For #307, refine performance

trunk/conf/full.conf

@@ -488,6 +488,7 @@ vhost rtc.vhost.srs.com {
     min_latency     on;
     play {
         # set the MW(merged-write) latency in ms.
+        # @remark For WebRTC, we enable pass-timestamp mode, so we ignore this config.
         # default: 0 (For WebRTC)
         mw_latency      0;
         # Set the MW(merged-write) min messages.
@@ -720,6 +721,7 @@ vhost play.srs.com {
         # SRS always set mw on, so we just set the latency value.
         # the latency of stream >= mw_latency + mr_latency
         # the value recomment is [300, 1800]
+        # @remark For WebRTC, we enable pass-timestamp mode, so we ignore this config.
         # default: 350 (For RTMP/HTTP-FLV)
         # default: 0 (For WebRTC)
         mw_latency      350;

trunk/src/app/srs_app_config.cpp

@@ -5421,7 +5421,13 @@ srs_utime_t SrsConfig::get_mw_sleep(string vhost, bool is_rtc)
         return DEFAULT;
     }
 
-    return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_MILLISECONDS);
+    int v = ::atoi(conf->arg0().c_str());
+    if (is_rtc && v > 0) {
+        srs_warn("For RTC, we ignore mw_latency");
+        return 0;
+    }
+    
+    return (srs_utime_t)(v * SRS_UTIME_MILLISECONDS);
 }
 
 int SrsConfig::get_mw_msgs(string vhost, bool is_realtime, bool is_rtc)

trunk/src/app/srs_app_rtc_conn.cpp

@@ -460,7 +460,7 @@ srs_error_t SrsDtlsSession::unprotect_rtcp(char* out_buf, const char* in_buf, in
     return srs_error_new(ERROR_RTC_SRTP_UNPROTECT, "rtcp unprotect failed");
 }
 
-SrsRtcPackets::SrsRtcPackets()
+SrsRtcPackets::SrsRtcPackets(int nn_cache_max)
 {
 #if defined(SRS_DEBUG)
     debug_id = 0;
@@ -477,22 +477,20 @@ SrsRtcPackets::SrsRtcPackets()
     nn_dropped = 0;
 
     cursor = 0;
+    nn_cache = nn_cache_max;
+    cache = new SrsRtpPacket2[nn_cache];
 }
 
 SrsRtcPackets::~SrsRtcPackets()
 {
-    vector<SrsRtpPacket2*>::iterator it;
+    srs_freepa(cache);
-    for (it = packets.begin(); it != packets.end(); ++it) {
+    nn_cache = 0;
-        SrsRtpPacket2* p = *it;
-        srs_freep(p);
-    }
-    packets.clear();
 }
 
 void SrsRtcPackets::reset(bool gso, bool merge_nalus)
 {
     for (int i = 0; i < cursor; i++) {
-        SrsRtpPacket2* packet = packets[i];
+        SrsRtpPacket2* packet = cache + i;
         packet->reset();
     }
 
@@ -515,17 +513,16 @@ void SrsRtcPackets::reset(bool gso, bool merge_nalus)
 
 SrsRtpPacket2* SrsRtcPackets::fetch()
 {
-    if (cursor >= (int)packets.size()) {
+    if (cursor >= nn_cache) {
-        packets.push_back(new SrsRtpPacket2());
+        return NULL;
     }
-
+    return cache + (cursor++);
-    return packets[cursor++];
 }
 
 SrsRtpPacket2* SrsRtcPackets::back()
 {
     srs_assert(cursor > 0);
-    return packets[cursor - 1];
+    return cache + cursor - 1;
 }
 
 int SrsRtcPackets::size()
@@ -535,13 +532,13 @@ int SrsRtcPackets::size()
 
 int SrsRtcPackets::capacity()
 {
-    return (int)packets.size();
+    return nn_cache;
 }
 
 SrsRtpPacket2* SrsRtcPackets::at(int index)
 {
     srs_assert(index < cursor);
-    return packets[index];
+    return cache + index;
 }
 
 SrsRtcSenderThread::SrsRtcSenderThread(SrsRtcSession* s, SrsUdpMuxSocket* u, int parent_cid)
@@ -688,6 +685,13 @@ srs_error_t SrsRtcSenderThread::cycle()
         return srs_error_wrap(err, "rtc create consumer, source url=%s", req->get_stream_url().c_str());
     }
 
+    // For RTC, we enable pass-timestamp mode, ignore the timestamp in queue, never depends on the duration,
+    // because RTC allows the audio and video has its own timebase, that is the audio timestamp and video timestamp
+    // maybe not monotonically increase.
+    // In this mode, we use mw_msgs to set the delay. We never shrink the consumer queue, instead, we dumps the
+    // messages and drop them if the shared sender queue is full.
+    consumer->enable_pass_timestamp();
+
     realtime = _srs_config->get_realtime_enabled(req->vhost, true);
     mw_sleep = _srs_config->get_mw_sleep(req->vhost, true);
     mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime, true);
@@ -702,8 +706,8 @@ srs_error_t SrsRtcSenderThread::cycle()
     srs_trace("RTC source url=%s, source_id=[%d][%d], encrypt=%d, realtime=%d, mw_sleep=%dms, mw_msgs=%d", req->get_stream_url().c_str(),
         ::getpid(), source->source_id(), rtc_session->encrypt, realtime, srsu2msi(mw_sleep), mw_msgs);
 
