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RTC: Rename SrsRtcOutgoingInfo to SrsRtcPlayStreamStatistic
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parent
0131472ae5
commit
bf3b917f1f
4 changed files with 25 additions and 25 deletions
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@ -196,7 +196,7 @@ srs_error_t SrsSecurityTransport::unprotect_rtcp(const char* cipher, char* plain
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return srtp_->unprotect_rtcp(cipher, plaintext, nb_plaintext);
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return srtp_->unprotect_rtcp(cipher, plaintext, nb_plaintext);
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}
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}
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SrsRtcOutgoingInfo::SrsRtcOutgoingInfo()
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SrsRtcPlayStreamStatistic::SrsRtcPlayStreamStatistic()
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{
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{
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#if defined(SRS_DEBUG)
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#if defined(SRS_DEBUG)
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debug_id = 0;
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debug_id = 0;
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@ -209,7 +209,7 @@ SrsRtcOutgoingInfo::SrsRtcOutgoingInfo()
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nn_padding_bytes = nn_paddings = 0;
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nn_padding_bytes = nn_paddings = 0;
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}
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}
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SrsRtcOutgoingInfo::~SrsRtcOutgoingInfo()
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SrsRtcPlayStreamStatistic::~SrsRtcPlayStreamStatistic()
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{
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{
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}
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}
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@ -425,7 +425,7 @@ srs_error_t SrsRtcPlayStream::cycle()
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}
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}
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}
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}
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srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingInfo& info)
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srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info)
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{
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{
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srs_error_t err = srs_success;
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srs_error_t err = srs_success;
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@ -1555,7 +1555,7 @@ SrsRtcConnection::SrsRtcConnection(SrsRtcServer* s, SrsContextId context_id)
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state_ = INIT;
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state_ = INIT;
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last_stun_time = 0;
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last_stun_time = 0;
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sessionStunTimeout = 0;
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session_timeout = 0;
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disposing_ = false;
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disposing_ = false;
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twcc_id_ = 0;
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twcc_id_ = 0;
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@ -1757,11 +1757,11 @@ srs_error_t SrsRtcConnection::initialize(SrsRtcStream* source, SrsRequest* r, bo
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}
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}
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// TODO: FIXME: Support reload.
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// TODO: FIXME: Support reload.
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sessionStunTimeout = _srs_config->get_rtc_stun_timeout(req->vhost);
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session_timeout = _srs_config->get_rtc_stun_timeout(req->vhost);
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last_stun_time = srs_get_system_time();
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last_stun_time = srs_get_system_time();
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srs_trace("RTC init session, DTLS(role=%s, version=%s), timeout=%dms",
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srs_trace("RTC init session, DTLS(role=%s, version=%s), timeout=%dms",
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cfg->dtls_role.c_str(), cfg->dtls_version.c_str(), srsu2msi(sessionStunTimeout));
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cfg->dtls_role.c_str(), cfg->dtls_version.c_str(), srsu2msi(session_timeout));
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return err;
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return err;
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}
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}
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@ -1852,7 +1852,7 @@ srs_error_t SrsRtcConnection::on_connection_established()
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srs_error_t err = srs_success;
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srs_error_t err = srs_success;
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srs_trace("RTC %s session=%s, to=%dms connection established", (is_publisher_? "Publisher":"Subscriber"),
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srs_trace("RTC %s session=%s, to=%dms connection established", (is_publisher_? "Publisher":"Subscriber"),
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id().c_str(), srsu2msi(sessionStunTimeout));
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id().c_str(), srsu2msi(session_timeout));
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if (is_publisher_) {
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if (is_publisher_) {
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if ((err = start_publish()) != srs_success) {
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if ((err = start_publish()) != srs_success) {
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@ -1891,7 +1891,7 @@ srs_error_t SrsRtcConnection::start_publish()
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bool SrsRtcConnection::is_stun_timeout()
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bool SrsRtcConnection::is_stun_timeout()
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{
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{
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return last_stun_time + sessionStunTimeout < srs_get_system_time();
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return last_stun_time + session_timeout < srs_get_system_time();
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}
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}
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// TODO: FIXME: We should support multiple addresses, because client may use more than one addresses.
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// TODO: FIXME: We should support multiple addresses, because client may use more than one addresses.
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@ -2135,7 +2135,7 @@ void SrsRtcConnection::simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes
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nn_simulate_player_nack_drop--;
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nn_simulate_player_nack_drop--;
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}
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}
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srs_error_t SrsRtcConnection::do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingInfo& info)
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srs_error_t SrsRtcConnection::do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info)
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{
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{
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srs_error_t err = srs_success;
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srs_error_t err = srs_success;
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@ -88,6 +88,7 @@ enum SrsRtcConnectionStateType
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CLOSED = 5,
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CLOSED = 5,
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};
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};
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// The security transport, use DTLS/SRTP to protect the data.
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class SrsSecurityTransport : public ISrsDtlsCallback
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class SrsSecurityTransport : public ISrsDtlsCallback
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{
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{
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private:
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private:
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@ -127,8 +128,7 @@ private:
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};
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};
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// A group of RTP packets for outgoing(send to players).
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// A group of RTP packets for outgoing(send to players).
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// TODO: FIXME: Rename to stat for RTP packets.
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class SrsRtcPlayStreamStatistic
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class SrsRtcOutgoingInfo
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{
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{
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public:
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public:
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#if defined(SRS_DEBUG)
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#if defined(SRS_DEBUG)
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@ -160,8 +160,8 @@ public:
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// The number of padded packet.
