WebRtc_Word32 => int32_t for rtp_rtcp/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1279007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index 71d7940..1597ab2 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -159,7 +159,7 @@ rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport); // Make sure that RTCP objects are aware of our SSRC - WebRtc_UWord32 SSRC = rtp_sender_.SSRC(); + uint32_t SSRC = rtp_sender_.SSRC(); rtcp_sender_.SetSSRC(SSRC); rtcp_receiver_.SetSSRC(SSRC); @@ -228,14 +228,14 @@ // Returns the number of milliseconds until the module want a worker thread // to call Process. -WebRtc_Word32 ModuleRtpRtcpImpl::TimeUntilNextProcess() { - const WebRtc_Word64 now = clock_->TimeInMilliseconds(); +int32_t ModuleRtpRtcpImpl::TimeUntilNextProcess() { + const int64_t now = clock_->TimeInMilliseconds(); return kRtpRtcpMaxIdleTimeProcess - (now - last_process_time_); } // Process any pending tasks such as timeouts (non time critical events). -WebRtc_Word32 ModuleRtpRtcpImpl::Process() { - const WebRtc_Word64 now = clock_->TimeInMilliseconds(); +int32_t ModuleRtpRtcpImpl::Process() { + const int64_t now = clock_->TimeInMilliseconds(); last_process_time_ = now; if (now >= @@ -309,9 +309,8 @@ } void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() { - bool RTCPalive = false; - WebRtc_Word64 now = 0; + int64_t now = 0; bool do_callback = false; // Do operations on members under lock but avoid making the @@ -336,9 +335,9 @@ rtp_receiver_->ProcessDeadOrAlive(RTCPalive, now); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus( +int32_t ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus( const bool enable, - const WebRtc_UWord8 sample_time_seconds) { + const uint8_t sample_time_seconds) { if (enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -364,9 +363,9 @@ return 0; } -WebRtc_Word32 ModuleRtpRtcpImpl::PeriodicDeadOrAliveStatus( +int32_t ModuleRtpRtcpImpl::PeriodicDeadOrAliveStatus( bool& enable, - WebRtc_UWord8& sample_time_seconds) { + uint8_t& sample_time_seconds) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -374,13 +373,13 @@ enable = dead_or_alive_active_; sample_time_seconds = - static_cast<WebRtc_UWord8>(dead_or_alive_timeout_ms_ / 1000); + static_cast<uint8_t>(dead_or_alive_timeout_ms_ / 1000); return 0; } -WebRtc_Word32 ModuleRtpRtcpImpl::SetPacketTimeout( - const WebRtc_UWord32 rtp_timeout_ms, - const WebRtc_UWord32 rtcp_timeout_ms) { +int32_t ModuleRtpRtcpImpl::SetPacketTimeout( + const uint32_t rtp_timeout_ms, + const uint32_t rtcp_timeout_ms) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -394,7 +393,7 @@ return -1; } -WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload( +int32_t ModuleRtpRtcpImpl::RegisterReceivePayload( const CodecInst& voice_codec) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -409,7 +408,7 @@ (voice_codec.rate < 0) ? 0 : voice_codec.rate); } -WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload( +int32_t ModuleRtpRtcpImpl::RegisterReceivePayload( const VideoCodec& video_codec) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -423,9 +422,9 @@ video_codec.maxBitrate); } -WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType( +int32_t ModuleRtpRtcpImpl::ReceivePayloadType( const CodecInst& voice_codec, - WebRtc_Word8* pl_type) { + int8_t* pl_type) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -439,9 +438,9 @@ pl_type); } -WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType( +int32_t ModuleRtpRtcpImpl::ReceivePayloadType( const VideoCodec& video_codec, - WebRtc_Word8* pl_type) { + int8_t* pl_type) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -454,8 +453,8 @@ pl_type); } -WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterReceivePayload( - const WebRtc_Word8 payload_type) { +int32_t ModuleRtpRtcpImpl::DeRegisterReceivePayload( + const int8_t payload_type) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -466,17 +465,17 @@ } // Get the currently configured SSRC filter. -WebRtc_Word32 ModuleRtpRtcpImpl::SSRCFilter( - WebRtc_UWord32& allowed_ssrc) const { +int32_t ModuleRtpRtcpImpl::SSRCFilter( + uint32_t& allowed_ssrc) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SSRCFilter()"); return rtp_receiver_->SSRCFilter(allowed_ssrc); } // Set a SSRC to be used as a filter for incoming RTP streams. -WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRCFilter( +int32_t ModuleRtpRtcpImpl::SetSSRCFilter( const bool enable, - const WebRtc_UWord32 allowed_ssrc) { + const uint32_t allowed_ssrc) { if (enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -494,7 +493,7 @@ } // Get last received remote timestamp. -WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteTimestamp() const { +uint32_t