-    SrsRtcPackets pkts;
     SrsMessageArray msgs(SRS_PERF_MW_MSGS);
+    SrsRtcPackets pkts(SRS_PERF_RTC_RTP_PACKETS);
 
     SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_play();
     SrsAutoFree(SrsPithyPrint, pprint);
@@ -748,27 +752,29 @@ srs_error_t SrsRtcSenderThread::cycle()
         }
 
         // Stat for performance analysis.
-        if (stat_enabled) {
+        if (!stat_enabled) {
-            // Stat the original RAW AV frame, maybe h264+aac.
+            continue;
-            stat->perf_on_msgs(msg_count);
+        }
-            // Stat the RTC packets, RAW AV frame, maybe h.264+opus.
+
-            int nn_rtc_packets = srs_max(pkts.nn_audios, pkts.nn_extras) + pkts.nn_videos;
+        // Stat the original RAW AV frame, maybe h264+aac.
-            stat->perf_on_rtc_packets(nn_rtc_packets);
+        stat->perf_on_msgs(msg_count);
-            // Stat the RAW RTP packets, which maybe group by GSO.
+        // Stat the RTC packets, RAW AV frame, maybe h.264+opus.
-            stat->perf_on_rtp_packets(pkts.size());
+        int nn_rtc_packets = srs_max(pkts.nn_audios, pkts.nn_extras) + pkts.nn_videos;
-            // Stat the RTP packets going into kernel.
+        stat->perf_on_rtc_packets(nn_rtc_packets);
-            stat->perf_on_gso_packets(pkts.nn_rtp_pkts);
+        // Stat the RAW RTP packets, which maybe group by GSO.
-            // Stat the bytes and paddings.
+        stat->perf_on_rtp_packets(pkts.size());
-            stat->perf_on_rtc_bytes(pkts.nn_bytes, pkts.nn_rtp_bytes, pkts.nn_padding_bytes);
+        // Stat the RTP packets going into kernel.
-            // Stat the messages and dropped count.
+        stat->perf_on_gso_packets(pkts.nn_rtp_pkts);
-            stat->perf_on_dropped(msg_count, nn_rtc_packets, pkts.nn_dropped);
+        // Stat the bytes and paddings.
+        stat->perf_on_rtc_bytes(pkts.nn_bytes, pkts.nn_rtp_bytes, pkts.nn_padding_bytes);
+        // Stat the messages and dropped count.
+        stat->perf_on_dropped(msg_count, nn_rtc_packets, pkts.nn_dropped);
 
 #if defined(SRS_DEBUG)
-            srs_trace("RTC PLAY perf, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d/%d/%d bytes",
+        srs_trace("RTC PLAY perf, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d/%d/%d bytes",
-                msg_count, nn_rtc_packets, pkts.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
+            msg_count, nn_rtc_packets, pkts.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
-                pkts.nn_samples, pkts.nn_bytes, pkts.nn_rtp_bytes, pkts.nn_padding_bytes);
+            pkts.nn_samples, pkts.nn_bytes, pkts.nn_rtp_bytes, pkts.nn_padding_bytes);
 #endif
-        }
 
         pprint->elapse();
         if (pprint->can_print()) {
@@ -1193,6 +1199,10 @@ srs_error_t SrsRtcSenderThread::packet_nalus(SrsSharedPtrMessage* msg, SrsRtcPac
     if (nn_bytes < kRtpMaxPayloadSize) {
         // Package NALUs in a single RTP packet.
         SrsRtpPacket2* packet = packets.fetch();
+        if (!packet) {
+            srs_freep(raw);
+            return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
+        }
 
         packet->rtp_header.set_timestamp(msg->timestamp * 90);
         packet->rtp_header.set_sequence(video_sequence++);
@@ -1216,13 +1226,19 @@ srs_error_t SrsRtcSenderThread::packet_nalus(SrsSharedPtrMessage* msg, SrsRtcPac
             int packet_size = srs_min(nb_left, fu_payload_size);
 
             SrsRtpPacket2* packet = packets.fetch();
+            if (!packet) {
+                srs_freep(raw);
+                return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
+            }
+            packet->extra_payload = raw;
 
             packet->rtp_header.set_timestamp(msg->timestamp * 90);
             packet->rtp_header.set_sequence(video_sequence++);
             packet->rtp_header.set_ssrc(video_ssrc);
             packet->rtp_header.set_payload_type(video_payload_type);
 
