revert,sync

xiaozhihong · Jan 3, 2024 · a122756 · a122756
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commit a122756
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diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
@@ -9,7 +9,6 @@ The changelog for SRS.
 ## SRS 6.0 Changelog
 * v6.0, 2023-12-30, Merge [#3916](https://github.com/ossrs/srs/pull/3916): Enhancing the compatibility of options.sh. v6.0.108 (#3916)
 * v6.0, 2023-12-30, Merge [#3914](https://github.com/ossrs/srs/pull/3914): Forward: when unpublish crash caused by uninitialized forward connection. v6.0.107 (#3914)
-* v6.0, 2023-12-29, Merge [#3911](https://github.com/ossrs/srs/pull/3911): WHIP: Fix bug for converting WHIP to RTMP/HLS. v6.0.107 (#3911)
 * v6.0, 2023-12-15, Merge [#3854](https://github.com/ossrs/srs/pull/3854): Typo: line 263 - srs_app_srt_conn.cpp. v6.0.106 (#3854)
 * v6.0, 2023-12-14, Merge [#3910](https://github.com/ossrs/srs/pull/3910): RTC: Support OPUS stereo SDP option. v6.0.105 (#3910)
 * v6.0, 2023-12-14, Merge [#3902](https://github.com/ossrs/srs/pull/3902): Security: Support IP whitelist for HTTP-FLV, HLS, WebRTC, and SRT. v6.0.104 (#3902)
@@ -122,7 +121,6 @@ The changelog for SRS.
 
 ## SRS 5.0 Changelog
 * v5.0, 2023-12-30, Merge [#3916](https://github.com/ossrs/srs/pull/3916): Enhancing the compatibility of options.sh. v5.0.204 (#3916)
-* v5.0, 2023-12-29, Merge [#3911](https://github.com/ossrs/srs/pull/3911): WHIP: Fix bug for converting WHIP to RTMP/HLS. v5.0.204 (#3911)
 * v5.0, 2023-12-14, Merge [#3910](https://github.com/ossrs/srs/pull/3910): RTC: Support OPUS stereo SDP option. v5.0.203 (#3910)
 * v5.0, 2023-12-14, Merge [#3902](https://github.com/ossrs/srs/pull/3902): Security: Support IP whitelist for HTTP-FLV, HLS, WebRTC, and SRT. v5.0.202 (#3902)
 * v5.0, 2023-11-22, Merge [#3891](https://github.com/ossrs/srs/pull/3891): fix 'sed' error in options.sh. v5.0.201 (#3891)

diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
@@ -1318,16 +1318,12 @@ SrsRtcFrameBuilder::SrsRtcFrameBuilder(ISrsStreamBridge* bridge)
     header_sn_ = 0;
     memset(cache_video_pkts_, 0, sizeof(cache_video_pkts_));
     rtp_key_frame_ts_ = -1;
-    sync_state_ = -1;
-    obs_whip_sps_ = obs_whip_pps_ = NULL;
 }
 
 SrsRtcFrameBuilder::~SrsRtcFrameBuilder()
 {
     srs_freep(codec_);
     clear_cached_video();
-    srs_freep(obs_whip_sps_);
-    srs_freep(obs_whip_pps_);
 }
 
 srs_error_t SrsRtcFrameBuilder::initialize(SrsRequest* r)
@@ -1370,18 +1366,8 @@ srs_error_t SrsRtcFrameBuilder::on_rtp(SrsRtpPacket *pkt)
 
     // Have no received any sender report, can't calculate avsync_time,
     // discard it to avoid timestamp problem in live source
-    const SrsRtpHeader& h = pkt->header;
     if (pkt->get_avsync_time() <= 0) {
-        if (sync_state_ < 0) {
-            srs_trace("RTC: Discard no-sync %s, ssrc=%u, seq=%u, ts=%u, state=%d", pkt->is_audio() ? "Audio" : "Video",
-                h.get_ssrc(), h.get_sequence(), h.get_timestamp(), sync_state_);
-            sync_state_ = 0;
-        }
         return err;
-    } else if (sync_state_ < 1) {
-        srs_trace("RTC: Accept sync %s, ssrc=%u, seq=%u, ts=%u, state=%d", pkt->is_audio() ? "Audio" : "Video",
-            h.get_ssrc(), h.get_sequence(), h.get_timestamp(), sync_state_);
-        sync_state_ = 2;
     }
 