ModuleRtpRtcpImpl::RemoteTimestamp() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteTimestamp()"); return rtp_receiver_->TimeStamp(); @@ -508,8 +507,8 @@ } // Get the current estimated remote timestamp. -WebRtc_Word32 ModuleRtpRtcpImpl::EstimatedRemoteTimeStamp( - WebRtc_UWord32& timestamp) const { +int32_t ModuleRtpRtcpImpl::EstimatedRemoteTimeStamp( + uint32_t& timestamp) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -519,51 +518,50 @@ } // Get incoming SSRC. -WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteSSRC() const { +uint32_t ModuleRtpRtcpImpl::RemoteSSRC() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteSSRC()"); return rtp_receiver_->SSRC(); } // Get remote CSRC -WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCSRCs( - WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const { +int32_t ModuleRtpRtcpImpl::RemoteCSRCs( + uint32_t arr_of_csrc[kRtpCsrcSize]) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteCSRCs()"); return rtp_receiver_->CSRCs(arr_of_csrc); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXSendStatus( +int32_t ModuleRtpRtcpImpl::SetRTXSendStatus( const RtxMode mode, const bool set_ssrc, - const WebRtc_UWord32 ssrc) { + const uint32_t ssrc) { rtp_sender_.SetRTXStatus(mode, set_ssrc, ssrc); return 0; } -WebRtc_Word32 ModuleRtpRtcpImpl::RTXSendStatus(RtxMode* mode, - WebRtc_UWord32* ssrc) const { +int32_t ModuleRtpRtcpImpl::RTXSendStatus(RtxMode* mode, uint32_t* ssrc) const { rtp_sender_.RTXStatus(mode, ssrc); return 0; } -WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXReceiveStatus( +int32_t ModuleRtpRtcpImpl::SetRTXReceiveStatus( const bool enable, - const WebRtc_UWord32 ssrc) { + const uint32_t ssrc) { rtp_receiver_->SetRTXStatus(enable, ssrc); return 0; } -WebRtc_Word32 ModuleRtpRtcpImpl::RTXReceiveStatus(bool* enable, - WebRtc_UWord32* ssrc) const { +int32_t ModuleRtpRtcpImpl::RTXReceiveStatus(bool* enable, + uint32_t* ssrc) const { rtp_receiver_->RTXStatus(enable, ssrc); return 0; } // Called by the network module when we receive a packet. -WebRtc_Word32 ModuleRtpRtcpImpl::IncomingPacket( - const WebRtc_UWord8* incoming_packet, - const WebRtc_UWord16 incoming_packet_length) { +int32_t ModuleRtpRtcpImpl::IncomingPacket( + const uint8_t* incoming_packet, + const uint16_t incoming_packet_length) { WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, @@ -579,7 +577,7 @@ return -1; } // Check RTP version. - const WebRtc_UWord8 version = incoming_packet[0] >> 6; + const uint8_t version = incoming_packet[0] >> 6; if (version != 2) { WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, @@ -606,7 +604,7 @@ return -1; } RTCPHelp::RTCPPacketInformation rtcp_packet_information; - WebRtc_Word32 ret_val = rtcp_receiver_.IncomingRTCPPacket( + int32_t ret_val = rtcp_receiver_.IncomingRTCPPacket( rtcp_packet_information, &rtcp_parser); if (ret_val == 0) { rtcp_receiver_.TriggerCallbacksFromRTCPPacket(rtcp_packet_information); @@ -634,7 +632,7 @@ } } -WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload( +int32_t ModuleRtpRtcpImpl::RegisterSendPayload( const CodecInst& voice_codec) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -652,7 +650,7 @@ (voice_codec.rate < 0) ? 0 : voice_codec.rate); } -WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload( +int32_t ModuleRtpRtcpImpl::RegisterSendPayload( const VideoCodec& video_codec) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -670,8 +668,8 @@ video_codec.maxBitrate); } -WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterSendPayload( - const WebRtc_Word8 payload_type) { +int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload( + const int8_t payload_type) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -680,19 +678,19 @@ return rtp_sender_.DeRegisterSendPayload(payload_type); } -WebRtc_Word8 ModuleRtpRtcpImpl::SendPayloadType() const { +int8_t ModuleRtpRtcpImpl::SendPayloadType() const { return rtp_sender_.SendPayloadType(); } -WebRtc_UWord32 ModuleRtpRtcpImpl::StartTimestamp() const { +uint32_t ModuleRtpRtcpImpl::StartTimestamp() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "StartTimestamp()"); return rtp_sender_.StartTimestamp(); } // Configure start timestamp, default is a random number. -WebRtc_Word32 ModuleRtpRtcpImpl::SetStartTimestamp( - const WebRtc_UWord32 timestamp) { +int32_t ModuleRtpRtcpImpl::SetStartTimestamp( + const uint32_t timestamp) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -703,15 +701,15 @@ return 0; // TODO(pwestin): change to void. } -WebRtc_UWord16 ModuleRtpRtcpImpl::SequenceNumber() const { +uint16_t ModuleRtpRtcpImpl::SequenceNumber() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SequenceNumber()"); return rtp_sender_.SequenceNumber(); } // Set