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e2f0bb34d7e3474b10aafe71308236fedd91ee73
UltraGrid/src/transmit.cpp
Martin Pulec d17c5ea24c removed platform_time.c + references 
get_time_in_ns() should be used instead time_since_epoch_in_ms(). Having
both may be misleading and may lead to errors when interchanged.
2023-08-08 15:32:29 +02:00

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/*
* FILE: transmit.cpp
* AUTHOR: Colin Perkins <csp@csperkins.org>
* Ladan Gharai
* Martin Benes <martinbenesh@gmail.com>
* Lukas Hejtmanek <xhejtman@ics.muni.cz>
* Petr Holub <hopet@ics.muni.cz>
* Milos Liska <xliska@fi.muni.cz>
* Jiri Matela <matela@ics.muni.cz>
* Dalibor Matura <255899@mail.muni.cz>
* Ian Wesley-Smith <iwsmith@cct.lsu.edu>
* David Cassany <david.cassany@i2cat.net>
* Ignacio Contreras <ignacio.contreras@i2cat.net>
* Gerard Castillo <gerard.castillo@i2cat.net>
* Jordi "Txor" Casas Ríos <txorlings@gmail.com>
* Martin Pulec <pulec@cesnet.cz>
*
* Copyright (c) 2005-2010 Fundació i2CAT, Internet I Innovació Digital a Catalunya
* Copyright (c) 2001-2004 University of Southern California
* Copyright (c) 2005-2023 CESNET z.s.p.o.
*
* Redistribution and use in source and binary forms, with or without
* modification, is permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
*
* This product includes software developed by the University of Southern
* California Information Sciences Institute. This product also includes
* software developed by CESNET z.s.p.o.
*
* 4. Neither the name of the University, Institute, CESNET nor the names of
* its contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING,
* BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
* AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
* EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#include "config_unix.h"
#include "config_win32.h"
#endif // HAVE_CONFIG_H
#include "audio/codec.h"
#include "audio/types.h"
#include "audio/utils.h"
#include "control_socket.h"
#include "debug.h"
#include "host.h"
#include "lib_common.h"
#include "crypto/openssl_encrypt.h"
#include "module.h"
#include "rtp/fec.h"
#include "rtp/rtp.h"
#include "rtp/rtp_callback.h"
#include "rtp/rtpenc_h264.h"
#include "tv.h"
#include "transmit.h"
#include "utils/jpeg_reader.h"
#include "utils/misc.h" // unit_evaluate
#include "utils/random.h"
#include "video.h"
#include "video_codec.h"
#include <algorithm>
#include <array>
#include <iostream>
#include <vector>
#define MOD_NAME "[transmit] "
#define TRANSMIT_MAGIC 0xe80ab15f
#define FEC_MAX_MULT 10
#define CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_NS NS_IN_SEC
#ifdef HAVE_MACOSX
#define GET_STARTTIME gettimeofday(&start, NULL)
#define GET_STOPTIME gettimeofday(&stop, NULL)
#define GET_DELTA delta = (stop.tv_sec - start.tv_sec) * 1000000000l + (stop.tv_usec - start.tv_usec) * 1000L
#elif defined HAVE_LINUX
#define GET_STARTTIME clock_gettime(CLOCK_REALTIME, &start)
#define GET_STOPTIME clock_gettime(CLOCK_REALTIME, &stop)
#define GET_DELTA delta = (stop.tv_sec - start.tv_sec) * 1000000000l + stop.tv_nsec - start.tv_nsec
#else // Windows
#define GET_STARTTIME {QueryPerformanceFrequency(&freq); QueryPerformanceCounter(&start); }
#define GET_STOPTIME { QueryPerformanceCounter(&stop); }
#define GET_DELTA delta = (long)((double)(stop.QuadPart - start.QuadPart) * 1000 * 1000 * 1000 / freq.QuadPart);
#endif
#define DEFAULT_CIPHER_MODE MODE_AES128_GCM
using std::array;
using std::vector;
static void tx_update(struct tx *tx, struct video_frame *frame, int substream);
static void tx_done(struct module *tx);
static uint32_t format_interl_fps_hdr_row(enum interlacing_t interlacing, double input_fps);
static void
tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
uint32_t ts, int send_m,
unsigned int substream,
int fragment_offset);
static bool set_fec(struct tx *tx, const char *fec);
static void fec_check_messages(struct tx *tx);
struct rate_limit_dyn {
unsigned long avg_frame_size; ///< moving average
long long last_excess; ///< nr of frames last excessive frame was emitted
static constexpr int EXCESS_GAP = 4; ///< minimal gap between excessive frames
};
struct tx {
struct module mod;
uint32_t magic;
enum tx_media_type media_type;
unsigned mtu;
double max_loss;
uint32_t last_ts;
int last_frame_fragment_id;
unsigned int buffer:22;
unsigned long int sent_frames;
int32_t avg_len;
int32_t avg_len_last;
enum fec_type fec_scheme;
int mult_count;
int last_fragment;
struct control_state *control = nullptr;
size_t sent_since_report = 0;
uint64_t last_stat_report = 0;
const struct openssl_encrypt_info *enc_funcs;
struct openssl_encrypt *encryption;
long long int bitrate;
struct rate_limit_dyn dyn_rate_limit_state;
char tmp_packet[RTP_MAX_MTU];
};
static void tx_update(struct tx *tx, struct video_frame *frame, int substream)
{
if(!frame) {
return;
}
uint64_t tmp_avg = tx->avg_len * tx->sent_frames + frame->tiles[substream].data_len *
(frame->fec_params.type != FEC_NONE ?