     if (pkt->is_audio()) {
@@ -1513,71 +1499,46 @@ srs_error_t SrsRtcFrameBuilder::packet_video_key_frame(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
-    // For OBS WHIP, it uses RTP Raw packet with SPS/PPS/IDR frame.
-    SrsRtpRawPayload* raw_payload = dynamic_cast<SrsRtpRawPayload*>(pkt->payload());
-    if (raw_payload) {
-        if (pkt->nalu_type == SrsAvcNaluTypeSPS) {
-            srs_freep(obs_whip_sps_);
-            obs_whip_sps_ = pkt->copy();
-        } else if (pkt->nalu_type == SrsAvcNaluTypePPS) {
-            srs_freep(obs_whip_pps_);
-            obs_whip_pps_ = pkt->copy();
-        }
-        // Ignore if one of OBS WHIP SPS/PPS is not ready.
-        if (!obs_whip_pps_ || !obs_whip_pps_) {
-            return err;
-        }
-    }
-
-    // Generally, there will be SPS+PPS+IDR in a STAP-A packet.
+    // TODO: handle sps and pps in 2 rtp packets
     SrsRtpSTAPPayload* stap_payload = dynamic_cast<SrsRtpSTAPPayload*>(pkt->payload());
-
-    // Handle SPS/PPS in cache or STAP-A packet.
-    if (stap_payload || raw_payload) {
-        // Get the SPS/PPS from cache or STAP-A packet.
-        SrsSample* sps = stap_payload ? stap_payload->get_sps() : NULL;
-        if (!sps && obs_whip_sps_) sps = dynamic_cast<SrsRtpRawPayload*>(obs_whip_sps_->payload())->sample_;
-        SrsSample* pps = stap_payload ? stap_payload->get_pps() : NULL;
-        if (!pps && obs_whip_pps_) pps = dynamic_cast<SrsRtpRawPayload*>(obs_whip_pps_->payload())->sample_;
-        if (!sps || !pps) {
+    if (stap_payload) {
+        SrsSample* sps = stap_payload->get_sps();
+        SrsSample* pps = stap_payload->get_pps();
+        if (NULL == sps || NULL == pps) {
             return srs_error_new(ERROR_RTC_RTP_MUXER, "no sps or pps in stap-a rtp. sps: %p, pps:%p", sps, pps);
-        }
-
-        // Reset SPS/PPS cache, ensuring that the next SPS/PPS will be handled when both are received.
-        SrsAutoFree(SrsRtpPacket, obs_whip_sps_);
-        SrsAutoFree(SrsRtpPacket, obs_whip_pps_);
-
-        // h264 raw to h264 packet.
-        std::string sh;
-        SrsRawH264Stream* avc = new SrsRawH264Stream();
-        SrsAutoFree(SrsRawH264Stream, avc);
+        } else {
+            // h264 raw to h264 packet.
+            std::string sh;
+            SrsRawH264Stream* avc = new SrsRawH264Stream();
+            SrsAutoFree(SrsRawH264Stream, avc);
 
-        if ((err = avc->mux_sequence_header(string(sps->bytes, sps->size), string(pps->bytes, pps->size), sh)) != srs_success) {
-            return srs_error_wrap(err, "mux sequence header");
-        }
+            if ((err = avc->mux_sequence_header(string(sps->bytes, sps->size), string(pps->bytes, pps->size), sh)) != srs_success) {
+                return srs_error_wrap(err, "mux sequence header");
+            }
 
-        // h264 packet to flv packet.
-        char* flv = NULL;
-        int nb_flv = 0;
-        if ((err = avc->mux_avc2flv(sh, SrsVideoAvcFrameTypeKeyFrame, SrsVideoAvcFrameTraitSequenceHeader, pkt->get_avsync_time(),
-                                    pkt->get_avsync_time(), &flv, &nb_flv)) != srs_success) {
-            return srs_error_wrap(err, "avc to flv");
-        }
+            // h264 packet to flv packet.
+            char* flv = NULL;
+            int nb_flv = 0;
+            if ((err = avc->mux_avc2flv(sh, SrsVideoAvcFrameTypeKeyFrame, SrsVideoAvcFrameTraitSequenceHeader, pkt->get_avsync_time(),
+                                        pkt->get_avsync_time(), &flv, &nb_flv)) != srs_success) {
+                return srs_error_wrap(err, "avc to flv");
+            }
 
