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RTC: Refine stream/ssrc/sdp structure
This commit is contained in:
winlin
parent
426938cc8a
commit
991672bf41
11 changed files with 2204 additions and 563 deletions
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@ -57,6 +57,8 @@ class SrsRtpNackForReceiver;
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class SrsRtpIncommingVideoFrame;
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class SrsRtpRingBuffer;
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class SrsRtcConsumer;
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class SrsRtcAudioSendTrack;
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class SrsRtcVideoSendTrack;
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const uint8_t kSR = 200;
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const uint8_t kRR = 201;
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@ -75,23 +77,6 @@ const uint8_t kSLI = 2;
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const uint8_t kRPSI = 3;
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const uint8_t kAFB = 15;
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class SrsNtp
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{
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public:
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uint64_t system_ms_;
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uint64_t ntp_;
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uint32_t ntp_second_;
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uint32_t ntp_fractions_;
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public:
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SrsNtp();
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virtual ~SrsNtp();
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public:
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static SrsNtp from_time_ms(uint64_t ms);
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static SrsNtp to_time_ms(uint64_t ntp);
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public:
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static uint64_t kMagicNtpFractionalUnit;
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};
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enum SrsRtcConnectionStateType
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{
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// TODO: FIXME: Should prefixed by enum name.
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@ -187,16 +172,9 @@ protected:
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SrsCoroutine* trd;
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SrsRtcConnection* session_;
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private:
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// TODO: FIXME: How to handle timestamp overflow?
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// Information for audio.
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uint32_t audio_ssrc;
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uint16_t audio_payload_type;
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// Information for video.
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uint16_t video_payload_type;
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uint32_t video_ssrc;
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// NACK ARQ ring buffer.
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SrsRtpRingBuffer* audio_queue_;
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SrsRtpRingBuffer* video_queue_;
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// key: publish_ssrc, value: send track to process rtp/rtcp
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std::map<uint32_t, SrsRtcAudioSendTrack*> audio_tracks_;
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std::map<uint32_t, SrsRtcVideoSendTrack*> video_tracks_;
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// Simulators.
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int nn_simulate_nack_drop;
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private:
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@ -205,11 +183,15 @@ private:
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bool realtime;
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// Whether enabled nack.
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bool nack_enabled_;
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// Whether palyer started.
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bool is_started;
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// statistic send packets.
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SrsRtcOutgoingInfo info;
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public:
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SrsRtcPlayStream(SrsRtcConnection* s, SrsContextId parent_cid);
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virtual ~SrsRtcPlayStream();
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public:
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srs_error_t initialize(uint32_t vssrc, uint32_t assrc, uint16_t v_pt, uint16_t a_pt);
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srs_error_t initialize(SrsRequest* request, std::map<uint32_t, SrsRtcTrackDescription*> sub_relations);
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// interface ISrsReloadHandler
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public:
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virtual srs_error_t on_reload_vhost_play(std::string vhost);
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@ -238,6 +220,7 @@ private:
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srs_error_t on_rtcp_feedback(char* data, int nb_data);
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srs_error_t on_rtcp_ps_feedback(char* data, int nb_data);
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srs_error_t on_rtcp_rr(char* data, int nb_data);
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uint32_t get_video_publish_ssrc(uint32_t play_ssrc);
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};
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// A RTC publish stream, client push and publish stream to SRS.
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@ -248,48 +231,41 @@ private:
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uint64_t nn_audio_frames;
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private:
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SrsRtcConnection* session_;
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uint32_t video_ssrc;
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uint32_t audio_ssrc;
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uint16_t pt_to_drop_;
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// Whether enabled nack.
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bool nack_enabled_;
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private:
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bool request_keyframe_;
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SrsRtpRingBuffer* video_queue_;
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SrsRtpNackForReceiver* video_nack_;
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SrsRtpRingBuffer* audio_queue_;
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SrsRtpNackForReceiver* audio_nack_;
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private:
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SrsRequest* req;
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SrsRtcStream* source;
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// Simulators.