(double) frame->fec_params.k / (frame->fec_params.k + frame->fec_params.m) :
1);
tx->sent_frames++;
tx->avg_len = tmp_avg / tx->sent_frames;
if(tx->sent_frames >= 100) {
if(tx->fec_scheme == FEC_LDGM && tx->max_loss > 0.0) {
if(abs(tx->avg_len_last - tx->avg_len) > tx->avg_len / 3) {
int data_len = tx->mtu - (40 + (sizeof(fec_payload_hdr_t)));
data_len = (data_len / 48) * 48;
//void *fec_state_old = tx->fec_state;
struct msg_sender *msg = (struct msg_sender *)
new_message(sizeof(struct msg_sender));
snprintf(msg->fec_cfg, sizeof(msg->fec_cfg), "LDGM percents %d %d %f",
data_len, tx->avg_len, tx->max_loss);
msg->type = SENDER_MSG_CHANGE_FEC;
struct response *resp = send_message_to_receiver(get_parent_module(&tx->mod),
(struct message *) msg);
free_response(resp);
tx->avg_len_last = tx->avg_len;
}
}
tx->avg_len = 0;
tx->sent_frames = 0;
}
}
struct tx *tx_init(struct module *parent, unsigned mtu, enum tx_media_type media_type,
const char *fec, const char *encryption, long long int bitrate)
{
if (mtu > RTP_MAX_MTU) {
log_msg(LOG_LEVEL_ERROR, "Requested MTU exceeds maximal value allowed by RTP library (%d B).\n", RTP_MAX_MTU);
return NULL;
}
if (bitrate < RATE_MIN) {
log_msg(LOG_LEVEL_ERROR, "Invalid bitrate value %lld passed (either positive bitrate or magic values from %d supported)!\n", bitrate, RATE_MIN);
return NULL;
}
struct tx *tx = (struct tx *) calloc(1, sizeof(struct tx));
if (tx == nullptr) {
return tx;
}
module_init_default(&tx->mod);
tx->mod.cls = MODULE_CLASS_TX;
tx->mod.priv_data = tx;
tx->mod.deleter = tx_done;
module_register(&tx->mod, parent);
tx->magic = TRANSMIT_MAGIC;
tx->media_type = media_type;
tx->mult_count = 1;
tx->max_loss = 0.0;
tx->mtu = mtu;
tx->buffer = ug_rand() & 0x3fffff;
tx->avg_len = tx->avg_len_last = tx->sent_frames = 0u;
tx->fec_scheme = FEC_NONE;
tx->last_frame_fragment_id = -1;
if (fec) {
if(!set_fec(tx, fec)) {
module_done(&tx->mod);
return NULL;
}
}
if (encryption) {
tx->enc_funcs = static_cast<const struct openssl_encrypt_info *>(load_library("openssl_encrypt",
LIBRARY_CLASS_UNDEFINED, OPENSSL_ENCRYPT_ABI_VERSION));
if (!tx->enc_funcs) {
fprintf(stderr, "UltraGrid was build without OpenSSL support!\n");
module_done(&tx->mod);
return NULL;
}
if (tx->enc_funcs->init(&tx->encryption,
encryption, DEFAULT_CIPHER_MODE) != 0) {
log_msg(LOG_LEVEL_ERROR, MOD_NAME "Unable to initialize encryption\n");
module_done(&tx->mod);
return NULL;
}
}
tx->bitrate = bitrate;
tx->control = (struct control_state *) get_module(get_root_module(parent), "control");
return tx;
}
static bool set_fec(struct tx *tx, const char *fec_const)
{
char *fec = strdup(fec_const);
bool ret = true;
char *fec_cfg = NULL;
if(strchr(fec, ':')) {
char *delim = strchr(fec, ':');
*delim = '\0';
fec_cfg = delim + 1;
}
struct msg_sender *msg = (struct msg_sender *)
new_message(sizeof(struct msg_sender));
msg->type = SENDER_MSG_CHANGE_FEC;
snprintf(msg->fec_cfg, sizeof(msg->fec_cfg), "flush");
tx->mult_count = 1; // default
if (strcasecmp(fec, "none") == 0) {
tx->fec_scheme = FEC_NONE;
} else if(strcasecmp(fec, "mult") == 0) {
tx->fec_scheme = FEC_MULT;
assert(fec_cfg);
tx->mult_count = (unsigned int) atoi(fec_cfg);
assert(tx->mult_count <= FEC_MAX_MULT);
} else if(strcasecmp(fec, "LDGM") == 0) {
if(tx->media_type == TX_MEDIA_AUDIO) {
fprintf(stderr, "LDGM is not currently supported for audio!\n");
ret = false;
} else {
if(!fec_cfg || (strlen(fec_cfg) > 0 && strchr(fec_cfg, '%') == NULL)) {
snprintf(msg->fec_cfg, sizeof(msg->fec_cfg), "LDGM cfg %s",
fec_cfg ? fec_cfg : "");
} else { // delay creation until we have avarage frame size
tx->max_loss = atof(fec_cfg);
}
tx->fec_scheme = FEC_LDGM;
}
} else if(strcasecmp(fec, "RS") == 0) {
snprintf(msg->fec_cfg, sizeof(msg->fec_cfg), "RS cfg %s",
fec_cfg ? fec_cfg : "");
tx->fec_scheme = FEC_RS;
} else if(strcasecmp(fec, "help") == 0) {
std::cout << "Usage:\n"
"\t-f [A:|V:]{ mult:count | ldgm[:params] | rs[:params] }\n";
ret = false;
} else {
fprintf(stderr, "Unknown FEC: %s\n", fec);
ret = false;
}
if (ret) {
struct response *resp = send_message_to_receiver(get_parent_module(&tx->mod),
