36 {
"seq",
"Starting sequence number", offsetof(
RTPMuxContext, seq),
AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
47 #define RTCP_SR_SIZE 28 156 if (frame_size == 0) {
162 (
AVRational){ frame_size, st->codec->sample_rate },
297 for (len = (7 + len) % 4; len % 4; len++)
317 av_dlog(s1,
"rtp_send_data size=%d\n", len);
329 s->
seq = (s->
seq + 1) & 0xffff;
337 const uint8_t *buf1,
int size,
int sample_size_bits)
340 int len, max_packet_size, n;
342 int aligned_samples_size = sample_size_bits/
av_gcd(sample_size_bits, 8);
344 max_packet_size = (s->
max_payload_size / aligned_samples_size) * aligned_samples_size;
346 if ((sample_size_bits % 8) == 0 && ((8 *
size) % sample_size_bits) != 0)
351 len =
FFMIN(max_packet_size, size);
369 int len, count, max_packet_size;
375 if ((len + size) > max_packet_size) {
386 if (size > max_packet_size) {
390 len = max_packet_size - 4;
396 s->
buf[2] = count >> 8;
398 memcpy(s->
buf + 4, buf1, len);
412 memcpy(s->
buf_ptr, buf1, size);
421 int len, max_packet_size;
426 len = max_packet_size;
477 memcpy(s->
buf_ptr, buf, n * frame_size);
558 int mb_info_size = 0;
585 "Packet size %d too large for max RTP payload size %d\n",
620 .priv_class = &rtp_muxer_class,
void avio_wb64(AVIOContext *s, uint64_t val)
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
static int rtp_send_samples(AVFormatContext *s1, const uint8_t *buf1, int size, int sample_size_bits)
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
int64_t last_rtcp_ntp_time
#define RTCP_TX_RATIO_NUM
unsigned int last_octet_count
static const AVOption options[]
void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size)
Packetize AMR frames into RTP packets according to RFC 3267, in octet-aligned mode.
#define RTCP_TX_RATIO_DEN
An AV_PKT_DATA_H263_MB_INFO side data packet contains a number of structures with info about macroblo...
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
int nal_length_size
Number of bytes used for H.264 NAL length, if the MP4 syntax is used (1, 2 or 4)
#define FF_RTP_FLAG_MP4A_LATM
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (%s)\, len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic ? ac->func_descr_generic :ac->func_descr)
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
static const AVClass rtp_muxer_class
int id
Format-specific stream ID.
int max_frames_per_packet
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size)
Packetize H.263 frames into RTP packets according to RFC 4629.
AVStream ** streams
A list of all streams in the file.
static int rtp_write_header(AVFormatContext *s1)
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
#define AV_OPT_FLAG_ENCODING_PARAM
a generic parameter which can be set by the user for muxing or encoding
#define FF_RTP_FLAG_RFC2190
int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec, int idx)
Return the payload type for a given stream used in the given format context.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
static const uint8_t frame_size[4]
AVCodecID
Identify the syntax and semantics of the bitstream.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
void ff_rtp_send_jpeg(AVFormatContext *s1, const uint8_t *buff, int size)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
preferred ID for decoding MPEG audio layer 1, 2 or 3
void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size)
void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size)
int64_t av_gcd(int64_t a, int64_t b)
Return the greatest common divisor of a and b.
void av_log(void *avcl, int level, const char *fmt,...)
AVRational avg_frame_rate
Average framerate.
#define FF_RTP_FLAG_SKIP_RTCP
static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
AVCodecContext * codec
Codec context associated with this stream.
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
int64_t av_rescale_q_rnd(int64_t a, AVRational bq, AVRational cq, enum AVRounding rnd)
Rescale a 64-bit integer by 2 rational numbers with specified rounding.
void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
Packetize Xiph frames into RTP according to RFC 5215 (Vorbis) and the Theora RFC draft.
int void avio_flush(AVIOContext *s)
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size)
preferred ID for MPEG-1/2 video decoding
int frame_size
Number of samples per channel in an audio frame.
int64_t first_rtcp_ntp_time
AVOutputFormat ff_rtp_muxer
enum AVMediaType codec_type
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
int sample_rate
samples per second
AVIOContext * pb
I/O context.
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
void avio_w8(AVIOContext *s, int b)
static int is_supported(enum AVCodecID id)
Describe the class of an AVClass context structure.
rational number numerator/denominator
int av_get_audio_frame_duration(AVCodecContext *avctx, int frame_bytes)
Return audio frame duration.
void avio_wb16(AVIOContext *s, unsigned int val)
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
#define NTP_TO_RTP_FORMAT(x)
void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size)
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
unsigned int packet_count
FAKE codec to indicate a raw MPEG-2 TS stream (only used by libavformat)
int channels
number of audio channels
void * priv_data
Format private data.
static void write_header(FFV1Context *f)
static int rtp_write_trailer(AVFormatContext *s1)
void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size, const uint8_t *mb_info, int mb_info_size)
void avio_wb32(AVIOContext *s, unsigned int val)
uint8_t * av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
uint32_t av_get_random_seed(void)
Get random data.
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
This structure stores compressed data.
#define FF_RTP_FLAG_SEND_BYE
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...