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int nn_simulate_nack_drop;
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private:
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std::map<uint32_t, uint64_t> last_sender_report_sys_time;
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std::map<uint32_t, SrsNtp> last_sender_report_ntp;
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// track vector
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std::vector<SrsRtcAudioRecvTrack*> audio_tracks_;
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std::vector<SrsRtcVideoRecvTrack*> video_tracks_;
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private:
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srs_utime_t last_twcc_feedback_time_;
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int twcc_id_;
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uint8_t twcc_fb_count_;
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SrsRtcpTWCC rtcp_twcc_;
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SrsRtpExtensionTypes extension_types_;
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bool is_started;
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public:
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SrsRtcPublishStream(SrsRtcConnection* session);
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virtual ~SrsRtcPublishStream();
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public:
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srs_error_t initialize(uint32_t vssrc, uint32_t assrc, int twcc_id, SrsRequest* req);
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srs_error_t initialize(SrsRequest* req, SrsRtcStreamDescription* stream_desc);
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srs_error_t start();
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private:
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void check_send_nacks(SrsRtpNackForReceiver* nack, uint32_t ssrc);
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srs_error_t send_rtcp_rr(uint32_t ssrc, SrsRtpRingBuffer* rtp_queue);
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srs_error_t send_rtcp_xr_rrtr(uint32_t ssrc);
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srs_error_t send_rtcp_fb_pli(uint32_t ssrc);
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srs_error_t send_rtcp_rr();
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srs_error_t send_rtcp_xr_rrtr();
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public:
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srs_error_t on_rtp(char* buf, int nb_buf);
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virtual void on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload);
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private:
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srs_error_t on_audio(SrsRtpPacket2* pkt);
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srs_error_t on_video(SrsRtpPacket2* pkt);
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srs_error_t on_nack(SrsRtpPacket2* pkt);
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srs_error_t send_periodic_twcc();
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public:
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srs_error_t on_rtcp(char* data, int nb_data);
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@ -300,7 +276,7 @@ private:
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srs_error_t on_rtcp_ps_feedback(char* data, int nb_data);
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srs_error_t on_rtcp_rr(char* data, int nb_data);
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public:
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void request_keyframe();
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void request_keyframe(uint32_t ssrc);
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// interface ISrsHourGlass
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public:
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virtual srs_error_t notify(int type, srs_utime_t interval, srs_utime_t tick);
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@ -310,6 +286,10 @@ private:
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void simulate_drop_packet(SrsRtpHeader* h, int nn_bytes);
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private:
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srs_error_t on_twcc(uint16_t sn);
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SrsRtcAudioRecvTrack* get_audio_track(uint32_t ssrc);
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SrsRtcVideoRecvTrack* get_video_track(uint32_t ssrc);
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void update_rtt(uint32_t ssrc, int rtt);
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void update_send_report_time(uint32_t ssrc, const SrsNtp& ntp);
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};
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// A RTC Peer Connection, SDP level object.
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@ -320,6 +300,8 @@ class SrsRtcConnection
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friend class SrsRtcPublishStream;
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public:
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bool disposing_;
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private:
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static uint32_t ssrc_num;
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private:
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SrsRtcServer* server_;
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SrsRtcConnectionStateType state_;
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@ -360,6 +342,7 @@ public:
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SrsRtcConnection(SrsRtcServer* s);
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virtual ~SrsRtcConnection();
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public:
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// TODO: FIXME: save only connection info.
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SrsSdp* get_local_sdp();
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void set_local_sdp(const SrsSdp& sdp);
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SrsSdp* get_remote_sdp();
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@ -373,6 +356,8 @@ public:
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void set_encrypt(bool v);
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void switch_to_context();
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SrsContextId context_id();
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srs_error_t add_publisher(SrsRequest* request, const SrsSdp& remote_sdp, SrsSdp& local_sdp);
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srs_error_t add_player(SrsRequest* request, const SrsSdp& remote_sdp, SrsSdp& local_sdp);
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public:
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// Before initialize, user must set the local SDP, which is used to inititlize DTLS.
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srs_error_t initialize(SrsRtcStream* source, SrsRequest* r, bool is_publisher, std::string username, SrsContextId context_id);
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@ -387,11 +372,25 @@ public:
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srs_error_t start_publish();
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bool is_stun_timeout();
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void update_sendonly_socket(SrsUdpMuxSocket* skt);
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public:
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// send rtcp
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void check_send_nacks(SrsRtpNackForReceiver* nack, uint32_t ssrc);
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srs_error_t send_rtcp_rr(uint32_t ssrc, SrsRtpRingBuffer* rtp_queue, const uint64_t& last_send_systime, const SrsNtp& last_send_ntp);
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srs_error_t send_rtcp_xr_rrtr(uint32_t ssrc);
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srs_error_t send_rtcp_fb_pli(uint32_t ssrc);
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public:
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// Simulate the NACK to drop nn packets.
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void simulate_nack_drop(int nn);
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private:
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srs_error_t on_binding_request(SrsStunPacket* r);
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// publish media capabilitiy negotiate
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srs_error_t negotiate_publish_capability(SrsRequest* req, const SrsSdp& remote_sdp, SrsRtcStreamDescription* stream_desc);
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srs_error_t generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcStreamDescription* stream_desc);
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// play media capabilitiy negotiate
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srs_error_t negotiate_play_capability(SrsRequest* req, const SrsSdp& remote_sdp, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations);
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srs_error_t generate_play_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcStreamDescription* stream_desc);
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srs_error_t create_player(SrsRequest* request, std::map<uint32_t, SrsRtcTrackDescription*> sub_relations);
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srs_error_t create_publisher(SrsRequest* request, SrsRtcStreamDescription* stream_desc);
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};
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class ISrsRtcHijacker
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