(struct message *) msg);
free_response(resp);
} else {
free_message((struct message *) msg, NULL);
}
free(fec);
return ret;
}
static void fec_check_messages(struct tx *tx)
{
struct message *msg;
while ((msg = check_message(&tx->mod))) {
auto *data = reinterpret_cast<struct msg_universal *>(msg);
const char *text = data->text;
if (strstr(text, MSG_UNIVERSAL_TAG_TX) != text) {
LOG(LOG_LEVEL_ERROR) << "[Transmit] Unexpected TX message: " << text << "\n";
free_message(msg, new_response(RESPONSE_BAD_REQUEST, "Unexpected message"));
continue;
}
text += strlen(MSG_UNIVERSAL_TAG_TX);
struct response *r = nullptr;
if (strstr(text, "fec ") == text) {
text += strlen("fec ");
if (set_fec(tx, text)) {
r = new_response(RESPONSE_OK, nullptr);
LOG(LOG_LEVEL_NOTICE) << "[Transmit] FEC set to new setting: " << text << "\n";
} else {
r = new_response(RESPONSE_INT_SERV_ERR, "cannot set FEC");
LOG(LOG_LEVEL_ERROR) << "[Transmit] Unable to reconfiure FEC to: " << text << "\n";
}
} else if (strstr(text, "rate ") == text) {
text += strlen("rate ");
auto new_rate = unit_evaluate(text);
if (new_rate >= RATE_MIN) {
tx->bitrate = new_rate;
r = new_response(RESPONSE_OK, nullptr);
LOG(LOG_LEVEL_NOTICE) << "[Transmit] Bitrate set to: " << text << (new_rate > 0 ? "B" : "") << "\n";
} else {
r = new_response(RESPONSE_BAD_REQUEST, "Wrong value for bitrate");
LOG(LOG_LEVEL_ERROR) << "[Transmit] Wrong bitrate: " << text << "\n";
}
} else {
r = new_response(RESPONSE_BAD_REQUEST, "Unknown TX message");
LOG(LOG_LEVEL_ERROR) << "[Transmit] Unknown TX message: " << text << "\n";
}
free_message(msg, r);
}
}
static void tx_done(struct module *mod)
{
struct tx *tx = (struct tx *) mod->priv_data;
assert(tx->magic == TRANSMIT_MAGIC);
free(tx);
}
/*
* sends one or more frames (tiles) with same TS in one RTP stream. Only one m-bit is set.
*/
void
tx_send(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session)
{
unsigned int i;
assert(!frame->fragment || tx->fec_scheme == FEC_NONE); // currently no support for FEC with fragments
assert(!frame->fragment || frame->tile_count); // multiple tile are not currently supported for fragmented send
fec_check_messages(tx);
uint32_t ts =
incompatible_features && (frame->flags & TIMESTAMP_VALID) != 0
? get_local_mediatime_offset() + frame->timestamp
: get_local_mediatime();
if(frame->fragment &&
tx->last_frame_fragment_id == frame->frame_fragment_id) {
ts = tx->last_ts;
} else {
tx->last_frame_fragment_id = frame->frame_fragment_id;
tx->last_ts = ts;
}
for(i = 0; i < frame->tile_count; ++i)
{
int last = FALSE;
int fragment_offset = 0;
if (i == frame->tile_count - 1) {
if(!frame->fragment || frame->last_fragment)
last = TRUE;
}
if(frame->fragment)
fragment_offset = vf_get_tile(frame, i)->offset;
tx_send_base(tx, frame, rtp_session, ts, last,
i, fragment_offset);
}
tx->buffer++;
}
void format_video_header(struct video_frame *frame, int tile_idx, int buffer_idx, uint32_t *video_hdr)
{
uint32_t tmp;
video_hdr[3] = htonl(frame->tiles[tile_idx].width << 16 | frame->tiles[tile_idx].height);
video_hdr[4] = get_fourcc(frame->color_spec);
video_hdr[2] = htonl(frame->tiles[tile_idx].data_len);
tmp = tile_idx << 22;
tmp |= 0x3fffff & buffer_idx;
video_hdr[0] = htonl(tmp);
/* word 6 */
video_hdr[5] = format_interl_fps_hdr_row(frame->interlacing, frame->fps);
}
void
format_audio_header(struct audio_desc desc, int channel, size_t data_len,
int buffer_idx, uint32_t *audio_hdr)
{
uint32_t tmp = 0;
tmp = channel << 22U; /* bits 0-9 */
tmp |= buffer_idx; /* bits 10-31 */
audio_hdr[0] = htonl(tmp);
audio_hdr[2] = htonl(data_len);
/* fourth word */
tmp = (desc.bps * 8) << 26U;
tmp |= desc.sample_rate;
audio_hdr[3] = htonl(tmp);
/* fifth word */
audio_hdr[4] = htonl(get_audio_tag(desc.codec));
}
static uint32_t format_interl_fps_hdr_row(enum interlacing_t interlacing, double input_fps)
{
unsigned int fpsd, fd, fps, fi;
uint32_t tmp;
tmp = interlacing << 29;
fps = round(input_fps);
fpsd = 1; /// @todo make use of this value (for now it is always one)
fd = 0;
fi = 0;
if (input_fps > 1.0 && fabs(input_fps - round(input_fps) / 1.001) < 0.005) { // 29.97 etc.
