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Code Issues Releases Wiki Activity

Merge branch '4.0release' into develop

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This commit is contained in:
winlin 2022-09-03 18:13:11 +08:00
commit 79358673ef
6 changed files with 120 additions and 4 deletions
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1
trunk/doc/CHANGELOG.md
View file

@ -63,6 +63,7 @@ The changelog for SRS.
## SRS 4.0 Changelog
* v4.0, 2022-09-03, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Play stucked when republish. v4.0.260
* v4.0, 2022-09-02, For [#307](https://github.com/ossrs/srs/issues/307): WebRTC: Support use domain name as CANDIDATE. v4.0.259
* v4.0, 2022-08-29, Copy libxml2-dev for FFmpeg. v4.0.258
* v4.0, 2022-08-24, STAT: Support config server_id and generate one if empty. v4.0.257

15
trunk/src/app/srs_app_rtc_conn.cpp
View file

@ -517,10 +517,12 @@ srs_error_t SrsRtcPlayStream::initialize(SrsRequest* req, std::map<uint32_t, Srs
void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
{
if (!desc) return;
// Refresh the relation for audio.
// TODO: FIXME: Match by label?
if (desc && desc->audio_track_desc_ && audio_tracks_.size() == 1) {
if (! audio_tracks_.empty()) {
if (!audio_tracks_.empty()) {
uint32_t ssrc = desc->audio_track_desc_->ssrc_;
SrsRtcAudioSendTrack* track = audio_tracks_.begin()->second;
@ -532,7 +534,7 @@ void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
// Refresh the relation for video.
// TODO: FIMXE: Match by label?
if (desc && desc->video_track_descs_.size() == 1) {
if (! video_tracks_.empty()) {
if (!video_tracks_.empty()) {
SrsRtcTrackDescription* vdesc = desc->video_track_descs_.at(0);
uint32_t ssrc = vdesc->ssrc_;
SrsRtcVideoSendTrack* track = video_tracks_.begin()->second;
@ -541,6 +543,15 @@ void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
video_tracks_.insert(make_pair(ssrc, track));
}
}
// Request keyframe(PLI) when stream changed.
if (desc->audio_track_desc_) {
pli_worker_->request_keyframe(desc->audio_track_desc_->ssrc_, cid_);
}
for (vector<SrsRtcTrackDescription*>::iterator it = desc->video_track_descs_.begin(); it != desc->video_track_descs_.end(); ++it) {
SrsRtcTrackDescription* vdesc = *it;
pli_worker_->request_keyframe(vdesc->ssrc_, cid_);
}
}
srs_error_t SrsRtcPlayStream::on_reload_vhost_play(string vhost)

