37 #define MIN_FEEDBACK_INTERVAL 200000
140 uintptr_t
i = (uintptr_t)*opaque;
144 *opaque = (
void*)(
i + 1);
169 if (
handler->static_payload_id &&
handler->static_payload_id ==
id &&
185 if (payload_len < 20) {
191 s->last_rtcp_ntp_time =
AV_RB64(buf + 8);
192 s->last_rtcp_timestamp =
AV_RB32(buf + 16);
194 s->first_rtcp_ntp_time =
s->last_rtcp_ntp_time;
195 if (!
s->base_timestamp)
196 s->base_timestamp =
s->last_rtcp_timestamp;
197 s->rtcp_ts_offset = (
int32_t)(
s->last_rtcp_timestamp -
s->base_timestamp);
211 #define RTP_SEQ_MOD (1 << 16)
216 s->max_seq = base_sequence;
228 s->base_seq = seq - 1;
231 s->expected_prior = 0;
232 s->received_prior = 0;
240 uint16_t udelta = seq -
s->max_seq;
241 const int MAX_DROPOUT = 3000;
242 const int MAX_MISORDER = 100;
243 const int MIN_SEQUENTIAL = 2;
248 if (seq ==
s->max_seq + 1) {
251 if (
s->probation == 0) {
257 s->probation = MIN_SEQUENTIAL - 1;
260 }
else if (udelta < MAX_DROPOUT) {
262 if (seq < s->max_seq) {
269 if (seq ==
s->bad_seq) {
285 uint32_t arrival_timestamp)
288 uint32_t transit = arrival_timestamp - sent_timestamp;
289 uint32_t prev_transit =
s->transit;
290 int32_t d = transit - prev_transit;
296 s->transit = transit;
299 s->jitter += d - (
int32_t) ((
s->jitter + 8) >> 4);
311 uint32_t extended_max;
312 uint32_t expected_interval;
313 uint32_t received_interval;
318 if ((!fd && !avio) || (count < 1))
323 s->octet_count += count;
329 s->last_octet_count =
s->octet_count;
345 extended_max =
stats->cycles +
stats->max_seq;
346 expected = extended_max -
stats->base_seq;
347 lost = expected -
stats->received;
348 lost =
FFMIN(lost, 0xffffff);
349 expected_interval = expected -
stats->expected_prior;
350 stats->expected_prior = expected;
351 received_interval =
stats->received -
stats->received_prior;
353 lost_interval = expected_interval - received_interval;
354 if (expected_interval == 0 || lost_interval <= 0)
357 fraction = (lost_interval << 8) / expected_interval;
359 fraction = (fraction << 24) | lost;
369 uint32_t middle_32_bits =
s->last_rtcp_ntp_time >> 16;
380 len = strlen(
s->hostname);
395 if ((
len > 0) && buf) {
412 bytestream_put_byte(&ptr, 0);
413 bytestream_put_be16(&ptr, 0);
414 bytestream_put_be32(&ptr, 0);
415 bytestream_put_be32(&ptr, 0);
422 bytestream_put_byte(&ptr,
RTCP_RR);
423 bytestream_put_be16(&ptr, 1);
424 bytestream_put_be32(&ptr, 0);
430 uint16_t *missing_mask)
433 uint16_t next_seq =
s->seq + 1;
436 if (!