fd = 1;
} else if (fps < 1.0) {
fps = round(1.0 / input_fps);
fi = 1;
}
tmp |= fps << 19;
tmp |= fpsd << 15;
tmp |= fd << 14;
tmp |= fi << 13;
return htonl(tmp);
}
static inline void check_symbol_size(int fec_symbol_size, int payload_len)
{
thread_local static bool status_printed = false;
if (status_printed) {
return;
}
if (fec_symbol_size > payload_len) {
LOG(LOG_LEVEL_WARNING) << "Warning: FEC symbol size exceeds payload size! "
"FEC symbol size: " << fec_symbol_size << "\n";
} else {
LOG(LOG_LEVEL_INFO) << "FEC symbol size: " << fec_symbol_size << ", symbols per packet: " <<
payload_len / fec_symbol_size << ", payload size: " <<
payload_len / fec_symbol_size * fec_symbol_size << "\n";
}
status_printed = true;
}
/**
* Splits symbol (FEC symbol or uncompressed line) to 1 or more MTUs. Symbol starts
* always on beginning of packet.
*
* If symbol_size is longer than MTU (more symbols fit one packet), the aligned
* packet size is always the same.
*
* @param symbol_size FEC symbol size or linesize for uncompressed
*/
static inline int get_video_pkt_len(int mtu,
int symbol_size, int *symbol_offset)
{
if (symbol_size > mtu) {
if (symbol_size - *symbol_offset <= mtu) {
mtu = symbol_size - *symbol_offset;
*symbol_offset = 0;
} else {
*symbol_offset += mtu;
}
return mtu;
}
return mtu / symbol_size * symbol_size;
}
/// @param mtu is tx->mtu - hdrs_len
static vector<int> get_packet_sizes(struct video_frame *frame, int substream, int mtu) {
if (frame->fec_params.type != FEC_NONE) {
check_symbol_size(frame->fec_params.symbol_size, mtu);
}
unsigned int symbol_size = 1;
int symbol_offset = 0;
if (frame->fec_params.type == FEC_NONE && !is_codec_opaque(frame->color_spec)) {
symbol_size = vc_get_linesize(frame->tiles[substream].width, frame->color_spec);
int pf_block_size = PIX_BLOCK_LCM / get_pf_block_pixels(frame->color_spec) * get_pf_block_bytes(frame->color_spec);
assert(pf_block_size <= mtu);
mtu = mtu / pf_block_size * pf_block_size;
} else if (frame->fec_params.type != FEC_NONE) {
symbol_size = frame->fec_params.symbol_size;
}
vector<int> ret;
unsigned pos = 0;
do {
int len = symbol_size == 1
? mtu
: get_video_pkt_len(mtu, symbol_size, &symbol_offset);
pos += len;
ret.push_back(len);
} while (pos < frame->tiles[substream].data_len);
return ret;
}
/**
* Returns inter-packet interval in nanoseconds.
*/
static long
get_packet_rate(struct tx *tx, struct video_frame *frame, int substream, long packet_count)
{
if (tx->bitrate == RATE_UNLIMITED) {
return 0;
}
double time_for_frame = 1.0 / frame->fps / frame->tile_count;
double interval_between_pkts = time_for_frame / tx->mult_count / packet_count;
// use only 75% of the time - we less likely overshot the frame time and
// can minimize risk of swapping packets between 2 frames (out-of-order ones)
interval_between_pkts = interval_between_pkts * 0.75;
// prevent bitrate to be "too low", here 1 Mbps at minimum
interval_between_pkts = std::min<double>(interval_between_pkts, tx->mtu / 1000'000.0);
long packet_rate_auto = interval_between_pkts * 1000'000'000L;
if (tx->bitrate == RATE_AUTO) { // adaptive (spread packets to 75% frame time)
return packet_rate_auto;
}
if (tx->bitrate == RATE_DYNAMIC) {
if (frame->tiles[substream].data_len > 2 * tx->dyn_rate_limit_state.avg_frame_size
&& tx->dyn_rate_limit_state.last_excess > rate_limit_dyn::EXCESS_GAP) {
packet_rate_auto /= 2; // double packet rate for this frame
tx->dyn_rate_limit_state.last_excess = 0;
} else {
tx->dyn_rate_limit_state.last_excess += 1;
}
tx->dyn_rate_limit_state.avg_frame_size = (9 * tx->dyn_rate_limit_state.avg_frame_size + frame->tiles[substream].data_len) / 10;
return packet_rate_auto;
}
long long int bitrate = tx->bitrate & ~RATE_FLAG_FIXED_RATE;
int avg_packet_size = frame->tiles[substream].data_len / packet_count;
long packet_rate = 1000'000'000L * avg_packet_size * 8 / bitrate; // fixed rate
if ((tx->bitrate & RATE_FLAG_FIXED_RATE) == 0) { // adaptive capped rate
packet_rate = std::max(packet_rate, packet_rate_auto);
}
return packet_rate;
}
static void
tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
uint32_t ts, int send_m,
unsigned int substream,
int fragment_offset)
{
if (!rtp_has_receiver(rtp_session)) {
return;
}
struct tile *tile = &frame->tiles[substream];
int data_len;
// see definition in rtp_callback.h
uint32_t rtp_hdr[100];
int rtp_hdr_len;
