Rename SrsRtpPacket2 to SrsRtpPacket. 4.0.111

README.md

@@ -176,6 +176,7 @@ The ports used by SRS:
 
 ## V4 changes
 
+* v4.0, 2021-05-15, Rename SrsRtpPacket2 to SrsRtpPacket. 4.0.111
 * v4.0, 2021-05-14, RTC: Remove [Object Cache Pool](https://github.com/ossrs/srs/commit/14bfc98122bba369572417c19ebb2a61b373fc45#commitcomment-47655008), no effect. 4.0.110
 * v4.0, 2021-05-14, Change virtual public to public. 4.0.109
 * v4.0, 2021-05-14, Refine id and vid for statistic. 4.0.108

trunk/src/app/srs_app_gb28181.hpp

@@ -77,7 +77,7 @@
 #define STREAM_TYPE_AUDIO_PCM       0x9c
 
 class SrsConfDirective;
-class SrsRtpPacket;
+class SrsRtspPacket;
 class SrsRtmpClient;
 class SrsRawH264Stream;
 class SrsRawAacStream;
@@ -108,7 +108,7 @@ class SrsGb28181Caster;
 
 //ps rtp header packet parse
 
-class SrsPsRtpPacket: public SrsRtpPacket
+class SrsPsRtpPacket: public SrsRtspPacket
 {
 public:
     SrsPsRtpPacket();

trunk/src/app/srs_app_rtc_conn.cpp

@@ -594,7 +594,7 @@ srs_error_t SrsRtcPlayStream::cycle()
         }
 
         // Wait for amount of packets.
-        SrsRtpPacket2* pkt = NULL;
+        SrsRtpPacket* pkt = NULL;
         consumer->dump_packet(&pkt);
         if (!pkt) {
             // TODO: FIXME: We should check the quit event.
@@ -618,7 +618,7 @@ srs_error_t SrsRtcPlayStream::cycle()
     }
 }
 
-srs_error_t SrsRtcPlayStream::send_packet(SrsRtpPacket2*& pkt)
+srs_error_t SrsRtcPlayStream::send_packet(SrsRtpPacket*& pkt)
 {
     srs_error_t err = srs_success;
 
@@ -1287,7 +1287,7 @@ srs_error_t SrsRtcPublishStream::on_rtp_plaintext(char* plaintext, int nb_plaint
     }
 
     // Allocate packet form cache.
-    SrsRtpPacket2* pkt = new SrsRtpPacket2();
+    SrsRtpPacket* pkt = new SrsRtpPacket();
 
     // Copy the packet body.
     char* p = pkt->wrap(plaintext, nb_plaintext);
@@ -1305,7 +1305,7 @@ srs_error_t SrsRtcPublishStream::on_rtp_plaintext(char* plaintext, int nb_plaint
     return err;
 }
 
-srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket2*& pkt, SrsBuffer* buf)
+srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuffer* buf)
 {
     srs_error_t err = srs_success;
 
@@ -1389,7 +1389,7 @@ srs_error_t SrsRtcPublishStream::check_send_nacks()
     return err;
 }
 
-void SrsRtcPublishStream::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt)
+void SrsRtcPublishStream::on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt)
 {
     // No payload, ignore.
     if (buf->empty()) {
@@ -2528,7 +2528,7 @@ void SrsRtcConnection::simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes
     nn_simulate_player_nack_drop--;
 }
 