pkt ||
pkt->seq == next_seq)
440 for (
i = 1;
i <= 16;
i++) {
441 uint16_t missing_seq = next_seq +
i;
443 int16_t
diff =
pkt->seq - missing_seq;
450 if (
pkt->seq == missing_seq)
452 *missing_mask |= 1 << (
i - 1);
455 *first_missing = next_seq;
462 int len, need_keyframe, missing_packets;
466 uint16_t first_missing = 0, missing_mask = 0;
471 need_keyframe =
s->handler &&
s->handler->need_keyframe &&
472 s->handler->need_keyframe(
s->dynamic_protocol_context);
475 if (!need_keyframe && !missing_packets)
482 if (
s->last_feedback_time &&
485 s->last_feedback_time = now;
501 if (missing_packets) {
516 if (
len > 0 && buf) {
545 bytestream_put_byte (&bs, 0x1);
547 bytestream_put_byte (&bs, codecpar->
channels);
549 bytestream_put_le16 (&bs, 0);
551 bytestream_put_le32 (&bs, 48000);
553 bytestream_put_le16 (&bs, 0x0);
555 bytestream_put_byte (&bs, 0x0);
565 int payload_type,
int queue_size)
573 s->payload_type = payload_type;
578 s->queue_size = queue_size;
596 "Error creating opus extradata: %s\n",
607 gethostname(
s->hostname,
sizeof(
s->hostname));
614 s->dynamic_protocol_context =
ctx;
641 delta_timestamp = timestamp -
s->last_rtcp_timestamp;
643 addend =
av_rescale(
s->last_rtcp_ntp_time -
s->first_rtcp_ntp_time,
644 s->st->time_base.den,
645 (uint64_t)
s->st->time_base.num << 32);
646 pkt->
pts =
s->range_start_offset +
s->rtcp_ts_offset + addend +
651 if (!
s->base_timestamp)
652 s->base_timestamp = timestamp;
656 s->unwrapped_timestamp += timestamp;
658 s->unwrapped_timestamp += (
int32_t)(timestamp -
s->timestamp);
659 s->timestamp = timestamp;
660 pkt->
pts =
s->unwrapped_timestamp +
s->range_start_offset -
668 int payload_type, seq,
flags = 0;
674 csrc = buf[0] & 0x0f;
676 payload_type = buf[1] & 0x7f;
686 if (
s->payload_type != payload_type)
693 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
694 payload_type, seq, ((
s->seq + 1) & 0xffff));
699 int padding = buf[
len - 1];
700 if (
len >= 12 + padding)
719 ext = (
AV_RB16(buf + 2) + 1) << 2;
728 if (
s->handler &&
s->handler->parse_packet) {
729 rv =
s->handler->parse_packet(
s->ic,
s->dynamic_protocol_context,
730 s->st,
pkt, ×tamp, buf,
len, seq,
762 uint16_t seq =
AV_RB16(buf + 2);
767 int16_t
diff = seq - (*cur)->seq;
789 return s->queue &&
s->queue->seq == (uint16_t) (
s->seq + 1);
794 return s->queue ?
s->queue->recvtime : 0;
802 if (
s->queue_len <= 0)
807 "RTP: missed %d packets\n",
s->queue->seq -
s->seq - 1);
811 next =
s->queue->next;
831 if (
s->prev_ret <= 0)
834 if (
s->handler &&
s->handler->parse_packet) {
838 rv =
s->handler->parse_packet(
s->ic,
s->dynamic_protocol_context,
865 if ((
s->seq == 0 && !
s->queue) ||
s->queue_size <= 1) {
869 uint16_t seq =
AV_RB16(buf + 2);
870 int16_t
diff = seq -
s->seq;
874 "RTP: dropping old packet received too late\n");
876 }
else if (
diff <= 1) {
888 if (
s->queue_len >=
s->queue_size) {
931 const char *attr,
const char *
value))
936 int value_size = strlen(p) + 1;
944 while (*p && *p ==
' ')
946 while (*p && *p !=
' ')
948 while (*p && *p ==
' ')
953 value, value_size)) {
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
void avio_w8(AVIOContext *s, int b)
void avio_wb32(AVIOContext *s, unsigned int val)
void avio_wb16(AVIOContext *s, unsigned int val)
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
void avio_flush(AVIOContext *s)
Force flushing of buffered data.
int avio_open_dyn_buf(AVIOContext **s)
Open a write only memory stream.