int pt = fec_pt_from_fec_type(TX_MEDIA_VIDEO, frame->fec_params.type, tx->encryption); /* A value specified in our packet format */
#ifdef HAVE_LINUX
struct timespec start, stop;
#elif defined HAVE_MACOSX
struct timeval start, stop;
#else // Windows
LARGE_INTEGER start, stop, freq;
#endif
long delta, overslept = 0;
array <int, FEC_MAX_MULT> mult_pos{};
int mult_index = 0;
int hdrs_len = (rtp_is_ipv6(rtp_session) ? 40 : 20) + 8 + 12; // IP hdr size + UDP hdr size + RTP hdr size
assert(tx->magic == TRANSMIT_MAGIC);
tx_update(tx, frame, substream);
if (frame->fec_params.type == FEC_NONE) {
hdrs_len += (sizeof(video_payload_hdr_t));
rtp_hdr_len = sizeof(video_payload_hdr_t);
format_video_header(frame, substream, tx->buffer, rtp_hdr);
} else {
hdrs_len += (sizeof(fec_payload_hdr_t));
rtp_hdr_len = sizeof(fec_payload_hdr_t);
uint32_t tmp = substream << 22;
tmp |= 0x3fffff & tx->buffer;
// see definition in rtp_callback.h
rtp_hdr[0] = htonl(tmp);
rtp_hdr[2] = htonl(tile->data_len);
rtp_hdr[3] = htonl(
frame->fec_params.k << 19 |
frame->fec_params.m << 6 |
frame->fec_params.c);
rtp_hdr[4] = htonl(frame->fec_params.seed);
}
if (tx->encryption) {
hdrs_len += sizeof(crypto_payload_hdr_t) + tx->enc_funcs->get_overhead(tx->encryption);
rtp_hdr[rtp_hdr_len / sizeof(uint32_t)] = htonl(DEFAULT_CIPHER_MODE << 24);
rtp_hdr_len += sizeof(crypto_payload_hdr_t);
}
vector<int> packet_sizes = get_packet_sizes(frame, substream, tx->mtu - hdrs_len);
long packet_count = packet_sizes.size() * (tx->fec_scheme == FEC_MULT ? tx->mult_count : 1);
long packet_rate = get_packet_rate(tx, frame, substream, packet_count);
// initialize header array with values (except offset which is different among
// different packts)
void *rtp_headers = malloc(packet_count * rtp_hdr_len);
uint32_t *rtp_hdr_packet = (uint32_t *) rtp_headers;
for (int i = 0; i < packet_count; ++i) {
memcpy(rtp_hdr_packet, rtp_hdr, rtp_hdr_len);
rtp_hdr_packet += rtp_hdr_len / sizeof(uint32_t);
}
rtp_hdr_packet = (uint32_t *) rtp_headers;
if (!tx->encryption) {
rtp_async_start(rtp_session, packet_count);
}
int packet_idx = 0;
unsigned pos = 0;
do {
GET_STARTTIME;
int m = 0;
if(tx->fec_scheme == FEC_MULT) {
pos = mult_pos[mult_index];
}
int offset = pos + fragment_offset;
rtp_hdr_packet[1] = htonl(offset);
char *data = tile->data + pos;
data_len = packet_sizes.at(packet_idx);
if (pos + data_len >= (unsigned int) tile->data_len) {
if (send_m) {
m = 1;
}
data_len = tile->data_len - pos;
}
pos += data_len;
if(data_len) { /* check needed for FEC_MULT */
char encrypted_data[data_len + MAX_CRYPTO_EXCEED];
if (tx->encryption) {
data_len = tx->enc_funcs->encrypt(tx->encryption,
data, data_len,
(char *) rtp_hdr_packet,
frame->fec_params.type != FEC_NONE ? sizeof(fec_payload_hdr_t) :
sizeof(video_payload_hdr_t),
encrypted_data);
data = encrypted_data;
}
if (control_stats_enabled(tx->control)) {
const time_ns_t current_time_ns =
get_time_in_ns();
if (current_time_ns - tx->last_stat_report >=
CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_NS) {
std::ostringstream oss;
oss << "tx_send " << std::hex << rtp_my_ssrc(rtp_session) << std::dec << " video " << tx->sent_since_report;
control_report_stats(tx->control, oss.str());
tx->last_stat_report = current_time_ns;
tx->sent_since_report = 0;
}
tx->sent_since_report += data_len + rtp_hdr_len;
}
rtp_send_data_hdr(rtp_session, ts, pt, m, 0, 0,
(char *) rtp_hdr_packet, rtp_hdr_len,
data, data_len, 0, 0, 0);
}
if (mult_index + 1 == tx->mult_count) {
++packet_idx;
}
if(tx->fec_scheme == FEC_MULT) {
mult_pos[mult_index] = pos;
mult_index = (mult_index + 1) % tx->mult_count;
}
rtp_hdr_packet += rtp_hdr_len / sizeof(uint32_t);
// TRAFFIC SHAPER
if (pos < (unsigned int) tile->data_len) { // wait for all but last packet
do {
GET_STOPTIME;
GET_DELTA;
} while (packet_rate - delta - overslept > 0);
overslept = -(packet_rate - delta - overslept);
//fprintf(stdout, "%ld ", overslept);
}
} while (pos < tile->data_len || mult_index != 0); // when multiplying, we need all streams go to the end
if (!tx->encryption) {
rtp_async_wait(rtp_session);
}
free(rtp_headers);
}
static void audio_tx_send_pkt(struct tx *tx, struct rtp *rtp_session,
uint32_t timestamp,
const struct audio_tx_data *buffer, int channel,
int packet, int buflen, int pktoff, bool send_m);
/*
* This multiplication scheme relies upon the fact, that our RTP/pbuf implementation is
* not sensitive to packet duplication. Otherwise, we can get into serious problems.