-            SrsRtpFUAPayload* fua = packet->reuse_fua();
+            SrsRtpFUAPayload* fua = new SrsRtpFUAPayload();
+            packet->payload = fua;
 
             fua->nri = (SrsAvcNaluType)header;
             fua->nalu_type = (SrsAvcNaluType)nal_type;
@@ -1249,6 +1265,9 @@ srs_error_t SrsRtcSenderThread::packet_opus(SrsSample* sample, SrsRtcPackets& pa
     srs_error_t err = srs_success;
 
     SrsRtpPacket2* packet = packets.fetch();
+    if (!packet) {
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
+    }
     packet->rtp_header.set_marker(true);
     packet->rtp_header.set_timestamp(audio_timestamp);
     packet->rtp_header.set_sequence(audio_sequence++);
@@ -1291,23 +1310,24 @@ srs_error_t SrsRtcSenderThread::packet_fu_a(SrsSharedPtrMessage* msg, SrsSample*
         int packet_size = srs_min(nb_left, fu_payload_size);
 
         SrsRtpPacket2* packet = packets.fetch();
+        if (!packet) {
+            return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
+        }
 
         packet->rtp_header.set_timestamp(msg->timestamp * 90);
         packet->rtp_header.set_sequence(video_sequence++);
         packet->rtp_header.set_ssrc(video_ssrc);
         packet->rtp_header.set_payload_type(video_payload_type);
 
-        SrsRtpFUAPayload* fua = packet->reuse_fua();
+        SrsRtpFUAPayload2* fua = packet->reuse_fua();
 
         fua->nri = (SrsAvcNaluType)header;
         fua->nalu_type = (SrsAvcNaluType)nal_type;
         fua->start = bool(i == 0);
         fua->end = bool(i == num_of_packet - 1);
 
-        SrsSample* fragment_sample = new SrsSample();
+        fua->payload = p;
-        fragment_sample->bytes = p;
+        fua->size = packet_size;
-        fragment_sample->size = packet_size;
-        fua->nalus.push_back(fragment_sample);
 
         p += packet_size;
         nb_left -= packet_size;
@@ -1322,6 +1342,9 @@ srs_error_t SrsRtcSenderThread::packet_single_nalu(SrsSharedPtrMessage* msg, Srs
     srs_error_t err = srs_success;
 
     SrsRtpPacket2* packet = packets.fetch();
+    if (!packet) {
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
+    }
     packet->rtp_header.set_timestamp(msg->timestamp * 90);
     packet->rtp_header.set_sequence(video_sequence++);
     packet->rtp_header.set_ssrc(video_ssrc);
@@ -1355,6 +1378,9 @@ srs_error_t SrsRtcSenderThread::packet_stap_a(SrsSource* source, SrsSharedPtrMes
     }
 
     SrsRtpPacket2* packet = packets.fetch();
+    if (!packet) {
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
+    }
     packet->rtp_header.set_marker(false);
     packet->rtp_header.set_timestamp(msg->timestamp * 90);
     packet->rtp_header.set_sequence(video_sequence++);
@@ -2017,23 +2043,22 @@ srs_error_t SrsUdpMuxSender::cycle()
         cache.swap(hotspot);
         cache_pos = 0;
 
-        // Collect informations for GSO.
         int gso_pos = 0;
-        if (pos > 0 && stat_enabled) {
-            // For shared GSO cache, stat the messages.
-            // @see https://linux.die.net/man/2/sendmmsg
-            // @see https://linux.die.net/man/2/sendmsg
-            for (int i = 0; i < pos; i++) {
-                mmsghdr* mhdr = &hotspot[i];
-
-                int real_iovs = mhdr->msg_hdr.msg_iovlen;
-                gso_pos++; nn_gso_msgs++; nn_gso_iovs += real_iovs;
-                gso_iovs += real_iovs;
-            }
-        }
-
-        // Send out all messages.
         if (pos > 0) {
+            // Collect informations for GSO.
+            if (stat_enabled) {
+                // For shared GSO cache, stat the messages.
+                // @see https://linux.die.net/man/2/sendmmsg
+                // @see https://linux.die.net/man/2/sendmsg
+                for (int i = 0; i < pos; i++) {
+                    mmsghdr* mhdr = &hotspot[i];
+
+                    int real_iovs = mhdr->msg_hdr.msg_iovlen;
+                    gso_pos++; nn_gso_msgs++; nn_gso_iovs += real_iovs;
+                    gso_iovs += real_iovs;
+                }
+            }
+
             // Send out all messages.
             // @see https://linux.die.net/man/2/sendmmsg
             // @see https://linux.die.net/man/2/sendmsg
@@ -2053,6 +2078,10 @@ srs_error_t SrsUdpMuxSender::cycle()
             }
         }
 
