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https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
修复编译警告
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415bc95dda
commit
08e74256da
@ -267,7 +267,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
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void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
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void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
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if (_srtp_session_send) {
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if (_srtp_session_send) {
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//预留rtx加入的两个字节
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//预留rtx加入的两个字节
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CHECK(len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf));
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CHECK((size_t)len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf));
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memcpy(_srtp_buf, buf, len);
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memcpy(_srtp_buf, buf, len);
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onBeforeEncryptRtp((char *) _srtp_buf, len, ctx);
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onBeforeEncryptRtp((char *) _srtp_buf, len, ctx);
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if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
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if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
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@ -278,7 +278,7 @@ void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *
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void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx){
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void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx){
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if (_srtp_session_send) {
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if (_srtp_session_send) {
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CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
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CHECK((size_t)len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
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memcpy(_srtp_buf, buf, len);
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memcpy(_srtp_buf, buf, len);
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onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx);
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onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx);
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if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
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if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
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@ -442,7 +442,7 @@ void WebRtcTransportImp::onStartWebRTC() {
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InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
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InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
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});
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});
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int index = 0;
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size_t index = 0;
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for (auto &ssrc : m_offer->rtp_ssrc_sim) {
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for (auto &ssrc : m_offer->rtp_ssrc_sim) {
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//记录ssrc对应的MediaTrack
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//记录ssrc对应的MediaTrack
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_ssrc_to_track[ssrc.ssrc] = track;
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_ssrc_to_track[ssrc.ssrc] = track;
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