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);
}