SequenceNumber, default is a random number. -WebRtc_Word32 ModuleRtpRtcpImpl::SetSequenceNumber( - const WebRtc_UWord16 seq_num) { +int32_t ModuleRtpRtcpImpl::SetSequenceNumber( + const uint16_t seq_num) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -722,14 +720,14 @@ return 0; // TODO(pwestin): change to void. } -WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const { +uint32_t ModuleRtpRtcpImpl::SSRC() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SSRC()"); return rtp_sender_.SSRC(); } // Configure SSRC, default is a random number. -WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRC(const WebRtc_UWord32 ssrc) { +int32_t ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetSSRC(%d)", ssrc); rtp_sender_.SetSSRC(ssrc); @@ -738,22 +736,22 @@ return 0; // TODO(pwestin): change to void. } -WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) { +int32_t ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) { rtcp_sender_.SetCSRCStatus(include); rtp_sender_.SetCSRCStatus(include); return 0; // TODO(pwestin): change to void. } -WebRtc_Word32 ModuleRtpRtcpImpl::CSRCs( - WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const { +int32_t ModuleRtpRtcpImpl::CSRCs( + uint32_t arr_of_csrc[kRtpCsrcSize]) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "CSRCs()"); return rtp_sender_.CSRCs(arr_of_csrc); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCs( - const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize], - const WebRtc_UWord8 arr_length) { +int32_t ModuleRtpRtcpImpl::SetCSRCs( + const uint32_t arr_of_csrc[kRtpCsrcSize], + const uint8_t arr_length) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -785,13 +783,13 @@ return 0; // TODO(pwestin): change to void. } -WebRtc_UWord32 ModuleRtpRtcpImpl::PacketCountSent() const { +uint32_t ModuleRtpRtcpImpl::PacketCountSent() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "PacketCountSent()"); return rtp_sender_.Packets(); } -WebRtc_UWord32 ModuleRtpRtcpImpl::ByteCountSent() const { +uint32_t ModuleRtpRtcpImpl::ByteCountSent() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ByteCountSent()"); return rtp_sender_.Bytes(); @@ -804,7 +802,7 @@ return rtp_sender_.SendPayloadFrequency(); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { +int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { if (sending) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetSendingStatus(sending)"); @@ -831,7 +829,7 @@ // Make sure that RTCP objects are aware of our SSRC (it could have changed // Due to collision) - WebRtc_UWord32 SSRC = rtp_sender_.SSRC(); + uint32_t SSRC = rtp_sender_.SSRC(); rtcp_receiver_.SetSSRC(SSRC); rtcp_sender_.SetSSRC(SSRC); return 0; @@ -845,7 +843,7 @@ return rtcp_sender_.Sending(); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { +int32_t ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { if (sending) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetSendingMediaStatus(sending)"); @@ -877,13 +875,13 @@ return false; } -WebRtc_Word32 ModuleRtpRtcpImpl::SendOutgoingData( +int32_t ModuleRtpRtcpImpl::SendOutgoingData( FrameType frame_type, - WebRtc_Word8 payload_type, - WebRtc_UWord32 time_stamp, + int8_t payload_type, + uint32_t time_stamp, int64_t capture_time_ms, - const WebRtc_UWord8* payload_data, - WebRtc_UWord32 payload_size, + const uint8_t* payload_data, + uint32_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_hdr) { WEBRTC_TRACE( @@ -911,7 +909,7 @@ NULL, &(rtp_video_hdr->codecHeader)); } - WebRtc_Word32 ret_val = -1; + int32_t ret_val = -1; if (simulcast_) { if (rtp_video_hdr == NULL) { return -1; @@ -1025,20 +1023,20 @@ } } -WebRtc_UWord16 ModuleRtpRtcpImpl::MaxPayloadLength() const { +uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "MaxPayloadLength()"); return rtp_sender_.MaxPayloadLength(); } -WebRtc_UWord16 ModuleRtpRtcpImpl::MaxDataPayloadLength() const { +uint16_t ModuleRtpRtcpImpl::MaxDataPayloadLength() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "MaxDataPayloadLength()"); // Assuming IP/UDP. - WebRtc_UWord16 min_data_payload_length = IP_PACKET_SIZE - 28; + uint16_t min_data_payload_length = IP_PACKET_SIZE - 28; const bool default_instance(child_modules_.empty() ? false : true); if (default_instance) { @@ -1049,7 +1047,7 @@ while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { - WebRtc_UWord16 data_payload_length = + uint16_t data_payload_length = module->MaxDataPayloadLength(); if (data_payload_length < min_data_payload_length) { min_data_payload_length = data_payload_length; @@ -1059,17 +1057,17 @@ } } - WebRtc_UWord16 data_payload_length = rtp_sender_.MaxDataPayloadLength(); + uint16_t data_payload_length = rtp_sender_.MaxDataPayloadLength(); if (data_payload_length < min_data_payload_length) { min_data_payload_length = data_payload_length; } return min_data_payload_length; } -WebRtc_Word32 ModuleRtpRtcpImpl::SetTransportOverhead( +int32_t ModuleRtpRtcpImpl::SetTransportOverhead( const bool tcp, const bool ipv6, - const WebRtc_UWord8 authentication_overhead) { + const uint8_t authentication_overhead) { WEBRTC_TRACE( kTraceModuleCall, kTraceRtpRtcp, @@ -1077,7 +1075,7 @@ "SetTransportOverhead(TCP:%d, IPV6:%d authentication_overhead:%u)", tcp, ipv6, authentication_overhead); - WebRtc_UWord16 packet_overhead = 0; + uint16_t packet_overhead = 0; if (ipv6) { packet_overhead = 40; } else { @@ -1097,17 +1095,17 @@ return 0; } // Calc diff. - WebRtc_Word16 packet_over_head_diff = packet_overhead - packet_overhead_; + int16_t packet_over_head_diff = packet_overhead - packet_overhead_; // Store new. packet_overhead_ = packet_overhead; - WebRtc_UWord16 length = + uint16_t length = rtp_sender_.MaxPayloadLength() - packet_over_head_diff; return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetMaxTransferUnit(const WebRtc_UWord16 mtu) { +int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetMaxTransferUnit(%u)", mtu); @@ -1130,7 +1128,7 @@ } // Configure RTCP status i.e on/off. -WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) { +int32_t ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPStatus(%d)", method); @@ -1141,23 +1139,23 @@ } // Only for internal test. -WebRtc_UWord32 ModuleRtpRtcpImpl::LastSendReport( - WebRtc_UWord32& last_rtcptime) { +uint32_t ModuleRtpRtcpImpl::LastSendReport( + uint32_t& last_rtcptime) { return rtcp_sender_.LastSendReport(last_rtcptime); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetCNAME(const char c_name[RTCP_CNAME_SIZE]) { +int32_t ModuleRtpRtcpImpl::SetCNAME(const char c_name[RTCP_CNAME_SIZE]) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetCNAME(%s)", c_name); return rtcp_sender_.SetCNAME(c_name); } -WebRtc_Word32 ModuleRtpRtcpImpl::CNAME(char c_name[RTCP_CNAME_SIZE]) { +int32_t ModuleRtpRtcpImpl::CNAME(char c_name[RTCP_CNAME_SIZE]) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "CNAME()"); return rtcp_sender_.CNAME(c_name); } -WebRtc_Word32 ModuleRtpRtcpImpl::AddMixedCNAME( - const WebRtc_UWord32 ssrc, +int32_t ModuleRtpRtcpImpl::AddMixedCNAME( + const uint32_t ssrc, const char c_name[RTCP_CNAME_SIZE]) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "AddMixedCNAME(SSRC:%u)", ssrc); @@ -1165,14 +1163,14 @@ return rtcp_sender_.AddMixedCNAME(ssrc, c_name); } -WebRtc_Word32 ModuleRtpRtcpImpl::RemoveMixedCNAME(const WebRtc_UWord32 ssrc) { +int32_t ModuleRtpRtcpImpl::RemoveMixedCNAME(const uint32_t ssrc) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoveMixedCNAME(SSRC:%u)", ssrc); return rtcp_sender_.RemoveMixedCNAME(ssrc); } -WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCNAME( - const WebRtc_UWord32 remote_ssrc, +int32_t ModuleRtpRtcpImpl::RemoteCNAME( + const uint32_t remote_ssrc, char c_name[RTCP_CNAME_SIZE]) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteCNAME(SSRC:%u)", remote_ssrc); @@ -1180,18 +1178,18 @@ return rtcp_receiver_.CNAME(remote_ssrc, c_name); } -WebRtc_UWord16 ModuleRtpRtcpImpl::RemoteSequenceNumber() const { +uint16_t ModuleRtpRtcpImpl::RemoteSequenceNumber() const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteSequenceNumber()"); return rtp_receiver_->SequenceNumber(); } -WebRtc_Word32 ModuleRtpRtcpImpl::RemoteNTP( - WebRtc_UWord32* received_ntpsecs, - WebRtc_UWord32* received_ntpfrac, - WebRtc_UWord32* rtcp_arrival_time_secs, - WebRtc_UWord32* rtcp_arrival_time_frac, - WebRtc_UWord32* rtcp_timestamp) const { +int32_t ModuleRtpRtcpImpl::RemoteNTP( + uint32_t* received_ntpsecs, + uint32_t* received_ntpfrac, + uint32_t* rtcp_arrival_time_secs, + uint32_t* rtcp_arrival_time_frac, + uint32_t* rtcp_timestamp) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteNTP()"); return rtcp_receiver_.NTP(received_ntpsecs, @@ -1202,18 +1200,18 @@ } // Get RoundTripTime. -WebRtc_Word32 ModuleRtpRtcpImpl::RTT(const WebRtc_UWord32 remote_ssrc, - WebRtc_UWord16* rtt, - WebRtc_UWord16* avg_rtt, - WebRtc_UWord16* min_rtt, - WebRtc_UWord16* max_rtt) const { +int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc, + uint16_t* rtt, + uint16_t* avg_rtt, + uint16_t* min_rtt, + uint16_t* max_rtt) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RTT()"); return rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt); } // Reset RoundTripTime statistics. -WebRtc_Word32 ModuleRtpRtcpImpl::ResetRTT(const WebRtc_UWord32 remote_ssrc) { +int32_t ModuleRtpRtcpImpl::ResetRTT(const uint32_t remote_ssrc) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetRTT(SSRC:%u)", remote_ssrc); @@ -1222,18 +1220,18 @@ void ModuleRtpRtcpImpl:: SetRtt(uint32_t rtt) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRtt(rtt: %u)", rtt); - rtcp_receiver_.SetRTT(static_cast<WebRtc_UWord16>(rtt)); + rtcp_receiver_.SetRTT(static_cast<uint16_t>(rtt)); } // Reset RTP statistics. -WebRtc_Word32 ModuleRtpRtcpImpl::ResetStatisticsRTP() { +int32_t ModuleRtpRtcpImpl::ResetStatisticsRTP() { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetStatisticsRTP()"); return rtp_receiver_->ResetStatistics(); } // Reset RTP data counters for the receiving side. -WebRtc_Word32 ModuleRtpRtcpImpl::ResetReceiveDataCountersRTP() { +int32_t ModuleRtpRtcpImpl::ResetReceiveDataCountersRTP() { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetReceiveDataCountersRTP()"); @@ -1241,7 +1239,7 @@ } // Reset RTP data counters for the sending side. -WebRtc_Word32 ModuleRtpRtcpImpl::ResetSendDataCountersRTP() { +int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetSendDataCountersRTP()"); @@ -1251,18 +1249,18 @@ // Force a send of an RTCP packet. // Normal SR and RR are triggered via the process function. -WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCP(WebRtc_UWord32 rtcp_packet_type) { +int32_t ModuleRtpRtcpImpl::SendRTCP(uint32_t rtcp_packet_type) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendRTCP(0x%x)", rtcp_packet_type); return rtcp_sender_.SendRTCP(rtcp_packet_type); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData( - const WebRtc_UWord8 sub_type, - const WebRtc_UWord32 name, - const WebRtc_UWord8* data, - const WebRtc_UWord16 length) { +int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData( + const uint8_t sub_type, + const uint32_t name, + const uint8_t* data, + const uint16_t length) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPApplicationSpecificData(sub_type:%d name:0x%x)", sub_type, name); @@ -1271,7 +1269,7 @@ } // (XR) VOIP metric. -WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPVoIPMetrics( +int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics( const RTCPVoIPMetric* voip_metric) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPVoIPMetrics()"); @@ -1279,17 +1277,17 @@ } // Our locally created statistics of the received RTP stream. -WebRtc_Word32 ModuleRtpRtcpImpl::StatisticsRTP( - WebRtc_UWord8* fraction_lost, - WebRtc_UWord32* cum_lost, - WebRtc_UWord32* ext_max, - WebRtc_UWord32* jitter, - WebRtc_UWord32* max_jitter) const { +int32_t ModuleRtpRtcpImpl::StatisticsRTP( + uint8_t* fraction_lost, + uint32_t* cum_lost, + uint32_t* ext_max, + uint32_t* jitter, + uint32_t* max_jitter) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "StatisticsRTP()"); - WebRtc_UWord32 jitter_transmission_time_offset = 0; + uint32_t jitter_transmission_time_offset = 0; - WebRtc_Word32 ret_val = rtp_receiver_->Statistics( + int32_t ret_val = rtp_receiver_->Statistics( fraction_lost, cum_lost, ext_max, jitter, max_jitter, &jitter_transmission_time_offset, (rtcp_sender_.Status() == kRtcpOff)); if (ret_val == -1) { @@ -1299,11 +1297,11 @@ return ret_val; } -WebRtc_Word32 ModuleRtpRtcpImpl::DataCountersRTP( - WebRtc_UWord32* bytes_sent, - WebRtc_UWord32* packets_sent, - WebRtc_UWord32* bytes_received, - WebRtc_UWord32* packets_received) const { +int32_t ModuleRtpRtcpImpl::DataCountersRTP( + uint32_t* bytes_sent, + uint32_t* packets_sent, + uint32_t* bytes_received, + uint32_t* packets_received) const { WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, "DataCountersRTP()"); if (bytes_sent) { @@ -1315,21 +1313,21 @@ return rtp_receiver_->DataCounters(bytes_received, packets_received); } -WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics( - WebRtc_UWord8* fraction_lost, - WebRtc_UWord32* cum_lost, - WebRtc_UWord32* ext_max, - WebRtc_UWord32* jitter, - WebRtc_UWord32* jitter_transmission_time_offset) { +int32_t ModuleRtpRtcpImpl::ReportBlockStatistics( + uint8_t* fraction_lost, + uint32_t* cum_lost, + uint32_t* ext_max, + uint32_t* jitter, + uint32_t* jitter_transmission_time_offset) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ReportBlockStatistics()"); - WebRtc_Word32 missing = 0; - WebRtc_Word32 ret = rtp_receiver_->Statistics(fraction_lost, - cum_lost, - ext_max, - jitter, - NULL, - jitter_transmission_time_offset, - &missing, + int32_t missing = 0; + int32_t ret = rtp_receiver_->Statistics(fraction_lost, + cum_lost, + ext_max, + jitter, + NULL, + jitter_transmission_time_offset, + &missing, true); #ifdef MATLAB @@ -1344,30 +1342,30 @@ return ret; } -WebRtc_Word32 ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) { +int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteRTCPStat()"); return rtcp_receiver_.SenderInfoReceived(sender_info); } // Received RTCP report. -WebRtc_Word32 