[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1513303003
Cr-Commit-Position: refs/heads/master@{#11025}
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 334d811..796be13 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -112,7 +112,10 @@
MOCK_CONST_METHOD0(SendingMedia,
bool());
MOCK_CONST_METHOD4(BitrateSent,
- void(uint32_t* totalRate, uint32_t* videoRate, uint32_t* fecRate, uint32_t* nackRate));
+ void(uint32_t* totalRate,
+ uint32_t* videoRate,
+ uint32_t* fecRate,
+ uint32_t* nackRate));
MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver*));
MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver*(void));
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
@@ -176,7 +179,10 @@
MOCK_CONST_METHOD1(RemoteRTCPStat,
int32_t(std::vector<RTCPReportBlock>* receiveBlocks));
MOCK_METHOD4(SetRTCPApplicationSpecificData,
- int32_t(const uint8_t subType, const uint32_t name, const uint8_t* data, const uint16_t length));
+ int32_t(const uint8_t subType,
+ const uint32_t name,
+ const uint8_t* data,
+ const uint16_t length));
MOCK_METHOD1(SetRTCPVoIPMetrics,
int32_t(const RTCPVoIPMetric* VoIPMetric));
MOCK_METHOD1(SetRtcpXrRrtrStatus,
@@ -221,8 +227,6 @@
MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payloadType));
MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
int32_t(const bool enable, const uint8_t ID));
- MOCK_CONST_METHOD2(GetRTPAudioLevelIndicationStatus,
- int32_t(bool& enable, uint8_t& ID));
MOCK_METHOD1(SetAudioLevel,
int32_t(const uint8_t level_dBov));
MOCK_METHOD1(SetTargetSendBitrate,
@@ -242,8 +246,6 @@
int32_t(const KeyFrameRequestMethod method));
MOCK_METHOD0(RequestKeyFrame,
int32_t());
- MOCK_CONST_METHOD3(Version,
- int32_t(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position));
MOCK_METHOD0(TimeUntilNextProcess,
int64_t());
MOCK_METHOD0(Process,
diff --git a/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
new file mode 100644
index 0000000..7edbb96
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
@@ -0,0 +1,8 @@
+#tmmbr_help is refactored in CL#1474693002
+exclude_files=tmmbr_help.*
+#rtp_utility is refactored in CL#1481773004
+exclude_files=rtp_utility.*
+#rtcp_utility planned to be removed when webrtc:5260 will be finished.
+exclude_files=rtcp_utility.*
+#rtcp_receiver/rtcp_receiver_help will be refactored more deeply as part of webrtc:5260
+exclude_files=rtcp_receiver.*
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index b79f6ba..6a63813 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -25,7 +25,7 @@
class FecReceiverImpl : public FecReceiver {
public:
- FecReceiverImpl(RtpData* callback);
+ explicit FecReceiverImpl(RtpData* callback);
virtual ~FecReceiverImpl();
int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index e7d8fd7..e19c31b 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -53,7 +53,7 @@
class TestRtpFeedback : public NullRtpFeedback {
public:
- TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
+ explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
virtual ~TestRtpFeedback() {}
void OnIncomingSSRCChanged(const uint32_t ssrc) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index a639f7f..87c0259 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -26,7 +26,8 @@
class TestTransport : public Transport {
public:
- TestTransport(RTCPReceiver* rtcp_receiver) : rtcp_receiver_(rtcp_receiver) {}
+ explicit TestTransport(RTCPReceiver* rtcp_receiver)
+ : rtcp_receiver_(rtcp_receiver) {}
bool SendRtp(const uint8_t* /*data*/,
size_t /*len*/,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index e172139..d65b04c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -746,7 +746,7 @@
return updateBoundingSet;
}
-int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
+int32_t RTCPReceiver::BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec) {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
std::map<uint32_t, RTCPReceiveInformation*>::iterator receiveInfoIt =
@@ -766,7 +766,7 @@
i++) {
if(receiveInfo->TmmbnBoundingSet.Ssrc(i) == main_ssrc_) {
// owner of bounding set
- tmmbrOwner = true;
+ *tmmbrOwner = true;
}
boundingSetRec->SetEntry(i,