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// The number of padded packet.
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int nn_paddings;
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int nn_paddings;
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public:
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public:
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SrsRtcOutgoingInfo();
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SrsRtcPlayStreamStatistic();
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virtual ~SrsRtcOutgoingInfo();
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virtual ~SrsRtcPlayStreamStatistic();
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};
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};
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// A RTC play stream, client pull and play stream from SRS.
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// A RTC play stream, client pull and play stream from SRS.
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@ -184,8 +184,8 @@ private:
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private:
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private:
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// Whether palyer started.
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// Whether palyer started.
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bool is_started;
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bool is_started;
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// statistic send packets.
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// The statistic for consumer to send packets to player.
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SrsRtcOutgoingInfo info;
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SrsRtcPlayStreamStatistic info;
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public:
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public:
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SrsRtcPlayStream(SrsRtcConnection* s, SrsContextId parent_cid);
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SrsRtcPlayStream(SrsRtcConnection* s, SrsContextId parent_cid);
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virtual ~SrsRtcPlayStream();
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virtual ~SrsRtcPlayStream();
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@ -205,7 +205,7 @@ public:
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public:
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public:
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virtual srs_error_t cycle();
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virtual srs_error_t cycle();
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private:
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private:
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srs_error_t send_packets(SrsRtcStream* source, const std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingInfo& info);
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srs_error_t send_packets(SrsRtcStream* source, const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info);
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public:
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public:
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void nack_fetch(std::vector<SrsRtpPacket2*>& pkts, uint32_t ssrc, uint16_t seq);
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void nack_fetch(std::vector<SrsRtpPacket2*>& pkts, uint32_t ssrc, uint16_t seq);
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public:
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public:
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@ -328,7 +328,7 @@ private:
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std::string peer_id_;
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std::string peer_id_;
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private:
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private:
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// The timeout of session, keep alive by STUN ping pong.
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// The timeout of session, keep alive by STUN ping pong.
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srs_utime_t sessionStunTimeout;
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srs_utime_t session_timeout;
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srs_utime_t last_stun_time;
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srs_utime_t last_stun_time;
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private:
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private:
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// For each RTC session, we use a specified cid for debugging logs.
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// For each RTC session, we use a specified cid for debugging logs.
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@ -404,7 +404,7 @@ public:
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// Simulate the NACK to drop nn packets.
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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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void simulate_nack_drop(int nn);
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void simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes);
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void simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes);
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srs_error_t do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingInfo& info);
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srs_error_t do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info);
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private:
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private:
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srs_error_t on_binding_request(SrsStunPacket* r);
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srs_error_t on_binding_request(SrsStunPacket* r);
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// publish media capabilitiy negotiate
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// publish media capabilitiy negotiate
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@ -1728,7 +1728,7 @@ std::string SrsRtcSendTrack::get_track_id()
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return track_desc_->id_;
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return track_desc_->id_;
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}
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}
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srs_error_t SrsRtcSendTrack::on_rtp(SrsRtpPacket2* pkt, SrsRtcOutgoingInfo& info)
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srs_error_t SrsRtcSendTrack::on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info)
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{
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{
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return srs_success;
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return srs_success;
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}
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}
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@ -1747,7 +1747,7 @@ SrsRtcAudioSendTrack::~SrsRtcAudioSendTrack()
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{
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{
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}
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}
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srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket2* pkt, SrsRtcOutgoingInfo& info)
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srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info)
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{
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{
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srs_error_t err = srs_success;
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srs_error_t err = srs_success;
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@ -1794,7 +1794,7 @@ SrsRtcVideoSendTrack::~SrsRtcVideoSendTrack()
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{
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{
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}
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}
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srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket2* pkt, SrsRtcOutgoingInfo& info)
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srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info)
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{
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{
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srs_error_t err = srs_success;
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srs_error_t err = srs_success;
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@ -54,7 +54,7 @@ class SrsRtcConnection;
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class SrsRtpRingBuffer;
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class SrsRtpRingBuffer;
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class SrsRtpNackForReceiver;
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class SrsRtpNackForReceiver;
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class SrsJsonObject;
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class SrsJsonObject;
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class SrsRtcOutgoingInfo;
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class SrsRtcPlayStreamStatistic;
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class SrsNtp
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class SrsNtp
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{
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{
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@ -471,7 +471,7 @@ public:
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void set_track_status(bool active);
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void set_track_status(bool active);
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std::string get_track_id();
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std::string get_track_id();
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public:
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public:
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virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcOutgoingInfo& info);
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virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info);
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virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
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virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
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};
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};
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@ -481,7 +481,7 @@ public:
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SrsRtcAudioSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
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SrsRtcAudioSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
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virtual ~SrsRtcAudioSendTrack();
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virtual ~SrsRtcAudioSendTrack();
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public:
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public:
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virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcOutgoingInfo& info);
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virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info);
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virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
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virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
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};
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};
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@ -491,7 +491,7 @@ public:
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SrsRtcVideoSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
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SrsRtcVideoSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc);
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virtual ~SrsRtcVideoSendTrack();
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virtual ~SrsRtcVideoSendTrack();
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public:
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public:
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virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcOutgoingInfo& info);
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virtual srs_error_t on_rtp(SrsRtpPacket2* pkt, SrsRtcPlayStreamStatistic& info);
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virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
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virtual srs_error_t on_rtcp(SrsRtpPacket2* pkt);
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};
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};
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