-srs_error_t SrsRtcConnection::do_send_packet(SrsRtpPacket2* pkt)
+srs_error_t SrsRtcConnection::do_send_packet(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 

trunk/src/app/srs_app_rtc_conn.hpp

@@ -53,7 +53,7 @@ class SrsRtcServer;
 class SrsRtcConnection;
 class SrsSharedPtrMessage;
 class SrsRtcStream;
-class SrsRtpPacket2;
+class SrsRtpPacket;
 class ISrsCodec;
 class SrsRtpNackForReceiver;
 class SrsRtpIncommingVideoFrame;
@@ -266,7 +266,7 @@ class SrsRtcPlayStream : public ISrsCoroutineHandler, public ISrsReloadHandler
 public:
     virtual srs_error_t cycle();
 private:
-    srs_error_t send_packet(SrsRtpPacket2*& pkt);
+    srs_error_t send_packet(SrsRtpPacket*& pkt);
 public:
     // Directly set the status of track, generally for init to set the default value.
     void set_all_tracks_status(bool status);
@@ -310,7 +310,7 @@ class SrsRtcPublishTwccTimer : public ISrsFastTimer
 };
 
 // A RTC publish stream, client push and publish stream to SRS.
-class SrsRtcPublishStream : public ISrsRtpPacketDecodeHandler
+class SrsRtcPublishStream : public ISrsRtspPacketDecodeHandler
     , public ISrsRtcPublishStream, public ISrsRtcPLIWorkerHandler
 {
 private:
@@ -367,11 +367,11 @@ class SrsRtcPublishStream : public ISrsRtpPacketDecodeHandler
     // @remark We copy the plaintext, user should free it.
     srs_error_t on_rtp_plaintext(char* plaintext, int nb_plaintext);
 private:
-    srs_error_t do_on_rtp_plaintext(SrsRtpPacket2*& pkt, SrsBuffer* buf);
+    srs_error_t do_on_rtp_plaintext(SrsRtpPacket*& pkt, SrsBuffer* buf);
 public:
     srs_error_t check_send_nacks();
 public:
-    virtual void on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtpPacketPayloadType* ppt);
+    virtual void on_before_decode_payload(SrsRtpPacket* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload, SrsRtspPacketPayloadType* ppt);
 private:
     srs_error_t send_periodic_twcc();
 public:
@@ -547,7 +547,7 @@ class SrsRtcConnection : public ISrsResource, public ISrsDisposingHandler
     // Simulate the NACK to drop nn packets.
     void simulate_nack_drop(int nn);
     void simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes);
-    srs_error_t do_send_packet(SrsRtpPacket2* pkt);
+    srs_error_t do_send_packet(SrsRtpPacket* pkt);
     // Directly set the status of play track, generally for init to set the default value.
     void set_all_tracks_status(std::string stream_uri, bool is_publish, bool status);
 private:
@@ -579,7 +579,7 @@ class ISrsRtcHijacker
     // When stop publish by RTC.
     virtual void on_stop_publish(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req) = 0;
     // When got RTP plaintext packet.
-    virtual srs_error_t on_rtp_packet(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req, SrsRtpPacket2* pkt) = 0;
+    virtual srs_error_t on_rtp_packet(SrsRtcConnection* session, SrsRtcPublishStream* publisher, SrsRequest* req, SrsRtpPacket* pkt) = 0;
     // When before play by RTC. (wait source to ready in cascade scenario)
     virtual srs_error_t on_before_play(SrsRtcConnection* session, SrsRequest* req) = 0;
     // When start player by RTC.

trunk/src/app/srs_app_rtc_jitbuffer.hpp

@@ -44,7 +44,7 @@ class SrsRtpFrameBuffer;
 class SrsRtpDecodingState;
 class SrsGb28181RtmpMuxer;
 class VCMPacket;
-class SrsRtpPacket2;
+class SrsRtpPacket;
 
 ///jittbuffer
 

trunk/src/app/srs_app_rtc_queue.cpp

@@ -47,14 +47,14 @@ SrsRtpRingBuffer::SrsRtpRingBuffer(int capacity)
     capacity_ = (uint16_t)capacity;
     initialized_ = false;
 
-    queue_ = new SrsRtpPacket2*[capacity_];
-    memset(queue_, 0, sizeof(SrsRtpPacket2*) * capacity);
+    queue_ = new SrsRtpPacket*[capacity_];
+    memset(queue_, 0, sizeof(SrsRtpPacket*) * capacity);
 }
 