*/
void
audio_tx_send(struct tx *tx, struct rtp *rtp_session,
const struct audio_tx_data *buffer)
{
if (!rtp_has_receiver(rtp_session)) {
return;
}
fec_check_messages(tx);
const uint32_t timestamp =
incompatible_features && buffer->timestamp != -1
? get_local_mediatime_offset() + buffer->timestamp
: get_local_mediatime();
for (int iter = 0; iter < tx->mult_count; ++iter) {
for (int chan = 0; chan < buffer->desc.ch_count; ++chan) {
int buflen = 0;
for (int pkt = 0; pkt < buffer->channels[chan].pkt_count;
++pkt) {
buflen += buffer->channels[chan].pkts[pkt].len;
}
int pktoff = 0;
for (int pkt = 0; pkt < buffer->channels[chan].pkt_count;
++pkt) {
bool send_m =
iter == tx->mult_count - 1 &&
chan == buffer->desc.ch_count - 1 &&
pkt == buffer->channels[chan].pkt_count - 1;
audio_tx_send_pkt(tx, rtp_session, timestamp,
buffer, chan, pkt, buflen,
pktoff, send_m);
pktoff += buffer->channels[chan].pkts[pkt].len;
}
}
}
tx->buffer++;
}
/**
* @param buflen aggregate size of all audio packages to be sent
* @param pktoff offset of the audio packet inside the buffer
*/
static void
audio_tx_send_pkt(struct tx *tx, struct rtp *rtp_session, uint32_t timestamp,
const struct audio_tx_data *buffer, int channel, int packet,
int buflen, int pktoff, bool send_m)
{
const struct audio_tx_pkt *pkt = &buffer->channels[channel].pkts[packet];
int pt = fec_pt_from_fec_type(
TX_MEDIA_AUDIO, pkt->fec_desc.type,
tx->encryption); /* PT set for audio in our packet format */
unsigned m = 0U;
// see definition in rtp_callback.h
uint32_t rtp_hdr[100];
int rtp_hdr_len = 0;
int hdrs_len = (rtp_is_ipv6(rtp_session) ? 40 : 20) + 8 +
12; // MTU - IP hdr - UDP hdr - RTP hdr - payload_hdr
const unsigned int fec_symbol_size = pkt->fec_desc.symbol_size;
const char *chan_data = pkt->data;
const unsigned len = pkt->len;
unsigned pos = 0U;
if (pkt->fec_desc.type == FEC_NONE) {
hdrs_len += (sizeof(audio_payload_hdr_t));
rtp_hdr_len = sizeof(audio_payload_hdr_t);
format_audio_header(buffer->desc, channel, buflen, tx->buffer,
rtp_hdr);
} else {
hdrs_len += (sizeof(fec_payload_hdr_t));
rtp_hdr_len = sizeof(fec_payload_hdr_t);
uint32_t tmp = channel << 22;
tmp |= 0x3fffff & tx->buffer;
// see definition in rtp_callback.h
rtp_hdr[0] = htonl(tmp);
rtp_hdr[2] = htonl(buflen);
rtp_hdr[3] = htonl(pkt->fec_desc.k << 19 |
pkt->fec_desc.m << 6 |
pkt->fec_desc.c);
rtp_hdr[4] = htonl(pkt->fec_desc.seed);
}
if (tx->encryption) {
hdrs_len += sizeof(crypto_payload_hdr_t) +
tx->enc_funcs->get_overhead(tx->encryption);
rtp_hdr[rtp_hdr_len / sizeof(uint32_t)] =
htonl(DEFAULT_CIPHER_MODE << 24);
rtp_hdr_len += sizeof(crypto_payload_hdr_t);
}
if (pkt->fec_desc.type != FEC_NONE) {
check_symbol_size(fec_symbol_size, tx->mtu - hdrs_len);
}
do {
const char *data = chan_data + pos;
unsigned data_len = tx->mtu - hdrs_len;
if (pos + data_len >= len) {
data_len = len - pos;
if (send_m) {
m = 1;
}
}
rtp_hdr[1] = htonl(pktoff + pos);
pos += data_len;
char encrypted_data[data_len + MAX_CRYPTO_EXCEED];
if (tx->encryption) {
data_len = tx->enc_funcs->encrypt(
tx->encryption, const_cast<char *>(data), data_len,
(char *) rtp_hdr,
rtp_hdr_len - sizeof(crypto_payload_hdr_t),
encrypted_data);
data = encrypted_data;
}
if (control_stats_enabled(tx->control)) {
const time_ns_t current_time_ns = get_time_in_ns();
if (current_time_ns - tx->last_stat_report >=
CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_NS) {
std::ostringstream oss;
oss << "tx_send " << std::hex
<< rtp_my_ssrc(rtp_session) << std::dec
<< " audio " << tx->sent_since_report;
control_report_stats(tx->control, oss.str());
tx->last_stat_report = current_time_ns;
tx->sent_since_report = 0;
}
tx->sent_since_report += data_len + rtp_hdr_len;
}
rtp_send_data_hdr(rtp_session, timestamp, pt, m,
0, /* contributing sources */
0, /* contributing sources length */
(char *) rtp_hdr, rtp_hdr_len,
const_cast<char *>(data), data_len, 0, 0, 0);
} while (pos < len);
}
/**
* audio_tx_send_standard - Send interleaved channels from the audio_frame2,
* as the mulaw and A-law standards (dynamic or std PT).
*/
void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session,
const audio_frame2 * buffer) {
//TODO to be more abstract in order to accept A-law too and other supported standards with such implementation
assert(buffer->get_codec() == AC_MULAW || buffer->get_codec() == AC_ALAW || buffer->get_codec() == AC_OPUS);
int pt;
uint32_t ts;
static uint32_t ts_prev = 0;
// Configure the right Payload type,
// 8000 Hz, 1 channel and 2 bps is the ITU-T G.711 standard (should be 1 bps...)
