Rtcp sr and rr for rtc play #3748

runner365 · 2023-07-30T05:05:10Z

For WebRTC downlink, add:

Sending RTCP SR.
Receiving and processing RTCP RR to calculate RTT.
Retransmission of NACK based on RTT.

The main functionality is to calculate RTT in the downlink, which prevents unnecessary retransmissions within a time period less than RTT, avoiding unnecessary retransmissions.

TRANS_BY_GPT3

winlinvip · 2023-08-03T13:01:17Z

trunk/src/app/srs_app_rtc_conn.cpp

@@ -448,6 +450,9 @@ SrsRtcPlayStream::~SrsRtcPlayStream()

    _srs_config->unsubscribe(this);

+    if (timer_rtcp_) {
+        srs_freep(timer_rtcp_);


You can simply release the object directly, srs_freep will check if it is empty.

srs_freep(timer_rtcp_);

TRANS_BY_GPT3

winlinvip · 2023-08-03T13:02:39Z

trunk/src/app/srs_app_rtc_conn.cpp

@@ -681,6 +686,29 @@ srs_error_t SrsRtcPlayStream::cycle()
    }
 }

+srs_error_t SrsRtcPlayStream::send_rtcp_sr(int64_t now_ms) {


Please keep the coding style consistent, and the opening brace of member functions should be on a new line.

srs_error_t SrsRtcPlayStream::send_rtcp_sr(int64_t now_ms) {

TRANS_BY_GPT3

winlinvip · 2023-08-03T13:04:41Z

trunk/src/app/srs_app_rtc_conn.cpp

+    srs_error_t err = srs_success;
+    for(std::map<uint32_t, SrsRtcVideoSendTrack*>::iterator iter = video_tracks_.begin();
+        iter != video_tracks_.end();
+        iter++) {


These three lines can be written in one line, now the screen and editor are both very long, and can tolerate longer lines. Please change it to:

for(std::map<uint32_t, SrsRtcVideoSendTrack*>::iterator iter = video_tracks_.begin(); iter != video_tracks_.end(); iter++) {

TRANS_BY_GPT3

trunk/src/app/srs_app_rtc_conn.cpp

winlinvip · 2023-08-03T13:11:31Z

trunk/src/app/srs_app_rtc_source.cpp

+    sr->set_rtp_send_packets(send_count_);
+    sr->set_rtp_send_bytes(send_bytes_);
+
+    char data[1500];


There should not be a magic number 1500, but instead use existing constants or macro definitions. You can refer to the serialization of other RTC objects, I remember there will be a buffer cache and a method to specify the corresponding length.

TRANS_BY_GPT3

winlinvip · 2023-08-03T13:13:01Z

trunk/src/app/srs_app_rtc_source.cpp

+    char data[1500];
+    SrsBuffer buffer(data, sr->nb_bytes());
+    sr->encode(&buffer);
+    delete sr;


SrsAutoFree should be used instead of naked delete to automatically release memory upon creation, in order to avoid forgetting to delete sr and causing memory leaks.

SrsRtcpSR* sr = new SrsRtcpSR(); SrsAutoFree(SrsRtcpSR, sr);

TRANS_BY_GPT3

winlinvip · 2023-08-03T13:14:07Z

trunk/src/app/srs_app_rtc_source.cpp

+    send_count_++;
+    send_bytes_ += len;
+    last_rtp_pkt_ts_ = rtp_ts;
+    last_rtp_ms_     = srs_update_system_time() / 1000;//ms


Do not directly use int64_t and ms, instead use srs_utime_t and srsu2ms conversion to avoid unit errors and make it easier to understand.

TRANS_BY_GPT3

xiaozhihong · 2023-08-03T14:57:56Z

trunk/src/app/srs_app_rtc_conn.cpp

+                return video_iter->second->handle_rtcp_rr(rb, now_ms);
+            }
+        }
+        srs_warn("rtcp rr find to find track by ssrc:%u", ssrc);


xiaozhihong · 2023-08-03T15:07:57Z

trunk/src/app/srs_app_rtc_source.hpp

+    void update_rtp_static(int64_t len, uint32_t rtp_ts);
+public:
+    srs_error_t handle_rtcp_rr(const SrsRtcpRB& rb, int64_t now_ms);
+protected:


Directly assigning values to member variables in a C++ class seems to be a syntax introduced in C++11.