ModuleRtpRtcpImpl::RemoteRTCPStat( +int32_t ModuleRtpRtcpImpl::RemoteRTCPStat( std::vector<RTCPReportBlock>* receive_blocks) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteRTCPStat()"); return rtcp_receiver_.StatisticsReceived(receive_blocks); } -WebRtc_Word32 ModuleRtpRtcpImpl::AddRTCPReportBlock( - const WebRtc_UWord32 ssrc, +int32_t ModuleRtpRtcpImpl::AddRTCPReportBlock( + const uint32_t ssrc, const RTCPReportBlock* report_block) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "AddRTCPReportBlock()"); return rtcp_sender_.AddReportBlock(ssrc, report_block); } -WebRtc_Word32 ModuleRtpRtcpImpl::RemoveRTCPReportBlock( - const WebRtc_UWord32 ssrc) { +int32_t ModuleRtpRtcpImpl::RemoveRTCPReportBlock( + const uint32_t ssrc) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoveRTCPReportBlock()"); return rtcp_sender_.RemoveReportBlock(ssrc); @@ -1380,7 +1378,7 @@ return rtcp_sender_.REMB(); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) { +int32_t ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) { if (enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1395,9 +1393,9 @@ return rtcp_sender_.SetREMBStatus(enable); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetREMBData(const WebRtc_UWord32 bitrate, - const WebRtc_UWord8 number_of_ssrc, - const WebRtc_UWord32* ssrc) { +int32_t ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate, + const uint8_t number_of_ssrc, + const uint32_t* ssrc) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetREMBData(bitrate:%d,?,?)", bitrate); return rtcp_sender_.SetREMBData(bitrate, number_of_ssrc, ssrc); @@ -1410,7 +1408,7 @@ return rtcp_sender_.IJ(); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetIJStatus(const bool enable) { +int32_t ModuleRtpRtcpImpl::SetIJStatus(const bool enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -1419,24 +1417,24 @@ return rtcp_sender_.SetIJStatus(enable); } -WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension( +int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension( const RTPExtensionType type, - const WebRtc_UWord8 id) { + const uint8_t id) { return rtp_sender_.RegisterRtpHeaderExtension(type, id); } -WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension( +int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension( const RTPExtensionType type) { return rtp_sender_.DeregisterRtpHeaderExtension(type); } -WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceiveRtpHeaderExtension( +int32_t ModuleRtpRtcpImpl::RegisterReceiveRtpHeaderExtension( const RTPExtensionType type, - const WebRtc_UWord8 id) { + const uint8_t id) { return rtp_receiver_->RegisterRtpHeaderExtension(type, id); } -WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterReceiveRtpHeaderExtension( +int32_t ModuleRtpRtcpImpl::DeregisterReceiveRtpHeaderExtension( const RTPExtensionType type) { return rtp_receiver_->DeregisterRtpHeaderExtension(type); } @@ -1448,7 +1446,7 @@ return rtcp_sender_.TMMBR(); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) { +int32_t ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) { if (enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetTMMBRStatus(enable)"); @@ -1459,10 +1457,10 @@ return rtcp_sender_.SetTMMBRStatus(enable); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) { +int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetTMMBN()"); - WebRtc_UWord32 max_bitrate_kbit = + uint32_t max_bitrate_kbit = rtp_sender_.MaxConfiguredBitrateVideo() / 1000; return rtcp_sender_.SetTMMBN(bounding_set, max_bitrate_kbit); } @@ -1501,7 +1499,7 @@ } // Turn negative acknowledgment requests on/off. -WebRtc_Word32 ModuleRtpRtcpImpl::SetNACKStatus( +int32_t ModuleRtpRtcpImpl::SetNACKStatus( NACKMethod method, int max_reordering_threshold) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1534,24 +1532,24 @@ } // Send a Negative acknowledgment packet. -WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nack_list, - const WebRtc_UWord16 size) { +int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list, + const uint16_t size) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendNACK(size:%u)", size); - WebRtc_UWord16 avg_rtt = 0; + uint16_t avg_rtt = 0; rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL); - WebRtc_Word64 wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5. + int64_t wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5. if (wait_time == 5) { wait_time = 100; // During startup we don't have an RTT. } - const WebRtc_Word64 now = clock_->TimeInMilliseconds(); - const WebRtc_Word64 time_limit = now - wait_time; - WebRtc_UWord16 nackLength = size; - WebRtc_UWord16 start_id = 0; + const int64_t now = clock_->TimeInMilliseconds(); + const int64_t