// Other channels or Hz goes to DynRTP-Type97
if (buffer->get_channel_count() == 1 && buffer->get_sample_rate() == 8000) {
if (buffer->get_codec() == AC_MULAW)
pt = PT_ITU_T_G711_PCMU;
else if (buffer->get_codec() == AC_ALAW)
pt = PT_ITU_T_G711_PCMA;
else pt = PT_DynRTP_Type97;
} else {
pt = PT_DynRTP_Type97;
}
// The sizes for the different audio_frame2 channels must be the same.
for (int i = 1; i < buffer->get_channel_count(); i++)
assert(buffer->get_data_len(0) == buffer->get_data_len(i));
int data_len = buffer->get_data_len(0) * buffer->get_channel_count(); /* Number of samples to send */
int payload_size = tx->mtu - 40 - 8 - 12; /* Max size of an RTP payload field (minus IPv6, UDP and RTP header lengths) */
if (buffer->get_codec() == AC_OPUS) { // OPUS needs to fit one package
if (payload_size < data_len) {
log_msg(LOG_LEVEL_ERROR, "Transmit: Opus frame larger than packet! Discarding...\n");
return;
}
} else { // we may split the data into more packets, compute chunk size
int frame_size = buffer->get_channel_count() * buffer->get_bps();
payload_size = payload_size / frame_size * frame_size; // align to frame size
}
int pos = 0;
do {
int pkt_len = std::min(payload_size, data_len - pos);
// interleave
if (buffer->get_codec() == AC_OPUS) {
if (buffer->get_channel_count() > 1) { // we cannot interleave Opus here
LOG(LOG_LEVEL_ERROR) << "Transmit: Only Opus with 1 channel is supported in RFC-compliant mode! Discarding...\n";
return;
}
memcpy(tx->tmp_packet, buffer->get_data(0), pkt_len);
} else {
for (int ch = 0; ch < buffer->get_channel_count(); ch++) {
remux_channel(tx->tmp_packet, buffer->get_data(ch) + pos / buffer->get_channel_count(), buffer->get_bps(), pkt_len / buffer->get_channel_count(), 1, buffer->get_channel_count(), 0, ch);
}
}
// Update first sample timestamp
if (buffer->get_codec() == AC_OPUS) {
/* OPUS packet will be the whole contained in one packet
* according to RFC 7587. For PCMA/PCMU there may be more
* packets so we cannot use the whole frame duration. */
ts = get_std_audio_local_mediatime(buffer->get_duration(), 48000);
} else {
ts = get_std_audio_local_mediatime((double) pkt_len / (double) buffer->get_channel_count() / (double) buffer->get_sample_rate(), buffer->get_sample_rate());
}
rtp_send_ctrl(rtp_session, ts_prev, 0, get_time_in_ns()); //send RTCP SR
ts_prev = ts;
// Send the packet
rtp_send_data(rtp_session, ts, pt, 0, 0, /* contributing sources */
0, /* contributing sources length */
tx->tmp_packet, pkt_len, 0, 0, 0);
pos += pkt_len;
} while (pos < data_len);
}
/**
* H.264 standard transmission
*/
void tx_send_h264(struct tx *tx, struct video_frame *frame,
struct rtp *rtp_session) {
assert(frame->tile_count == 1); // std transmit doesn't handle more than one tile
assert(!frame->fragment || tx->fec_scheme == FEC_NONE); // currently no support for FEC with fragments
assert(!frame->fragment || frame->tile_count); // multiple tiles are not currently supported for fragmented send
uint32_t ts = get_std_video_local_mediatime();
struct tile *tile = &frame->tiles[0];
char pt = PT_DynRTP_Type96;
unsigned char hdr[2];
int cc = 0;
uint32_t csrc = 0;
int m = 0;
char *extn = 0;
uint16_t extn_len = 0;
uint16_t extn_type = 0;
const uint8_t *start = (uint8_t *) tile->data;
int data_len = tile->data_len;
unsigned maxPacketSize = tx->mtu - 40;
const unsigned char *endptr = 0;
const unsigned char *nal = start;
while ((nal = rtpenc_h264_get_next_nal(nal, data_len - (nal - start), &endptr))) {
unsigned int nalsize = endptr - nal;
bool eof = endptr == start + data_len;
bool lastNALUnitFragment = false; // by default
unsigned curNALOffset = 0;
char *nalc = const_cast<char *>(reinterpret_cast<const char *>(nal));
while(!lastNALUnitFragment){
// We have NAL unit data in the buffer. There are three cases to consider:
// 1. There is a new NAL unit in the buffer, and it's small enough to deliver
// to the RTP sink (as is).
// 2. There is a new NAL unit in the buffer, but it's too large to deliver to
// the RTP sink in its entirety. Deliver the first fragment of this data,
// as a FU packet, with one extra preceding header byte (for the "FU header").