+        if (!stat_enabled) {
+            continue;
+        }
+
         // Increase total messages.
         nn_msgs += pos + gso_iovs;
         nn_msgs_max = srs_max(pos, nn_msgs_max);

trunk/src/app/srs_app_rtc_conn.hpp

@@ -155,9 +155,10 @@ public:
     int nn_dropped;
 private:
     int cursor;
-    std::vector<SrsRtpPacket2*> packets;
+    int nn_cache;
+    SrsRtpPacket2* cache;
 public:
-    SrsRtcPackets();
+    SrsRtcPackets(int nn_cache_max);
     virtual ~SrsRtcPackets();
 public:
     void reset(bool gso, bool merge_nalus);

trunk/src/app/srs_app_source.cpp

@@ -269,10 +269,18 @@ void SrsMessageQueue::set_queue_size(srs_utime_t queue_size)
 	max_queue_size = queue_size;
 }
 
-srs_error_t SrsMessageQueue::enqueue(SrsSharedPtrMessage* msg, bool* is_overflow)
+srs_error_t SrsMessageQueue::enqueue(SrsSharedPtrMessage* msg, bool* is_overflow, bool pass_timestamp)
 {
     srs_error_t err = srs_success;
 
+    msgs.push_back(msg);
+
+    // For RTC, we never care about the timestamp and duration, so we never shrink queue here,
+    // but we will drop messages in each consumer coroutine.
+    if (pass_timestamp) {
+        return err;
+    }
+    
     if (msg->is_av()) {
         if (av_start_time == -1) {
             av_start_time = srs_utime_t(msg->timestamp * SRS_UTIME_MILLISECONDS);
@@ -281,8 +289,6 @@ srs_error_t SrsMessageQueue::enqueue(SrsSharedPtrMessage* msg, bool* is_overflow
         av_end_time = srs_utime_t(msg->timestamp * SRS_UTIME_MILLISECONDS);
     }
     
-    msgs.push_back(msg);
-    
     while (av_end_time - av_start_time > max_queue_size) {
         // notice the caller queue already overflow and shrinked.
         if (is_overflow) {
@@ -295,7 +301,7 @@ srs_error_t SrsMessageQueue::enqueue(SrsSharedPtrMessage* msg, bool* is_overflow
     return err;
 }
 
-srs_error_t SrsMessageQueue::dump_packets(int max_count, SrsSharedPtrMessage** pmsgs, int& count)
+srs_error_t SrsMessageQueue::dump_packets(int max_count, SrsSharedPtrMessage** pmsgs, int& count, bool pass_timestamp)
 {
     srs_error_t err = srs_success;
     
@@ -308,12 +314,14 @@ srs_error_t SrsMessageQueue::dump_packets(int max_count, SrsSharedPtrMessage** p
     count = srs_min(max_count, nb_msgs);
     
     SrsSharedPtrMessage** omsgs = msgs.data();
-    for (int i = 0; i < count; i++) {
+    memcpy(pmsgs, omsgs, count * sizeof(SrsSharedPtrMessage*));
-        pmsgs[i] = omsgs[i];
-    }
 
-    SrsSharedPtrMessage* last = omsgs[count - 1];
+    // For RTC, we enable pass_timestamp mode, which never care about the timestamp and duration,
-    av_start_time = srs_utime_t(last->timestamp * SRS_UTIME_MILLISECONDS);
+    // so we do not have to update the start time here.
+    if (!pass_timestamp) {
+        SrsSharedPtrMessage* last = omsgs[count - 1];
+        av_start_time = srs_utime_t(last->timestamp * SRS_UTIME_MILLISECONDS);
+    }
     
     if (count >= nb_msgs) {
         // the pmsgs is big enough and clear msgs at most time.
@@ -433,6 +441,8 @@ SrsConsumer::SrsConsumer(SrsSource* s, SrsConnection* c)
     mw_duration = 0;
     mw_waiting = false;
 #endif
+
+    pass_timestamp = false;
 }
 
 SrsConsumer::~SrsConsumer()
@@ -467,19 +477,34 @@ srs_error_t SrsConsumer::enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsR
     
     SrsSharedPtrMessage* msg = shared_msg->copy();
 
-    if (!atc) {
+    // For RTC, we enable pass_timestamp mode, which never correct or depends on monotonic increasing of
+    // timestamp. And in RTC, the audio and video timebase can be different, so we ignore time_jitter here.
+    if (!pass_timestamp && !atc) {
         if ((err = jitter->correct(msg, ag)) != srs_success) {
             return srs_error_wrap(err, "consume message");
         }
     }
 
-    if ((err = queue->enqueue(msg, NULL)) != srs_success) {
+    // Put message in queue, here we may enable pass_timestamp mode.
+    if ((err = queue->enqueue(msg, NULL, pass_timestamp)) != srs_success) {
         return srs_error_wrap(err, "enqueue message");
     }
     