xiaozhihong · 2023-08-03T15:14:31Z

trunk/src/app/srs_app_rtc_source.cpp

+            interval = (interval > 10) ? (interval - 10) : interval;//for resend interval Residual
+            if (diff_t < (int64_t)interval) {
+                return NULL;
+            }


Should limit the count of retrans packet?

winlinvip · 2023-08-11T03:04:00Z

trunk/src/app/srs_app_rtc_conn.cpp

+        return err;
+    }
+
+    srs_utime_t now_ms = srs_update_system_time() / 1000;


You cannot divide by 1000 here, because this function returns srs_utime_t, and the variable cannot have a unit, because this type already has a unit.

It should be changed to:

srs_utime_t now = srs_update_system_time(); if ((err = p_->send_rtcp_sr(now)) != srs_success) {

Your send_rtcp_sr should not be in milliseconds or seconds, it should be srs_utime_t, because it has a unit, which is a time unit. Don't convert it to milliseconds or seconds, only convert it to milliseconds using srsu2ms when it is finally used.

TRANS_BY_GPT4

winlinvip · 2023-08-11T03:21:17Z

trunk/src/app/srs_app_rtc_source.cpp

@@ -2710,6 +2743,70 @@ void SrsRtcSendTrack::rebuild_packet(SrsRtpPacket* pkt)
    srs_info("RTC: Correct %s seq=%u/%u, ts=%u/%u", track_desc_->type_.c_str(), seq, pkt->header.get_sequence(), ts, pkt->header.get_timestamp());
 }

+srs_error_t SrsRtcSendTrack::send_rtcp_sr(srs_utime_t now_ms) {


This srs_utime_t is utime, which is in microseconds (us), so the variable should not carry the unit, or the variable should be now_us. However, this type is already in microseconds, so there is no need to carry the us unit. It should be changed to:

srs_error_t SrsRtcSendTrack::send_rtcp_sr(srs_utime_t now) { srs_utime_t diff = now_ms - last_rtp_ms_; srs_utime_t diff_ts = diff * track_desc_->media_->sample_ / 1000; srs_utime_t video_rtp_ts = last_rtp_pkt_ts_ + diff_xxx;

You should convert track_desc_->media_->sample_ to microseconds (us), so it can be directly added or subtracted with srs_utime_t.

TRANS_BY_GPT4

winlinvip added the EnglishNative This issue is conveyed exclusively in English. label Jul 30, 2023

runner365 force-pushed the rtcp_update branch 2 times, most recently from 51c4d56 to a627c54 Compare August 1, 2023 13:52

winlinvip added the TransByAI Translated by AI/GPT. label Aug 2, 2023

runner365 requested review from winlinvip, lipeng19811218 and xiaozhihong August 3, 2023 00:54

runner365 added 8 commits August 3, 2023 19:48

add rtcp sr for play stream

3c8c708

add send rtcp sr and receive rtcp rr

835c7a6

remove c++11 code style

5de6330

update for check rtc and resend count in nack requires

1b99abc

update for srs_update_system_time recording lsr

e749fb4

update

00ff60f

udpate for no c++11

de6622b

update for utest

07345e8

runner365 force-pushed the rtcp_update branch from a627c54 to 07345e8 Compare August 3, 2023 11:48

winlinvip reviewed Aug 3, 2023

View reviewed changes

trunk/src/app/srs_app_rtc_conn.cpp Show resolved Hide resolved

winlinvip reviewed Aug 3, 2023

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xiaozhihong reviewed Aug 3, 2023

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runner365 added 4 commits August 8, 2023 22:37

update for pr review

db122a7

update for pr review

feabe4b

update for typo

d3b51b5

update typo

fd689cd

winlinvip reviewed Aug 11, 2023

View reviewed changes

winlinvip force-pushed the develop branch from d2de910 to cf46dae Compare August 28, 2023 02:53

xiaozhihong assigned runner365 Nov 23, 2023

winlinvip force-pushed the develop branch 2 times, most recently from 0bb9637 to 2e211f6 Compare August 12, 2024 10:42

winlinvip force-pushed the develop branch from e52ad4b to d70e735 Compare September 1, 2024 09:04

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Rtcp sr and rr for rtc play #3748

Rtcp sr and rr for rtc play #3748

runner365 commented Jul 30, 2023 •

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Rtcp sr and rr for rtc play #3748

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