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修复webrtc srtp相关功能在大端64位系统崩溃的问题

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This commit is contained in:
ziyue
2021-08-13 16:07:27 +08:00
parent 8fd9853bc9
commit 1d161bcdc2
4 changed files with 19 additions and 19 deletions
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8
webrtc/WebRtcTransport.cpp
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@@ -236,7 +236,7 @@ bool is_rtcp(char *buf) {
return ((header->pt >= 64) && (header->pt < 96));
}
void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple) {
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len);
if (packet == nullptr) {
@@ -264,7 +264,7 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
}
}
void WebRtcTransport::sendRtpPacket(const char *buf, size_t len, bool flush, void *ctx) {
void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
if (_srtp_session_send) {
//预留rtx加入的两个字节
CHECK(len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf));
@@ -276,7 +276,7 @@ void WebRtcTransport::sendRtpPacket(const char *buf, size_t len, bool flush, voi
}
}
void WebRtcTransport::sendRtcpPacket(const char *buf, size_t len, bool flush, void *ctx){
void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx){
if (_srtp_session_send) {
CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
memcpy(_srtp_buf, buf, len);
@@ -901,7 +901,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
}
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) {
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx) {
auto pr = (pair<bool/*rtx*/, MediaTrack *> *) ctx;
auto header = (RtpHeader *) buf;