 #ifdef SRS_PERF_QUEUE_COND_WAIT
     // fire the mw when msgs is enough.
     if (mw_waiting) {
+        // For RTC, we use pass_timestamp mode, we don't care about the timestamp in queue,
+        // so we only check the messages in queue.
+        if (pass_timestamp) {
+            if (queue->size() > mw_min_msgs) {
+                srs_cond_signal(mw_wait);
+                mw_waiting = false;
+                return err;
+            }
+            return err;
+        }
+
+        // For RTMP, we wait for messages and duration.
         srs_utime_t duration = queue->duration();
         bool match_min_msgs = queue->size() > mw_min_msgs;
         
@@ -529,7 +554,7 @@ srs_error_t SrsConsumer::dump_packets(SrsMessageArray* msgs, int& count)
     }
     
     // pump msgs from queue.
-    if ((err = queue->dump_packets(max, msgs->msgs, count)) != srs_success) {
+    if ((err = queue->dump_packets(max, msgs->msgs, count, pass_timestamp)) != srs_success) {
         return srs_error_wrap(err, "dump packets");
     }
     

trunk/src/app/srs_app_source.hpp

@@ -151,12 +151,13 @@ public:
     // Enqueue the message, the timestamp always monotonically.
     // @param msg, the msg to enqueue, user never free it whatever the return code.
     // @param is_overflow, whether overflow and shrinked. NULL to ignore.
-    virtual srs_error_t enqueue(SrsSharedPtrMessage* msg, bool* is_overflow = NULL);
+    // @remark If pass_timestamp, we never shrink and never care about the timestamp or duration.
+    virtual srs_error_t enqueue(SrsSharedPtrMessage* msg, bool* is_overflow = NULL, bool pass_timestamp = false);
     // Get packets in consumer queue.
     // @pmsgs SrsSharedPtrMessage*[], used to store the msgs, user must alloc it.
     // @count the count in array, output param.
     // @max_count the max count to dequeue, must be positive.
-    virtual srs_error_t dump_packets(int max_count, SrsSharedPtrMessage** pmsgs, int& count);
+    virtual srs_error_t dump_packets(int max_count, SrsSharedPtrMessage** pmsgs, int& count, bool pass_timestamp = false);
     // Dumps packets to consumer, use specified args.
     // @remark the atc/tba/tbv/ag are same to SrsConsumer.enqueue().
     virtual srs_error_t dump_packets(SrsConsumer* consumer, bool atc, SrsRtmpJitterAlgorithm ag);
@@ -203,10 +204,17 @@ private:
     int mw_min_msgs;
     srs_utime_t mw_duration;
 #endif
+private:
+    // For RTC, we never use jitter to correct timestamp.
+    // But we should not change the atc or time_jitter for source or RTMP.
+    // @remark In this mode, we also never check the queue by timstamp, but only by count.
+    bool pass_timestamp;
 public:
     SrsConsumer(SrsSource* s, SrsConnection* c);
     virtual ~SrsConsumer();
 public:
+    // Use pass timestamp mode.
+    void enable_pass_timestamp() { pass_timestamp = true; }
     // Set the size of queue.
     virtual void set_queue_size(srs_utime_t queue_size);
     // when source id changed, notice client to print.

trunk/src/core/srs_core_performance.hpp

@@ -211,5 +211,8 @@
 // For RTC, the max iovs in msghdr, the max packets sent in a msghdr.
 #define SRS_PERF_RTC_GSO_MAX 64
 
+// For RTC, the max count of RTP packets we process in one loop.
+#define SRS_PERF_RTC_RTP_PACKETS 1024
+
 #endif
 

trunk/src/kernel/srs_kernel_buffer.cpp

@@ -67,11 +67,6 @@ SrsBuffer::~SrsBuffer()
 {
 }
 
-char* SrsBuffer::data()
-{
-    return bytes;
-}
-
 int SrsBuffer::size()
 {
     return nb_bytes;

trunk/src/kernel/srs_kernel_buffer.hpp

@@ -113,7 +113,8 @@ public:
      * get data of stream, set by initialize.
      * current bytes = data() + pos()
      */
-    virtual char* data();
+    inline char* data() { return bytes; }
+    inline char* head() { return p; }
     /**
      * the total stream size, set by initialize.
      * left bytes = size() - pos().

trunk/src/kernel/srs_kernel_rtp.cpp

@@ -85,7 +85,7 @@ srs_error_t SrsRtpHeader::encode(SrsBuffer* stream)
 
     // Encode the RTP fix header, 12bytes.
     // @see https://tools.ietf.org/html/rfc1889#section-5.1
-    char* op = stream->data() + stream->pos();
+    char* op = stream->head();
     char* p = op;
 
     // The version, padding, extension and cc, total 1 byte.
@@ -148,60 +148,45 @@ size_t SrsRtpHeader::header_size()
     return kRtpHeaderFixedSize + cc * 4 + (extension ? (extension_length + 1) * 4 : 0);
 }
 
