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RTC: Support keep original seq

Browse Source
pull/1830/head
winlin 5 years ago
parent 29661802b6
commit c30565b4c0
7 changed files with 64 additions and 22 deletions
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4
trunk/conf/full.conf
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@ -491,6 +491,10 @@ vhost rtc.vhost.srs.com {
# The strick check when process stun.
# default: off
stun_strict_check on;
# Whether keep original sequence number.
# If off, we will regenerate the sequence number for RTP packet.
# default: off
keep_sequence off;
}
# whether enable min delay mode for vhost.
# default: on, for RTC.

21
trunk/src/app/srs_app_config.cpp
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@ -3937,7 +3937,8 @@ srs_error_t SrsConfig::check_normal_config()
} else if (n == "rtc") {
for (int j = 0; j < (int)conf->directives.size(); j++) {
string m = conf->at(j)->name;
if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check") {
if (m != "enabled" && m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check"
&& m != "keep_sequence") {
return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
}
}
@ -5020,6 +5021,24 @@ bool SrsConfig::get_rtc_stun_strict_check(string vhost)
return SRS_CONF_PERFER_FALSE(conf->arg0());
}
bool SrsConfig::get_rtc_keep_sequence(string vhost)
{
static bool DEFAULT = false;
SrsConfDirective* conf = get_rtc(vhost);
if (!conf) {
return DEFAULT;
}
conf = conf->get("keep_sequence");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}
return SRS_CONF_PERFER_FALSE(conf->arg0());
}
bool SrsConfig::get_rtc_nack_enabled(string vhost)
{
static bool DEFAULT = true;

1
trunk/src/app/srs_app_config.hpp
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@ -544,6 +544,7 @@ public:
bool get_rtc_aac_discard(std::string vhost);
srs_utime_t get_rtc_stun_timeout(std::string vhost);
bool get_rtc_stun_strict_check(std::string vhost);
bool get_rtc_keep_sequence(std::string vhost);
bool get_rtc_nack_enabled(std::string vhost);
bool get_rtc_twcc_enabled(std::string vhost);

14
trunk/src/app/srs_app_rtc_conn.cpp
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@ -505,6 +505,7 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, int parent_cid)
nn_simulate_nack_drop = 0;
nack_enabled_ = false;
keep_sequence_ = false;
_srs_config->subscribe(this);
}
@ -530,8 +531,9 @@ srs_error_t SrsRtcPlayer::initialize(const uint32_t& vssrc, const uint32_t& assr
// TODO: FIXME: Support reload.
nack_enabled_ = _srs_config->get_rtc_nack_enabled(session_->req->vhost);
srs_trace("RTC publisher video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), nack=%d",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, nack_enabled_);
keep_sequence_ = _srs_config->get_rtc_keep_sequence(session_->req->vhost);
srs_trace("RTC publisher video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), nack=%d, keep-seq=%d",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, nack_enabled_, keep_sequence_);
if (_srs_rtc_hijacker) {
if ((err = _srs_rtc_hijacker->on_start_play(session_, this, session_->req)) != srs_success) {
@ -708,7 +710,9 @@ srs_error_t SrsRtcPlayer::send_packets(SrsRtcSource* source, const vector<SrsRtp
if (pkt->is_audio()) {
info.nn_audios++;
pkt->header.set_sequence(audio_sequence++);
if (!keep_sequence_) {
pkt->header.set_sequence(audio_sequence++);
}
pkt->header.set_ssrc(audio_ssrc);
pkt->header.set_payload_type(audio_payload_type);
@ -721,7 +725,9 @@ srs_error_t SrsRtcPlayer::send_packets(SrsRtcSource* source, const vector<SrsRtp
info.nn_videos++;
// For video, we should set the RTP packet informations about this consumer.
pkt->header.set_sequence(video_sequence++);
if (!keep_sequence_) {
pkt->header.set_sequence(video_sequence++);
}
pkt->header.set_ssrc(video_ssrc);
pkt->header.set_payload_type(video_payload_type);
}

2
trunk/src/app/srs_app_rtc_conn.hpp
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@ -213,6 +213,8 @@ private:
bool realtime;
// Whether enabled nack.
bool nack_enabled_;
// Whether keep original sequence number.
bool keep_sequence_;
public:
SrsRtcPlayer(SrsRtcSession* s, int parent_cid);
virtual ~SrsRtcPlayer();

42
trunk/src/app/srs_app_rtc_source.cpp
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@ -425,6 +425,8 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
merge_nalus = false;
meta = new SrsMetaCache();
audio_timestamp = 0;
audio_sequence = 0;
video_sequence = 0;
}
SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
@ -593,6 +595,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_opus(char* data, int size, SrsRtpPack
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeAudio;
pkt->header.set_marker(true);
pkt->header.set_sequence(audio_sequence++);
pkt->header.set_timestamp(audio_timestamp);
// TODO: FIXME: Why 960? Need Refactoring?
@ -654,25 +657,25 @@ srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
if ((err = package_nalus(msg, samples, pkts)) != srs_success) {
return srs_error_wrap(err, "package nalus as one");
}
}
// By default, we package each NALU(sample) to a RTP or FUA packet.
for (int i = 0; i < nn_samples; i++) {
SrsSample* sample = samples[i];
// We always ignore bframe here, if config to discard bframe,
// the bframe flag will not be set.
if (sample->bframe) {
continue;
}
} else {
// By default, we package each NALU(sample) to a RTP or FUA packet.
for (int i = 0; i < nn_samples; i++) {
SrsSample* sample = samples[i];
if (sample->size <= kRtpMaxPayloadSize) {
if ((err = package_single_nalu(msg, sample, pkts)) != srs_success) {
return srs_error_wrap(err, "package single nalu");
// We always ignore bframe here, if config to discard bframe,
// the bframe flag will not be set.
if (sample->bframe) {
continue;
}
} else {
if ((err = package_fu_a(msg, sample, kRtpMaxPayloadSize, pkts)) != srs_success) {
return srs_error_wrap(err, "package fu-a");
if (sample->size <= kRtpMaxPayloadSize) {
if ((err = package_single_nalu(msg, sample, pkts)) != srs_success) {
return srs_error_wrap(err, "package single nalu");
}
} else {
if ((err = package_fu_a(msg, sample, kRtpMaxPayloadSize, pkts)) != srs_success) {
return srs_error_wrap(err, "package fu-a");
}
}
}
}
@ -733,6 +736,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcSource* source, SrsShare
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_marker(false);
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
SrsRtpSTAPPayload* stap = new SrsRtpSTAPPayload();
@ -802,6 +806,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
// Package NALUs in a single RTP packet.
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
pkt->payload = raw;
pkt->shared_msg = msg->copy();
@ -831,6 +836,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
fua->nri = (SrsAvcNaluType)header;
@ -856,6 +862,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg,
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
SrsRtpRawPayload* raw = new SrsRtpRawPayload();
@ -885,6 +892,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSam
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
SrsRtpFUAPayload2* fua = new SrsRtpFUAPayload2();

2
trunk/src/app/srs_app_rtc_source.hpp
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@ -174,6 +174,8 @@ private:
bool discard_bframe;
bool merge_nalus;
uint32_t audio_timestamp;
uint16_t audio_sequence;
uint16_t video_sequence;
public:
SrsRtcFromRtmpBridger(SrsRtcSource* source);
virtual ~SrsRtcFromRtmpBridger();

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