Go to the documentation of this file.
48 #define RTCP_SR_SIZE 28
102 if (
s1->nb_streams != 1) {
113 if (
s->payload_type < 0) {
118 s->payload_type = st->
id;
121 st->
id =
s->payload_type;
125 s->timestamp =
s->base_timestamp;
126 s->cur_timestamp = 0;
133 s->first_rtcp_ntp_time = (
s1->start_time_realtime / 1000) * 1000 +
146 if (
s1->packet_size) {
147 if (
s1->pb->max_packet_size)
149 s1->pb->max_packet_size);
151 s1->packet_size =
s1->pb->max_packet_size;
152 if (
s1->packet_size <= 12) {
160 s->max_payload_size =
s1->packet_size - 12;
171 s->buf_ptr =
s->buf + 4;
186 "Packetizing VC-2 is experimental and does not use all values "
187 "of the specification "
188 "(even though most receivers may handle it just fine). "
189 "Please set -strict experimental in order to enable it.\n");
197 "Packetizing H.261 is experimental and produces incorrect "
198 "packetization for cases where GOBs don't fit into packets "
199 "(even though most receivers may handle it just fine). "
200 "Please set -f_strict experimental in order to enable it.\n");
223 "Packetizing VP9 is experimental and its specification is "
224 "still in draft state. "
225 "Please set -strict experimental in order to enable it.\n");
232 s->max_frames_per_packet = 15;
258 s->max_frames_per_packet = 50;
264 if (1 +
s->max_frames_per_packet + n >
s->max_payload_size) {
274 s->max_frames_per_packet = 50;
295 s->last_rtcp_ntp_time = ntp_time;
297 s1->streams[0]->time_base) +
s->base_timestamp;
303 avio_wb32(
s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
343 avio_w8(
s1->pb, (
s->payload_type & 0x7f) | ((m & 0x01) << 7));
351 s->seq = (
s->seq + 1) & 0xffff;
352 s->octet_count +=
len;
359 const uint8_t *buf1,
int size,
int sample_size_bits)
362 int len, max_packet_size, n;
364 int aligned_samples_size = sample_size_bits/
av_gcd(sample_size_bits, 8);
366 max_packet_size = (
s->max_payload_size / aligned_samples_size) * aligned_samples_size;
368 if ((sample_size_bits % 8) == 0 && ((8 *
size) % sample_size_bits) != 0)
376 memcpy(
s->buf_ptr, buf1,
len);
380 s->timestamp =
s->cur_timestamp + n * 8 / sample_size_bits;
382 n += (
s->buf_ptr -
s->buf);
388 const uint8_t *buf1,
int size)
391 int len, count, max_packet_size;
393 max_packet_size =
s->max_payload_size;
396 len = (
s->buf_ptr -
s->buf);
397 if ((
len +
size) > max_packet_size) {
400 s->buf_ptr =
s->buf + 4;
403 if (
s->buf_ptr ==
s->buf + 4) {
404 s->timestamp =
s->cur_timestamp;
408 if (
size > max_packet_size) {
412 len = max_packet_size - 4;
418 s->buf[2] = count >> 8;
420 memcpy(
s->buf + 4, buf1,
len);
427 if (
s->buf_ptr ==
s->buf + 4) {
434 memcpy(
s->buf_ptr, buf1,
size);
440 const uint8_t *buf1,
int size)
443 int len, max_packet_size;
445 max_packet_size =
s->max_payload_size;
448 len = max_packet_size;
452 s->timestamp =
s->cur_timestamp;
462 const uint8_t *buf1,
int size)
467 s->timestamp =
s->cur_timestamp;
469 len =
s->max_payload_size - (
s->buf_ptr -
s->buf);
472 memcpy(
s->buf_ptr, buf1,
len);
477 out_len =
s->buf_ptr -
s->buf;
478 if (out_len >=
s->max_payload_size) {
494 if (
s->num_frames > 0 &&
501 if (!
s->num_frames) {
503 s->timestamp =
s->cur_timestamp;
510 s->cur_timestamp += frame_duration;
512 if (
s->num_frames ==
s->max_frames_per_packet) {
535 s->last_octet_count =
s->octet_count;
538 s->cur_timestamp =
s->base_timestamp +
pkt->
pts;
635 if (
size >
s->max_payload_size) {
637 "Packet size %d too large for max RTP payload size %d\n",
638 size,
s->max_payload_size);
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
#define AVERROR_EXPERIMENTAL
Requested feature is flagged experimental. Set strict_std_compliance if you really want to use it.