-void SrsRtpHeader::set_marker(bool marker)
-{
-    this->marker = marker;
-}
-
-void SrsRtpHeader::set_payload_type(uint8_t payload_type)
-{
-    this->payload_type = payload_type;
-}
-
-void SrsRtpHeader::set_sequence(uint16_t sequence)
-{
-    this->sequence = sequence;
-}
-
-void SrsRtpHeader::set_timestamp(int64_t timestamp)
-{
-    this->timestamp = (uint32_t)timestamp;
-}
-
-void SrsRtpHeader::set_ssrc(uint32_t ssrc)
-{
-    this->ssrc = ssrc;
-}
-
 SrsRtpPacket2::SrsRtpPacket2()
 {
     payload = NULL;
+    extra_payload = NULL;
     padding = 0;
 
     cache_raw = new SrsRtpRawPayload();
-    cache_fua = new SrsRtpFUAPayload();
+    cache_fua = new SrsRtpFUAPayload2();
+    cache_payload = 0;
+    using_cache = false;
 }
 
 SrsRtpPacket2::~SrsRtpPacket2()
 {
     // We may use the cache as payload.
-    if (payload == cache_raw || payload == cache_fua) {
+    if (using_cache) {
         payload = NULL;
     }
 
     srs_freep(payload);
+    srs_freep(extra_payload);
     srs_freep(cache_raw);
 }
 
 void SrsRtpPacket2::set_padding(int size)
 {
     rtp_header.set_padding(size > 0);
+    if (cache_payload) {
+        cache_payload += size - padding;
+    }
     padding = size;
 }
 
 void SrsRtpPacket2::add_padding(int size)
 {
     rtp_header.set_padding(padding + size > 0);
+    if (cache_payload) {
+        cache_payload += size;
+    }
     padding += size;
 }
 
@@ -209,31 +194,39 @@ void SrsRtpPacket2::reset()
 {
     rtp_header.reset();
     padding = 0;
+    cache_payload = 0;
+    srs_freep(extra_payload);
 
     // We may use the cache as payload.
-    if (payload == cache_raw || payload == cache_fua) {
+    if (using_cache) {
         payload = NULL;
+    } else {
+        srs_freep(payload);
     }
 
-    srs_freep(payload);
+    using_cache = false;
 }
 
 SrsRtpRawPayload* SrsRtpPacket2::reuse_raw()
 {
+    using_cache = true;
     payload = cache_raw;
     return cache_raw;
 }
 
-SrsRtpFUAPayload* SrsRtpPacket2::reuse_fua()
+SrsRtpFUAPayload2* SrsRtpPacket2::reuse_fua()
 {
+    using_cache = true;
     payload = cache_fua;
-    cache_fua->reset();
     return cache_fua;
 }
 
 int SrsRtpPacket2::nb_bytes()
 {
-    return rtp_header.header_size() + (payload? payload->nb_bytes():0) + padding;
+    if (!cache_payload) {
+        cache_payload = rtp_header.header_size() + (payload? payload->nb_bytes():0) + padding;
+    }
+    return cache_payload;
 }
 
 srs_error_t SrsRtpPacket2::encode(SrsBuffer* buf)
@@ -297,12 +290,20 @@ SrsRtpRawNALUs::SrsRtpRawNALUs()
 
 SrsRtpRawNALUs::~SrsRtpRawNALUs()
 {
-    vector<SrsSample*>::iterator it;
+    if (true) {
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+        int nn_nalus = (int)nalus.size();
-        SrsSample* p = *it;
+        for (int i = 0; i < nn_nalus; i++) {
-        srs_freep(p);
+            SrsSample* p = nalus[i];
+            srs_freep(p);
+        }
+    }
+    if (true) {
+        int nn_nalus = (int)extra_nalus.size();
+        for (int i = 0; i < nn_nalus; i++) {
+            SrsSample* p = extra_nalus[i];
+            srs_freep(p);
+        }
     }
-    nalus.clear();
 }
 
 void SrsRtpRawNALUs::push_back(SrsSample* sample)
@@ -330,19 +331,19 @@ uint8_t SrsRtpRawNALUs::skip_first_byte()
     return uint8_t(nalus[0]->bytes[0]);
 }
 
-srs_error_t SrsRtpRawNALUs::read_samples(vector<SrsSample*>& samples, int size)
+srs_error_t SrsRtpRawNALUs::read_samples(vector<SrsSample*>& samples, int packet_size)
 {
-    if (cursor + size < 0 || cursor + size > nn_bytes) {
+    if (cursor + packet_size < 0 || cursor + packet_size > nn_bytes) {
-        return srs_error_new(ERROR_RTC_RTP_MUXER, "cursor=%d, max=%d, size=%d", cursor, nn_bytes, size);
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "cursor=%d, max=%d, size=%d", cursor, nn_bytes, packet_size);
     }
 
     int pos = cursor;
-    cursor += size;
+    cursor += packet_size;
-    int left = size;
+    int left = packet_size;
 
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end() && left > 0; ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
 
