s);
 }
 
@@ -769,19 +833,21 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
     int value_size = strlen(p) + 1;
 
     if (!(value = av_malloc(value_size))) {
-        av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
+        av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
         return AVERROR(ENOMEM);
     }
 
     // remove protocol identifier
-    while (*p && *p == ' ') p++; // strip spaces
-    while (*p && *p != ' ') p++; // eat protocol identifier
-    while (*p && *p == ' ') p++; // strip trailing spaces
+    while (*p && *p == ' ')
+        p++;                     // strip spaces
+    while (*p && *p != ' ')
+        p++;                     // eat protocol identifier
+    while (*p && *p == ' ')
+        p++;                     // strip trailing spaces
 
     while (ff_rtsp_next_attr_and_value(&p,
                                        attr, sizeof(attr),
                                        value, value_size)) {
-
         res = parse_fmtp(stream, data, attr, value);
         if (res < 0 && res != AVERROR_PATCHWELCOME) {
             av_free(value);
@@ -791,3 +857,14 @@ int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
     av_free(value);
     return 0;
 }
+
+int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
+{
+    av_init_packet(pkt);
+
+    pkt->size         = avio_close_dyn_buf(*dyn_buf, &pkt->data);
+    pkt->stream_index = stream_idx;
+    pkt->destruct     = av_destruct_packet;
+    *dyn_buf          = NULL;
+    return pkt->size;
+}