71
trunk/src/app/srs_app_rtc_source.cpp
View file

@ -2552,12 +2552,53 @@ srs_error_t SrsRtcVideoRecvTrack::check_send_nacks()
return err;
}
SrsRtcJitter::SrsRtcJitter(uint32_t base)
{
pkt_base_ = pkt_last_ = 0;
correct_last_ = correct_base_ = 0;
base_ = base;
init_ = false;
}
SrsRtcJitter::~SrsRtcJitter()
{
}
uint32_t SrsRtcJitter::correct(uint32_t ts)
{
if (!init_) {
init_ = true;
correct_base_ = base_;
srs_trace("RTC: Jitter init base=%u, ts=%u", base_, ts);
}
if (!pkt_base_) pkt_base_ = ts;
if (pkt_last_) {
int32_t distance = srs_rtp_ts_distance(ts, pkt_last_);
static int32_t max_deviation = 90 * 3 * 1000;
if (distance > max_deviation || distance < -1 * max_deviation) {
srs_trace("RTC: Jitter rebase ts=%u, last=%u, distance=%d, pkt-base=%u/%u, correct-base=%u/%u", ts, pkt_last_, distance, pkt_base_, ts, correct_base_, correct_last_);
pkt_base_ = ts;
correct_base_ = correct_last_;
}
}
pkt_last_ = ts;
correct_last_ = correct_base_ + ts - pkt_base_;
return correct_last_;
}
SrsRtcSendTrack::SrsRtcSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc, bool is_audio)
{
session_ = session;
track_desc_ = track_desc->copy();
nack_no_copy_ = false;
seqno_ = 0;
init_ = false;
// Make a different start of sequence number, for debugging.
jitter_ = new SrsRtcJitter(track_desc_->type_ == "audio" ? 10000 : 20000);
if (is_audio) {
rtp_queue_ = new SrsRtpRingBuffer(100);
} else {
@ -2572,6 +2613,7 @@ SrsRtcSendTrack::~SrsRtcSendTrack()
srs_freep(rtp_queue_);
srs_freep(track_desc_);
srs_freep(nack_epp);
srs_freep(jitter_);
}
bool SrsRtcSendTrack::has_ssrc(uint32_t ssrc)
@ -2622,6 +2664,23 @@ std::string SrsRtcSendTrack::get_track_id()
return track_desc_->id_;
}
void SrsRtcSendTrack::rebuild_packet(SrsRtpPacket* pkt)
{
// Rebuild the sequence number.
if (!init_) {
init_ = true;
// Make a different start of sequence number, for debugging.
seqno_ = track_desc_->type_ == "audio" ? 1000 : 2000;
srs_trace("RTC: Seqno rebuild %s track=%s, ssrc=%d, seqno=%d to %d", track_desc_->type_.c_str(), track_desc_->id_.c_str(),
pkt->header.get_ssrc(), pkt->header.get_sequence(), seqno_);
}
pkt->header.set_sequence(seqno_++);
// Rebuild the timestamp.
uint32_t ts = pkt->header.get_timestamp();
pkt->header.set_timestamp(jitter_->correct(ts));
}
srs_error_t SrsRtcSendTrack::on_nack(SrsRtpPacket** ppkt)
{
srs_error_t err = srs_success;
@ -2699,10 +2758,16 @@ srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket* pkt)
// TODO: FIXME: Should update PT for RTX.
}
// Rebuild the sequence number and timestamp of packet, see https://github.com/ossrs/srs/issues/3167
rebuild_packet(pkt);
if ((err = session_->do_send_packet(pkt)) != srs_success) {
return srs_error_wrap(err, "raw send");
}
srs_info("RTC: Send audio ssrc=%d, seqno=%d, keyframe=%d, ts=%u", pkt->header.get_ssrc(),
pkt->header.get_sequence(), pkt->is_keyframe(), pkt->header.get_timestamp());
return err;
}
@ -2743,10 +2808,16 @@ srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket* pkt)
// TODO: FIXME: Should update PT for RTX.
}
// Rebuild the sequence number and timestamp of packet, see https://github.com/ossrs/srs/issues/3167
rebuild_packet(pkt);
if ((err = session_->do_send_packet(pkt)) != srs_success) {
return srs_error_wrap(err, "raw send");
}
srs_info("RTC: Send video ssrc=%d, seqno=%d, keyframe=%d, ts=%u", pkt->header.get_ssrc(),
pkt->header.get_sequence(), pkt->is_keyframe(), pkt->header.get_timestamp());
return err;
}

31
trunk/src/app/srs_app_rtc_source.hpp
View file

@ -579,9 +579,29 @@ public:
virtual srs_error_t check_send_nacks();
};
class SrsRtcJitter
{
private:
// The ts about packet.
uint32_t pkt_base_;
uint32_t pkt_last_;
// The ts after corrected.
uint32_t correct_base_;
uint32_t correct_last_;
// The base timestamp by config, start from it.
uint32_t base_;
// Whether initialized. Note that we should not use correct_base_(0) as init state, because it might flip back.
bool init_;
public:
SrsRtcJitter(uint32_t base);
virtual ~SrsRtcJitter();
public:
uint32_t correct(uint32_t ts);
};
class SrsRtcSendTrack
{
protected:
public:
// send track description
SrsRtcTrackDescription* track_desc_;
protected:
@ -589,6 +609,13 @@ protected:
SrsRtcConnection* session_;
// NACK ARQ ring buffer.
SrsRtpRingBuffer* rtp_queue_;
protected:
// Current sequence number.
uint64_t seqno_;
// Whether initialized. Note that we should not use seqno_(0) as init state, because it might flip back.
bool init_;
// The jitter to correct ts.
SrsRtcJitter* jitter_;
private:
// By config, whether no copy.
bool nack_no_copy_;
@ -605,6 +632,8 @@ public:
bool set_track_status(bool active);
bool get_track_status();
std::string get_track_id();
protected:
void rebuild_packet(SrsRtpPacket* pkt);
public:
// Note that we can set the pkt to NULL to avoid copy, for example, if the NACK cache the pkt and
// set to NULL, nack nerver copy it but set the pkt to NULL.

2
trunk/src/core/srs_core_version4.hpp
View file

@ -9,6 +9,6 @@
#define VERSION_MAJOR 4
#define VERSION_MINOR 0
#define VERSION_REVISION 259
#define VERSION_REVISION 260
#endif

4
trunk/src/kernel/srs_kernel_rtc_rtp.hpp
View file

@ -61,6 +61,10 @@ inline int16_t srs_rtp_seq_distance(const uint16_t& prev_value, const uint16_t&
{
return (int16_t)(value - prev_value);
}
inline int32_t srs_rtp_ts_distance(const uint32_t& prev_value, const uint32_t& value)
{
return (int32_t)(value - prev_value);
}
// For map to compare the sequence of RTP.
struct SrsSeqCompareLess {