         // Ignore previous consumed samples.
         if (pos && pos - p->size >= 0) {
@@ -355,9 +356,11 @@ srs_error_t SrsRtpRawNALUs::read_samples(vector<SrsSample*>& samples, int size)
         srs_assert(nn > 0);
 
         SrsSample* sample = new SrsSample();
+        extra_nalus.push_back(sample);
+        samples.push_back(sample);
+
         sample->bytes = p->bytes + pos;
         sample->size = nn;
-        samples.push_back(sample);
 
         left -= nn;
         pos = 0;
@@ -370,9 +373,9 @@ int SrsRtpRawNALUs::nb_bytes()
 {
     int size = 0;
 
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
         size += p->size;
     }
 
@@ -381,9 +384,9 @@ int SrsRtpRawNALUs::nb_bytes()
 
 srs_error_t SrsRtpRawNALUs::encode(SrsBuffer* buf)
 {
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
 
         if (!buf->require(p->size)) {
             return srs_error_new(ERROR_RTC_RTP_MUXER, "requires %d bytes", p->size);
@@ -402,21 +405,20 @@ SrsRtpSTAPPayload::SrsRtpSTAPPayload()
 
 SrsRtpSTAPPayload::~SrsRtpSTAPPayload()
 {
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
         srs_freep(p);
     }
-    nalus.clear();
 }
 
 int SrsRtpSTAPPayload::nb_bytes()
 {
     int size = 1;
 
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
         size += 2 + p->size;
     }
 
@@ -436,9 +438,10 @@ srs_error_t SrsRtpSTAPPayload::encode(SrsBuffer* buf)
     buf->write_1bytes(v);
 
     // NALUs.
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
+
         if (!buf->require(2 + p->size)) {
             return srs_error_new(ERROR_RTC_RTP_MUXER, "requires %d bytes", 2 + p->size);
         }
@@ -458,31 +461,20 @@ SrsRtpFUAPayload::SrsRtpFUAPayload()
 
 SrsRtpFUAPayload::~SrsRtpFUAPayload()
 {
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
         srs_freep(p);
     }
-    nalus.clear();
-}
-
-void SrsRtpFUAPayload::reset()
-{
-    vector<SrsSample*>::iterator it;
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
-        SrsSample* p = *it;
-        srs_freep(p);
-    }
-    nalus.clear();
 }
 
 int SrsRtpFUAPayload::nb_bytes()
 {
     int size = 2;
 
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
         size += p->size;
     }
 
@@ -511,9 +503,10 @@ srs_error_t SrsRtpFUAPayload::encode(SrsBuffer* buf)
     buf->write_1bytes(fu_header);
 
     // FU payload, @see https://tools.ietf.org/html/rfc6184#section-5.8
-    vector<SrsSample*>::iterator it;
+    int nn_nalus = (int)nalus.size();
-    for (it = nalus.begin(); it != nalus.end(); ++it) {
+    for (int i = 0; i < nn_nalus; i++) {
-        SrsSample* p = *it;
+        SrsSample* p = nalus[i];
+
         if (!buf->require(p->size)) {
             return srs_error_new(ERROR_RTC_RTP_MUXER, "requires %d bytes", p->size);
         }
@@ -524,6 +517,57 @@ srs_error_t SrsRtpFUAPayload::encode(SrsBuffer* buf)
     return srs_success;
 }
 
+SrsRtpFUAPayload2::SrsRtpFUAPayload2()
+{
+    start = end = false;
+    nri = nalu_type = (SrsAvcNaluType)0;
+
+    payload = NULL;
+    size = 0;
+}
+
+SrsRtpFUAPayload2::~SrsRtpFUAPayload2()
+{
+}
+
+int SrsRtpFUAPayload2::nb_bytes()
+{
+    return 2 + size;
+}
+
+srs_error_t SrsRtpFUAPayload2::encode(SrsBuffer* buf)
+{
+    if (!buf->require(2 + size)) {
+        return srs_error_new(ERROR_RTC_RTP_MUXER, "requires %d bytes", 1);
+    }
+
+    // Fast encoding.
+    char* p = buf->head();
+
+    // FU indicator, @see https://tools.ietf.org/html/rfc6184#section-5.8
+    uint8_t fu_indicate = kFuA;
+    fu_indicate |= (nri & (~kNalTypeMask));
+    *p++ = fu_indicate;
+
+    // FU header, @see https://tools.ietf.org/html/rfc6184#section-5.8
+    uint8_t fu_header = nalu_type;
+    if (start) {
+        fu_header |= kStart;
+    }
+    if (end) {
+        fu_header |= kEnd;
+    }
+    *p++ = fu_header;
+
+    // FU payload, @see https://tools.ietf.org/html/rfc6184#section-5.8
+    memcpy(p, payload, size);
+
+    // Consume bytes.
+    buf->skip(2 + size);
+
+    return srs_success;
+}
+
 SrsRtpSharedPacket::SrsRtpSharedPacketPayload::SrsRtpSharedPacketPayload()
 {
     payload = NULL;