 SrsRtpRingBuffer::~SrsRtpRingBuffer()
 {
     for (int i = 0; i < capacity_; ++i) {
-        SrsRtpPacket2* pkt = queue_[i];
+        SrsRtpPacket* pkt = queue_[i];
         srs_freep(pkt);
     }
     srs_freepa(queue_);
@@ -77,9 +77,9 @@ void SrsRtpRingBuffer::advance_to(uint16_t seq)
     begin = seq;
 }
 
-void SrsRtpRingBuffer::set(uint16_t at, SrsRtpPacket2* pkt)
+void SrsRtpRingBuffer::set(uint16_t at, SrsRtpPacket* pkt)
 {
-    SrsRtpPacket2* p = queue_[at % capacity_];
+    SrsRtpPacket* p = queue_[at % capacity_];
     srs_freep(p);
 
     queue_[at % capacity_] = pkt;
@@ -143,7 +143,7 @@ bool SrsRtpRingBuffer::update(uint16_t seq, uint16_t& nack_first, uint16_t& nack
     return true;
 }
 
-SrsRtpPacket2* SrsRtpRingBuffer::at(uint16_t seq) {
+SrsRtpPacket* SrsRtpRingBuffer::at(uint16_t seq) {
     return queue_[seq % capacity_];
 }
 
@@ -165,7 +165,7 @@ void SrsRtpRingBuffer::clear_histroy(uint16_t seq)
 {
     // TODO FIXME Did not consider loopback
     for (uint16_t i = 0; i < capacity_; i++) {
-        SrsRtpPacket2* p = queue_[i];
+        SrsRtpPacket* p = queue_[i];
         if (p && p->header.get_sequence() < seq) {
             srs_freep(p);
             queue_[i] = NULL;
@@ -176,7 +176,7 @@ void SrsRtpRingBuffer::clear_histroy(uint16_t seq)
 void SrsRtpRingBuffer::clear_all_histroy()
 {
     for (uint16_t i = 0; i < capacity_; i++) {
-        SrsRtpPacket2* p = queue_[i];
+        SrsRtpPacket* p = queue_[i];
         if (p) {
             srs_freep(p);
             queue_[i] = NULL;

trunk/src/app/srs_app_rtc_queue.hpp

@@ -33,7 +33,7 @@
 #include <srs_kernel_rtc_rtp.hpp>
 #include <srs_kernel_rtc_rtcp.hpp>
 
-class SrsRtpPacket2;
+class SrsRtpPacket;
 class SrsRtpQueue;
 class SrsRtpRingBuffer;
 
@@ -51,7 +51,7 @@ class SrsRtpRingBuffer
     // Capacity of the ring-buffer.
     uint16_t capacity_;
     // Ring bufer.
-    SrsRtpPacket2** queue_;
+    SrsRtpPacket** queue_;
     // Increase one when uint16 flip back, for get_extended_highest_sequence.
     uint64_t nn_seq_flip_backs;
     // Whether initialized, because we use uint16 so we can't use -1.
@@ -74,15 +74,15 @@ class SrsRtpRingBuffer
     // Move the low position of buffer to seq.
     void advance_to(uint16_t seq);
     // Free the packet at position.
-    void set(uint16_t at, SrsRtpPacket2* pkt);
+    void set(uint16_t at, SrsRtpPacket* pkt);
     void remove(uint16_t at);
     // The highest sequence number, calculate the flip back base.
     uint32_t get_extended_highest_sequence();
     // Update the sequence, got the nack range by [first, last).
     // @return If false, the seq is too old.
     bool update(uint16_t seq, uint16_t& nack_first, uint16_t& nack_last);
     // Get the packet by seq.
-    SrsRtpPacket2* at(uint16_t seq);
+    SrsRtpPacket* at(uint16_t seq);
 public:
     // TODO: FIXME: Refine it?
     void notify_nack_list_full();