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
#define flags(name, subs,...)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
@ AV_CODEC_ID_GSM
as in Berlin toast format
@ AV_CODEC_ID_TEXT
raw UTF-8 text
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
int av_packet_from_data(AVPacket *pkt, uint8_t *data, int size)
Initialize a reference-counted packet from av_malloc()ed data.
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define av_err2str(errnum)
Convenience macro, the return value should be used only directly in function arguments but never stan...
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
#define AV_LOG_WARNING
Something somehow does not look correct.
#define AV_LOG_VERBOSE
Detailed information.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
int av_strcasecmp(const char *a, const char *b)
Locale-independent case-insensitive compare.
#define AV_NOPTS_VALUE
Undefined timestamp value.
#define AV_TIME_BASE
Internal time base represented as integer.
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
common internal API header
static void handler(vbi_event *ev, void *user_data)
enum AVMediaType codec_type
#define RTP_PT_IS_RTCP(x)
#define RTCP_TX_RATIO_NUM
#define RTCP_TX_RATIO_DEN
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with a matching codec ID.
static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing, uint16_t *missing_mask)
int ff_parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *p, int(*parse_fmtp)(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value))
const RTPDynamicProtocolHandler * ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type)
Find a registered rtp dynamic protocol handler with the specified name.
static const RTPDynamicProtocolHandler opus_dynamic_handler
const RTPDynamicProtocolHandler ff_rdt_live_audio_handler
void ff_rtp_parse_close(RTPDemuxContext *s)
static const RTPDynamicProtocolHandler gsm_dynamic_handler
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
Send a dummy packet on both port pairs to set up the connection state in potential NAT routers,...
static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[]
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params)
const RTPDynamicProtocolHandler ff_rdt_audio_handler
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
static const RTPDynamicProtocolHandler l24_dynamic_handler
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count)
some rtp servers assume client is dead if they don't hear from them...
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, const RTPDynamicProtocolHandler *handler)
static int has_next_packet(RTPDemuxContext *s)
static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
RTPDemuxContext * ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size)
open a new RTP parse context for stream 'st'.
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio)
const RTPDynamicProtocolHandler ff_rdt_live_video_handler
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
const RTPDynamicProtocolHandler * ff_rtp_handler_iterate(void **opaque)
Iterate over all registered rtp dynamic protocol handlers.
const RTPDynamicProtocolHandler ff_rdt_video_handler
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
Parse an RTP or RTCP packet directly sent as a buffer.
static const RTPDynamicProtocolHandler t140_dynamic_handler
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
This was the second switch in rtp_parse packet.
static const RTPDynamicProtocolHandler speex_dynamic_handler
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
static int opus_write_extradata(AVCodecParameters *codecpar)
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len)
int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
Close the dynamic buffer and make a packet from it.
#define MIN_FEEDBACK_INTERVAL
static const RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
#define RTP_MIN_PACKET_LENGTH
#define RTP_FLAG_MARKER
RTP marker bit was set for this packet.
const RTPDynamicProtocolHandler ff_ac3_dynamic_handler
const RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler
const RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler
const RTPDynamicProtocolHandler ff_dv_dynamic_handler
static int parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)
int ff_srtp_set_crypto(struct SRTPContext *s, const char *suite, const char *params)
int ff_srtp_decrypt(struct SRTPContext *s, uint8_t *buf, int *lenptr)
void ff_srtp_free(struct SRTPContext *s)
This struct describes the properties of an encoded stream.
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
int sample_rate
Audio only.
This structure stores compressed data.
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed.
AVCodecParameters * codecpar
Codec parameters associated with this stream.
int index
stream index in AVFormatContext
RTP/JPEG specific private data.
int64_t av_gettime_relative(void)
Get the current time in microseconds since some unspecified starting point.
unbuffered private I/O API
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static void stats(AVPacket *const *in, int n_in, unsigned *_max, unsigned *_sum)