#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,
+};
+
+static RTPDynamicProtocolHandler speex_dynamic_handler = {
+ .enc_name = "speex",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_SPEEX,
+};
- - 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 opus_dynamic_handler = {
+ .enc_name = "opus",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_OPUS,
+};
-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 t140_dynamic_handler = { /* RFC 4103 */
+ .enc_name = "t140",
+ .codec_type = AVMEDIA_TYPE_DATA,
+ .codec_id = AV_CODEC_ID_TEXT,
};
-/* 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)
+void ff_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_ac3_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_dv_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_h261_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_hevc_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_ms_rtp_asf_pfv_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_audio_robust_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_pfa_handler);
-
+ ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_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);
+ ff_register_dynamic_payload_handler(&t140_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) &&
+ if (handler->enc_name &&
+ !av_strcasecmp(name, handler->enc_name) &&
codec_type == handler->codec_type)
return handler;
return NULL;
}
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)
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) {
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_relative();
+ 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;
if (!s->base_timestamp)
s->base_timestamp = s->last_rtcp_timestamp;
- s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
+ s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
}
break;
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 {
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;
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
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_relative() - 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 */
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(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext *rtp_handle)
{
AVIOContext *pb;
uint8_t *buf;
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 = 0, missing_mask = 0;
+
+ 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_relative();
+ 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_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)
}
}
// 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)
{
/* 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;
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;
- int ext;
+ unsigned int ssrc;
+ int payload_type, seq, flags = 0;
+ int ext, csrc;
AVStream *st;
uint32_t timestamp;
- int rv= 0;
+ int rv = 0;
- ext = buf[0] & 0x10;
+ csrc = buf[0] & 0x0f;
+ 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;
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;
}
}
s->seq = seq;
- len -= 12;
- buf += 12;
+ len -= 12;
+ buf += 12;
+
+ len -= 4 * csrc;
+ buf += 4 * csrc;
+ if (len < 0)
+ return AVERROR_INVALIDDATA;
/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
if (ext) {
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, ×tamp, 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, ×tamp, buf, len, seq,
+ flags);
+ } else if (st) {
+ 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....
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->recvtime = av_gettime_relative();
+ packet->seq = seq;
+ packet->len = len;
+ packet->buf = buf;
+ packet->next = *cur;
+ *cur = packet;
s->queue_len++;
}
"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);
}
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
- uint8_t **bufptr, int len)
+ uint8_t **bufptr, int len)
{
- uint8_t* buf = 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,
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, ×tamp, NULL, 0, flags);
+ rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
+ s->st, pkt, ×tamp, 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;
}
}
return rtcp_parse_packet(s, buf, len);
}
+ if (s->st) {
+ int64_t received = av_gettime_relative();
+ 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);
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);
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);
}
-int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
- int (*parse_fmtp)(AVStream *stream,
+int ff_parse_fmtp(AVFormatContext *s,
+ AVStream *stream, PayloadContext *data, const char *p,
+ int (*parse_fmtp)(AVFormatContext *s,
+ AVStream *stream,
PayloadContext *data,
- char *attr, char *value))
+ const char *attr, const char *value))
{
char attr[256];
char *value;
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);
+ res = parse_fmtp(s, stream, data, attr, value);
if (res < 0 && res != AVERROR_PATCHWELCOME) {
av_free(value);
return res;
av_free(value);
return 0;
}
+
+int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
+{
+ int ret;
+ av_init_packet(pkt);
+
+ pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
+ pkt->stream_index = stream_idx;
+ *dyn_buf = NULL;
+ if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
+ av_freep(&pkt->data);
+ return ret;
+ }
+ return pkt->size;
+}
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