-        SrsMessageHeader header;
-        header.initialize_video(nb_flv, pkt->get_avsync_time(), 1);
-        SrsCommonMessage rtmp;
-        if ((err = rtmp.create(&header, flv, nb_flv)) != srs_success) {
-            return srs_error_wrap(err, "create rtmp");
-        }
+            SrsMessageHeader header;
+            header.initialize_video(nb_flv, pkt->get_avsync_time(), 1);
+            SrsCommonMessage rtmp;
+            if ((err = rtmp.create(&header, flv, nb_flv)) != srs_success) {
+                return srs_error_wrap(err, "create rtmp");
+            }
 
-        SrsSharedPtrMessage msg;
-        if ((err = msg.create(&rtmp)) != srs_success) {
-            return srs_error_wrap(err, "create message");
-        }
+            SrsSharedPtrMessage msg;
+            if ((err = msg.create(&rtmp)) != srs_success) {
+                return srs_error_wrap(err, "create message");
+            }
 
-        if ((err = bridge_->on_frame(&msg)) != srs_success) {
-            return err;
+            if ((err = bridge_->on_frame(&msg)) != srs_success) {
+                return err;
+            }
         }
     }
 
@@ -1622,7 +1583,7 @@ srs_error_t SrsRtcFrameBuilder::packet_video_key_frame(SrsRtpPacket* pkt)
     if (-1 == sn) {
         if (check_frame_complete(header_sn_, tail_sn)) {
             if ((err = packet_video_rtmp(header_sn_, tail_sn)) != srs_success) {
-                err = srs_error_wrap(err, "fail to packet frame");
+                err = srs_error_wrap(err, "fail to packet key frame");
             }
         }
     } else if (-2 == sn) {

diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
@@ -318,13 +318,6 @@ class SrsRtcFrameBuilder
     uint16_t header_sn_;
     uint16_t lost_sn_;
     int64_t rtp_key_frame_ts_;
-private:
-    // The state for timestamp sync state. -1 for init. 0 not sync. 1 sync.
-    int sync_state_;
-private:
-    // For OBS WHIP, send SPS/PPS in dedicated RTP packet.
-    SrsRtpPacket* obs_whip_sps_;
-    SrsRtpPacket* obs_whip_pps_;
 public:
     SrsRtcFrameBuilder(ISrsStreamBridge* bridge);
     virtual ~SrsRtcFrameBuilder();

diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp
@@ -1,3 +1,4 @@
+//
 // Copyright (c) 2013-2024 The SRS Authors
 //
 // SPDX-License-Identifier: MIT

diff --git a/trunk/src/kernel/srs_kernel_rtc_rtp.cpp b/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
@@ -970,14 +970,12 @@ SrsRtpRawPayload::SrsRtpRawPayload()
 {
     payload = NULL;
     nn_payload = 0;
-    sample_ = new SrsSample();
 
     ++_srs_pps_objs_rraw->sugar;
 }
 
 SrsRtpRawPayload::~SrsRtpRawPayload()
 {
-    srs_freep(sample_);
 }
 
 uint64_t SrsRtpRawPayload::nb_bytes()
@@ -1009,9 +1007,6 @@ srs_error_t SrsRtpRawPayload::decode(SrsBuffer* buf)
     payload = buf->head();
     nn_payload = buf->left();
 
-    sample_->bytes = payload;
-    sample_->size = nn_payload;
-
     return srs_success;
 }
 
@@ -1021,7 +1016,6 @@ ISrsRtpPayloader* SrsRtpRawPayload::copy()
 
     cp->payload = payload;
     cp->nn_payload = nn_payload;
-    cp->sample_ = sample_->copy();
 
     return cp;
 }

diff --git a/trunk/src/kernel/srs_kernel_rtc_rtp.hpp b/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
@@ -348,9 +348,6 @@ class SrsRtpRawPayload : public ISrsRtpPayloader
     // @remark We only refer to the memory, user must free its bytes.
     char* payload;
     int nn_payload;
-public:
-    // Use the whole RAW RTP payload as a sample.
-    SrsSample* sample_;
 public:
     SrsRtpRawPayload();
     virtual ~SrsRtpRawPayload();