static int rtp_write_header(AVFormatContext *s1)
#define FF_RTP_FLAG_MP4A_LATM
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
enum AVMediaType codec_type
General type of the encoded data.
void ff_rtp_send_jpeg(AVFormatContext *s1, const uint8_t *buff, int size)
int av_compare_ts(int64_t ts_a, AVRational tb_a, int64_t ts_b, AVRational tb_b)
Compare two timestamps each in its own time base.
void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size, const uint8_t *mb_info, int mb_info_size)
void ff_rtp_send_h261(AVFormatContext *s1, const uint8_t *buf1, int size)
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
int nb_channels
Number of channels in this layout.
#define FF_RTP_FLAG_RFC2190
#define FF_RTP_FLAG_OPTS(ctx, fieldname)
uint32_t av_get_random_seed(void)
Get a seed to use in conjunction with random functions.
int64_t av_gcd(int64_t a, int64_t b)
Compute the greatest common divisor of two integer operands.
void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
if it could not because there are no more frames
static int rtp_write_trailer(AVFormatContext *s1)
void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size)
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
void ff_rtp_send_raw_rfc4175(AVFormatContext *s1, const uint8_t *buf, int size, int interlaced, int field)
#define FF_RTP_FLAG_SKIP_RTCP
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
static int is_supported(enum AVCodecID id)
static const AVOption options[]
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define RTCP_TX_RATIO_NUM
#define RTCP_TX_RATIO_DEN
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
void avio_flush(AVIOContext *s)
Force flushing of buffered data.
AVCodecParameters * codecpar
Codec parameters associated with this stream.
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
Rational number (pair of numerator and denominator).
const char * av_default_item_name(void *ptr)
Return the context name.
@ AV_CODEC_ID_MPEG2TS
FAKE codec to indicate a raw MPEG-2 TS stream (only used by libavformat)
void avio_w8(AVIOContext *s, int b)
static const AVClass rtp_muxer_class
AVChannelLayout ch_layout
Audio only.
#define AV_OPT_FLAG_ENCODING_PARAM
A generic parameter which can be set by the user for muxing or encoding.
int sample_rate
Audio only.
AVCodecID
Identify the syntax and semantics of the bitstream.
int extradata_size
Size of the extradata content in bytes.
static int rtp_send_samples(AVFormatContext *s1, const uint8_t *buf1, int size, int sample_size_bits)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
#define AV_NOPTS_VALUE
Undefined timestamp value.
int av_get_audio_frame_duration2(AVCodecParameters *par, int frame_bytes)
This function is the same as av_get_audio_frame_duration(), except it works with AVCodecParameters in...
static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size)
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
void avio_wb32(AVIOContext *s, unsigned int val)
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
const char * avcodec_get_name(enum AVCodecID id)
Get the name of a codec.
@ AV_PKT_DATA_H263_MB_INFO
An AV_PKT_DATA_H263_MB_INFO side data packet contains a number of structures with info about macroblo...
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, size_t *size)
Get side information from packet.
void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size)
int block_align
Audio only.
#define FF_RTP_FLAG_SEND_BYE
void ff_rtp_send_vp9(AVFormatContext *s1, const uint8_t *buff, int size)
void ff_rtp_send_vc2hq(AVFormatContext *s1, const uint8_t *buf, int size, int interlaced)
const FFOutputFormat ff_rtp_muxer
enum AVFieldOrder field_order
Video only.
int id
Format-specific stream ID.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
void ff_rtp_send_h264_hevc(AVFormatContext *s1, const uint8_t *buf1, int size)
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.
@ AV_CODEC_ID_ADPCM_G726LE
void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size)
Packetize H.263 frames into RTP packets according to RFC 4629.
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
int bits_per_coded_sample
The number of bits per sample in the codedwords.
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
This structure stores compressed data.
void avio_wb16(AVIOContext *s, unsigned int val)
#define flags(name, subs,...)
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.
static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size)
@ AV_CODEC_ID_MPEG2VIDEO
preferred ID for MPEG-1/2 video decoding
int ff_rtp_get_payload_type(const AVFormatContext *fmt, const AVCodecParameters *par, int idx)
Return the payload type for a given stream used in the given format context.