trunk/src/kernel/srs_kernel_rtp.hpp

@@ -39,7 +39,7 @@ const uint8_t kNalTypeMask      = 0x1F;
 
 class SrsBuffer;
 class SrsRtpRawPayload;
-class SrsRtpFUAPayload;
+class SrsRtpFUAPayload2;
 
 class SrsRtpHeader
 {
@@ -65,17 +65,17 @@ public:
 public:
     size_t header_size();
 public:
-    void set_marker(bool marker);
+    inline void set_marker(bool v) { marker = v; }
     bool get_marker() const { return marker; }
-    void set_payload_type(uint8_t payload_type);
+    inline void set_payload_type(uint8_t v) { payload_type = v; }
     uint8_t get_payload_type() const { return payload_type; }
-    void set_sequence(uint16_t sequence);
+    inline void set_sequence(uint16_t v) { sequence = v; }
     uint16_t get_sequence() const  { return sequence; }
-    void set_timestamp(int64_t timestamp);
+    inline void set_timestamp(int64_t v) { timestamp = (uint32_t)v; }
     int64_t get_timestamp() const { return timestamp; }
-    void set_ssrc(uint32_t ssrc);
+    inline void set_ssrc(uint32_t v) { ssrc = v; }
     uint32_t get_ssrc() const { return ssrc; }
-    void set_padding(bool v) { padding = v; }
+    inline void set_padding(bool v) { padding = v; }
 };
 
 class SrsRtpPacket2
@@ -84,9 +84,16 @@ public:
     SrsRtpHeader rtp_header;
     ISrsEncoder* payload;
     int padding;
+public:
+    // User can set an extra payload, we will free it.
+    // For example, when reassemble NALUs by SrsRtpRawNALUs, we can set the extra payload to
+    // SrsRtpRawNALUs, then we can use SrsRtpFUAPayload which never free samples.
+    ISrsEncoder* extra_payload;
 private:
     SrsRtpRawPayload* cache_raw;
-    SrsRtpFUAPayload* cache_fua;
+    SrsRtpFUAPayload2* cache_fua;
+    bool using_cache;
+    int cache_payload;
 public:
     SrsRtpPacket2();
     virtual ~SrsRtpPacket2();
@@ -100,7 +107,7 @@ public:
     // Reuse the cached raw message as payload.
     SrsRtpRawPayload* reuse_raw();
     // Reuse the cached fua message as payload.
-    SrsRtpFUAPayload* reuse_fua();
+    SrsRtpFUAPayload2* reuse_fua();
 // interface ISrsEncoder
 public:
     virtual int nb_bytes();
@@ -111,7 +118,8 @@ public:
 class SrsRtpRawPayload : public ISrsEncoder
 {
 public:
-    // @remark We only refer to the memory, user must free it.
+    // The RAW payload, directly point to the shared memory.
+    // @remark We only refer to the memory, user must free its bytes.
     char* payload;
     int nn_payload;
 public:
@@ -127,6 +135,9 @@ public:
 class SrsRtpRawNALUs : public ISrsEncoder
 {
 private:
+    // The returned samples.
+    std::vector<SrsSample*> extra_nalus;
+    // We will manage the samples, but the sample itself point to the shared memory.
     std::vector<SrsSample*> nalus;
     int nn_bytes;
     int cursor;
@@ -137,7 +148,8 @@ public:
     void push_back(SrsSample* sample);
 public:
     uint8_t skip_first_byte();
-    srs_error_t read_samples(std::vector<SrsSample*>& samples, int size);
+    // We will manage the returned samples, if user want to manage it, please copy it.
+    srs_error_t read_samples(std::vector<SrsSample*>& samples, int packet_size);
 // interface ISrsEncoder
 public:
     virtual int nb_bytes();
@@ -163,6 +175,7 @@ public:
 };
 
 // FU-A, for one NALU with multiple fragments.
+// With more than one payload.
 class SrsRtpFUAPayload : public ISrsEncoder
 {
 public:
@@ -178,8 +191,29 @@ public:
 public:
     SrsRtpFUAPayload();
     virtual ~SrsRtpFUAPayload();
+// interface ISrsEncoder
 public:
-    void reset();
+    virtual int nb_bytes();
+    virtual srs_error_t encode(SrsBuffer* buf);
+};
+
+// FU-A, for one NALU with multiple fragments.
+// With only one payload.
+class SrsRtpFUAPayload2 : public ISrsEncoder
+{
+public:
+    // The NRI in NALU type.
+    SrsAvcNaluType nri;
+    // The FUA header.
+    bool start;
+    bool end;
+    SrsAvcNaluType nalu_type;
+    // The payload and size,
+    char* payload;
+    int size;
+public:
+    SrsRtpFUAPayload2();
+    virtual ~SrsRtpFUAPayload2();
 // interface ISrsEncoder
 public:
     virtual int nb_bytes();