time_limit = now - wait_time; + uint16_t nackLength = size; + uint16_t start_id = 0; if (nack_last_time_sent_full_ < time_limit) { // Send list. Set the timer to make sure we only send a full NACK list once @@ -1592,9 +1590,9 @@ // Store the sent packets, needed to answer to a Negative acknowledgment // requests. -WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus( +int32_t ModuleRtpRtcpImpl::SetStorePacketsStatus( const bool enable, - const WebRtc_UWord16 number_to_store) { + const uint16_t number_to_store) { if (enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetStorePacketsStatus(enable, number_to_store:%d)", @@ -1631,10 +1629,10 @@ } // Send a TelephoneEvent tone using RFC 2833 (4733). -WebRtc_Word32 ModuleRtpRtcpImpl::SendTelephoneEventOutband( - const WebRtc_UWord8 key, - const WebRtc_UWord16 time_ms, - const WebRtc_UWord8 level) { +int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband( + const uint8_t key, + const uint16_t time_ms, + const uint8_t level) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendTelephoneEventOutband(key:%u, time_ms:%u, level:%u)", key, time_ms, level); @@ -1643,7 +1641,7 @@ } bool ModuleRtpRtcpImpl::SendTelephoneEventActive( - WebRtc_Word8& telephone_event) const { + int8_t& telephone_event) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1655,8 +1653,8 @@ // Set audio packet size, used to determine when it's time to send a DTMF // packet in silence (CNG). -WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioPacketSize( - const WebRtc_UWord16 packet_size_samples) { +int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( + const uint16_t packet_size_samples) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1667,9 +1665,9 @@ return rtp_sender_.SetAudioPacketSize(packet_size_samples); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus( +int32_t ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus( const bool enable, - const WebRtc_UWord8 id) { + const uint8_t id) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1686,9 +1684,9 @@ return rtp_sender_.SetAudioLevelIndicationStatus(enable, id); } -WebRtc_Word32 ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus( +int32_t ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus( bool& enable, - WebRtc_UWord8& id) const { + uint8_t& id) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1697,8 +1695,8 @@ return rtp_sender_.AudioLevelIndicationStatus(&enable, &id); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioLevel( - const WebRtc_UWord8 level_d_bov) { +int32_t ModuleRtpRtcpImpl::SetAudioLevel( + const uint8_t level_d_bov) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -1708,8 +1706,8 @@ } // Set payload type for Redundant Audio Data RFC 2198. -WebRtc_Word32 ModuleRtpRtcpImpl::SetSendREDPayloadType( - const WebRtc_Word8 payload_type) { +int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType( + const int8_t payload_type) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -1720,8 +1718,8 @@ } // Get payload type for Redundant Audio Data RFC 2198. -WebRtc_Word32 ModuleRtpRtcpImpl::SendREDPayloadType( - WebRtc_Word8& payload_type) const { +int32_t ModuleRtpRtcpImpl::SendREDPayloadType( + int8_t& payload_type) const { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendREDPayloadType()"); return rtp_sender_.RED(&payload_type); @@ -1774,7 +1772,7 @@ } } -WebRtc_Word32 ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( +int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( const KeyFrameRequestMethod method) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1786,7 +1784,7 @@ return 0; } -WebRtc_Word32 ModuleRtpRtcpImpl::RequestKeyFrame() { +int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -1803,8 +1801,8 @@ return -1; } -WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCPSliceLossIndication( - const WebRtc_UWord8 picture_id) { +int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication( + const uint8_t picture_id) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -1813,7 +1811,7 @@ return rtcp_sender_.SendRTCP(kRtcpSli, 0, 0, false, picture_id); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetCameraDelay(const WebRtc_Word32 delay_ms) { +int32_t ModuleRtpRtcpImpl::SetCameraDelay(const int32_t delay_ms) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, @@ -1837,10 +1835,10 @@ return rtcp_sender_.SetCameraDelay(delay_ms); } -WebRtc_Word32 ModuleRtpRtcpImpl::SetGenericFECStatus( +int32_t ModuleRtpRtcpImpl::SetGenericFECStatus( const bool enable, - const WebRtc_UWord8 payload_type_red, - const WebRtc_UWord8 payload_type_fec) { + const uint8_t payload_type_red, + const uint8_t payload_type_fec) { if (enable) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, @@ -1858,10 +1856,10 @@ payload_type_fec); } -WebRtc_Word32 ModuleRtpRtcpImpl::GenericFECStatus( +int32_t ModuleRtpRtcpImpl::GenericFECStatus( bool& enable, - WebRtc_UWord8& payload_type_red, - WebRtc_UWord8& payload_type_fec) { + uint8_t& payload_type_red, + uint8_t& payload_type_fec) { WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "GenericFECStatus()"); @@ -1875,8 +1873,8 @@ RtpRtcp* module = *it; if (module) { bool enabled = false; - WebRtc_UWord8 dummy_ptype_red = 0; - WebRtc_UWord8 dummy_ptype_fec = 0; + uint8_t dummy_ptype_red = 0; + uint8_t dummy_ptype_fec = 0; if (module->GenericFECStatus(enabled, dummy_ptype_red, dummy_ptype_fec) == 0 && enabled) { @@ -1887,9 +1885,9 @@ it++; } } - WebRtc_Word32 ret_val = rtp_sender_.GenericFECStatus(&enable, - &payload_type_red, - &payload_type_fec); + int32_t ret_val = rtp_sender_.GenericFECStatus(&enable, + &payload_type_red, + &payload_type_fec); if (child_enabled) { // Returns true if enabled for any child module. enable = child_enabled; @@ -1897,7 +1895,7 @@ return ret_val; } -WebRtc_Word32 ModuleRtpRtcpImpl::SetFecParameters( +int32_t ModuleRtpRtcpImpl::SetFecParameters( const FecProtectionParams* delta_params, const FecProtectionParams* key_params) { const bool default_instance(child_modules_.empty() ? false : true); @@ -1918,7 +1916,7 @@ return rtp_sender_.SetFecParameters(delta_params, key_params); } -void ModuleRtpRtcpImpl::SetRemoteSSRC(const WebRtc_UWord32 ssrc) { +void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) { // Inform about the incoming SSRC. rtcp_sender_.SetRemoteSSRC(ssrc); rtcp_receiver_.SetRemoteSSRC(ssrc); @@ -1927,7 +1925,7 @@ if (rtp_sender_.SSRC() == ssrc && !collision_detected_) { // If we detect a collision change the SSRC but only once. collision_detected_ = true; - WebRtc_UWord32 new_ssrc = rtp_sender_.GenerateNewSSRC(); + uint32_t new_ssrc = rtp_sender_.GenerateNewSSRC(); if (new_ssrc == 0) { // Configured via API ignore. return; @@ -1942,14 +1940,14 @@ } } -WebRtc_UWord32 ModuleRtpRtcpImpl::BitrateReceivedNow() const { +uint32_t ModuleRtpRtcpImpl::BitrateReceivedNow() const { return rtp_receiver_->BitrateNow(); } -void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* total_rate, - WebRtc_UWord32* video_rate, - WebRtc_UWord32* fec_rate, - WebRtc_UWord32* nack_rate) const { +void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, + uint32_t* video_rate, + uint32_t* fec_rate, + uint32_t* nack_rate) const { const bool default_instance(child_modules_.empty() ? false : true); if (default_instance) { @@ -1970,10 +1968,10 @@ while (it != child_modules_.end()) { RtpRtcp* module = *it; if (module) { - WebRtc_UWord32 child_total_rate = 0; - WebRtc_UWord32 child_video_rate = 0; - WebRtc_UWord32 child_fec_rate = 0; - WebRtc_UWord32 child_nack_rate = 0; + uint32_t child_total_rate = 0; + uint32_t child_video_rate = 0; + uint32_t child_fec_rate = 0; + uint32_t child_nack_rate = 0; module->BitrateSent(&child_total_rate, &child_video_rate, &child_fec_rate, @@ -2010,13 +2008,13 @@ rtcp_sender_.SendRTCP(kRtcpSr); } -WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection( - const WebRtc_UWord64 picture_id) { +int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection( + const uint64_t picture_id) { return rtcp_sender_.SendRTCP(kRtcpRpsi, 0, 0, false, picture_id); } -WebRtc_UWord32 ModuleRtpRtcpImpl::SendTimeOfSendReport( - const WebRtc_UWord32 send_report) { +uint32_t ModuleRtpRtcpImpl::SendTimeOfSendReport( + const uint32_t send_report) { return rtcp_sender_.SendTimeOfSendReport(send_report); } @@ -2026,18 +2024,18 @@ nack_sequence_numbers.size() == 0) { return; } - WebRtc_UWord16 avg_rtt = 0; + uint16_t avg_rtt = 0; rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL); rtp_sender_.OnReceivedNACK(nack_sequence_numbers, avg_rtt); } -WebRtc_Word32 ModuleRtpRtcpImpl::LastReceivedNTP( - WebRtc_UWord32& rtcp_arrival_time_secs, // When we got the last report. - WebRtc_UWord32& rtcp_arrival_time_frac, - WebRtc_UWord32& remote_sr) { +int32_t ModuleRtpRtcpImpl::LastReceivedNTP( + uint32_t& rtcp_arrival_time_secs, // When we got the last report. + uint32_t& rtcp_arrival_time_frac, + uint32_t& remote_sr) { // Remote SR: NTP inside the last received (mid 16 bits from sec and frac). - WebRtc_UWord32 ntp_secs = 0; - WebRtc_UWord32 ntp_frac = 0; + uint32_t ntp_secs = 0; + uint32_t ntp_frac = 0; if (-1 == rtcp_receiver_.NTP(&ntp_secs, &ntp_frac, @@ -2057,8 +2055,8 @@ } // Called from RTCPsender. -WebRtc_Word32 ModuleRtpRtcpImpl::BoundingSet(bool& tmmbr_owner, - TMMBRSet*& bounding_set) { +int32_t ModuleRtpRtcpImpl::BoundingSet(bool& tmmbr_owner, + TMMBRSet*& bounding_set) { return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set); }