// 3. There is a NAL unit in the buffer, and we've already delivered some
// fragment(s) of this. Deliver the next fragment of this data,
// as a FU packet, with two (H.264) extra preceding header bytes
// (for the "NAL header" and the "FU header").
if (curNALOffset == 0) { // case 1 or 2
if (nalsize <= maxPacketSize) { // case 1
if (eof) m = 1;
if (rtp_send_data(rtp_session, ts, pt, m, cc, &csrc,
nalc, nalsize,
extn, extn_len, extn_type) < 0) {
error_msg("There was a problem sending the RTP packet\n");
}
lastNALUnitFragment = true;
} else { // case 2
// We need to send the NAL unit data as FU packets. Deliver the first
// packet now. Note that we add "NAL header" and "FU header" bytes to the front
// of the packet (overwriting the existing "NAL header").
hdr[0] = (nal[0] & 0xE0) | 28; //FU indicator
hdr[1] = 0x80 | (nal[0] & 0x1F); // FU header (with S bit)
if (rtp_send_data_hdr(rtp_session, ts, pt, m, cc, &csrc,
(char *) hdr, 2,
nalc + 1, maxPacketSize - 2,
extn, extn_len, extn_type) < 0) {
error_msg("There was a problem sending the RTP packet\n");
}
curNALOffset += maxPacketSize - 1;
lastNALUnitFragment = false;
nalsize -= maxPacketSize - 1;
}
} else { // case 3
// We are sending this NAL unit data as FU packets. We've already sent the
// first packet (fragment). Now, send the next fragment. Note that we add
// "NAL header" and "FU header" bytes to the front. (We reuse these bytes that
// we already sent for the first fragment, but clear the S bit, and add the E
// bit if this is the last fragment.)
hdr[1] = hdr[1] & ~0x80;// FU header (no S bit)
if (nalsize + 1 > maxPacketSize) {
// We can't send all of the remaining data this time:
if (rtp_send_data_hdr(rtp_session, ts, pt, m, cc, &csrc,
(char *) hdr, 2,
nalc + curNALOffset,
maxPacketSize - 2, extn, extn_len,
extn_type) < 0) {
error_msg("There was a problem sending the RTP packet\n");
}
curNALOffset += maxPacketSize - 2;
lastNALUnitFragment = false;
nalsize -= maxPacketSize - 2;
} else {
// This is the last fragment:
if (eof) m = 1;
hdr[1] |= 0x40;// set the E bit in the FU header
if (rtp_send_data_hdr(rtp_session, ts, pt, m, cc, &csrc,
(char *) hdr, 2,
nalc + curNALOffset,
nalsize, extn, extn_len, extn_type) < 0) {
error_msg("There was a problem sending the RTP packet\n");
}
lastNALUnitFragment = true;
}
}
}
}
if (endptr != start + data_len) {
error_msg("No NAL found!\n");
}
}
void tx_send_jpeg(struct tx *tx, struct video_frame *frame,
struct rtp *rtp_session) {
uint32_t ts = 0;
assert(frame->tile_count == 1); // std transmit doesn't handle more than one tile
assert(!frame->fragment || tx->fec_scheme == FEC_NONE); // currently no support for FEC with fragments
assert(!frame->fragment || frame->tile_count); // multiple tiles are not currently supported for fragmented send
ts = get_std_video_local_mediatime();
struct tile *tile = &frame->tiles[0];
char pt = PT_JPEG;
struct jpeg_rtp_data d;
if (!jpeg_get_rtp_hdr_data((uint8_t *) frame->tiles[0].data, frame->tiles[0].data_len, &d)) {
exit_uv(1);
return;
}
uint32_t jpeg_hdr[2 /* JPEG hdr */ + 1 /* RM hdr */ + 129 /* QT hdr */];
int hdr_off = 0;
unsigned int type_spec = 0u;
jpeg_hdr[hdr_off++] = htonl(type_spec << 24u);
jpeg_hdr[hdr_off++] = htonl(d.type << 24u | d.q << 16u | d.width / 8u << 8u | d.height / 8u);
if (d.restart_interval != 0) {
// we do not align restart interval on packet boundaries yet
jpeg_hdr[hdr_off++] = htonl(d.restart_interval << 16u | 1u << 15u | 1u << 14u | 0x3fffu);
}
// quantization headers
if (d.q == 255u) { // we must include the tables
unsigned int mbz = 0u; // must be zero
unsigned int precision = 0u;
unsigned int qt_len = 2 * 64u;
jpeg_hdr[hdr_off++] = htonl(mbz << 24u | precision << 16u | qt_len);
memcpy(&jpeg_hdr[hdr_off], d.quantization_tables[0], 64);
hdr_off += 64 / sizeof(uint32_t);
memcpy(&jpeg_hdr[hdr_off], d.quantization_tables[1], 64);
hdr_off += 64 / sizeof(uint32_t);
}
char *data = (char *) d.data;
int bytes_left = tile->data_len - ((char *) d.data - tile->data);
int max_mtu = tx->mtu - ((rtp_is_ipv6(rtp_session) ? 40 : 20) + 8 + 12); // IP hdr size + UDP hdr size + RTP hdr size
int fragment_offset = 0;
do {
int hdr_len;
if (fragment_offset == 0) { // include quantization header only in 1st pkt
hdr_len = hdr_off * sizeof(uint32_t);
} else {
hdr_len = 8 + (d.restart_interval > 0 ? 4 : 0);
}
int data_len = max_mtu - hdr_len;
int m = 0;
if (bytes_left <= data_len) {
data_len = bytes_left;
m = 1;
}
jpeg_hdr[0] = htonl(type_spec << 24u | fragment_offset);
int ret = rtp_send_data_hdr(rtp_session, ts, pt, m, 0, 0,
(char *) &jpeg_hdr, hdr_len,
data, data_len, 0, 0, 0);
if (ret < 0) {
log_msg(LOG_LEVEL_ERROR, "Error sending RTP/JPEG packet!\n");
}
data += data_len;
bytes_left -= data_len;
fragment_offset += data_len;
} while (bytes_left > 0);
}
int tx_get_buffer_id(struct tx *tx)
{
return tx->buffer;
}
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