receiveInfo->TmmbnBoundingSet.Tmmbr(i),
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index a22fd81..24861bd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -103,7 +103,7 @@
bool UpdateRTCPReceiveInformationTimers();
- int32_t BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec);
+ int32_t BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec);
int32_t UpdateTMMBR();
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index ea62fcb..15f9388 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -645,7 +645,7 @@
// will accuire criticalSectionRTCPReceiver_ is a potental deadlock but
// since RTCPreceiver is not doing the reverse we should be fine
int32_t lengthOfBoundingSet =
- ctx.feedback_state_.module->BoundingSet(tmmbrOwner, candidateSet);
+ ctx.feedback_state_.module->BoundingSet(&tmmbrOwner, candidateSet);
if (lengthOfBoundingSet > 0) {
for (int32_t i = 0; i < lengthOfBoundingSet; i++) {
@@ -1077,7 +1077,7 @@
bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
public:
- Sender(Transport* transport)
+ explicit Sender(Transport* transport)
: transport_(transport), send_failure_(false) {}
void OnPacketReady(uint8_t* data, size_t length) override {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 37966de..342e38a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -28,7 +28,7 @@
const size_t kTransportSequenceNumberLength = 3;
struct HeaderExtension {
- HeaderExtension(RTPExtensionType extension_type)
+ explicit HeaderExtension(RTPExtensionType extension_type)
: type(extension_type), length(0), active(true) {
Init();
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 3d4c09a..8e1a732 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -29,7 +29,7 @@
class RTPPacketHistory {
public:
- RTPPacketHistory(Clock* clock);
+ explicit RTPPacketHistory(Clock* clock);
~RTPPacketHistory();
void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index bdc05c6..fb9b206 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -170,7 +170,7 @@
int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
bool is_red = false;
- if (CheckPayloadChanged(rtp_header, first_payload_byte, is_red,
+ if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
&payload_specific) == -1) {
if (payload_length == 0) {
// OK, keep-alive packet.
@@ -320,7 +320,7 @@
// last known payload).
int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
const int8_t first_payload_byte,
- bool& is_red,
+ bool* is_red,
PayloadUnion* specific_payload) {
bool re_initialize_decoder = false;
@@ -338,7 +338,7 @@
if (rtp_payload_registry_->red_payload_type() == payload_type) {
// Get the real codec payload type.
payload_type = first_payload_byte & 0x7f;
- is_red = true;
+ *is_red = true;
if (rtp_payload_registry_->red_payload_type() == payload_type) {
// Invalid payload type, traced by caller. If we proceeded here,
@@ -360,7 +360,7 @@
&should_discard_changes);
if (should_discard_changes) {
- is_red = false;
+ *is_red = false;
return 0;
}
@@ -390,7 +390,7 @@
}
} else {
rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
- is_red = false;
+ *is_red = false;
}
} // End critsect.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 4ec3531..eaa07d9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -71,7 +71,7 @@
void CheckCSRC(const WebRtcRTPHeader& rtp_header);
int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
const int8_t first_payload_byte,
- bool& is_red,
+ bool* is_red,
PayloadUnion* payload);
Clock* clock_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 0e0a34a..0f7ad30 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -95,7 +95,7 @@
// Note: Implementations may call the callback for other reasons than calls
// to ParseRtpPacket, for instance if the implementation somehow recovers a
// packet.
- RTPReceiverStrategy(RtpData* data_callback);
+ explicit RTPReceiverStrategy(RtpData* data_callback);
rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
PayloadUnion last_payload_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 30b1b96..450eed6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -956,8 +956,8 @@
}
// Called from RTCPsender.
-int32_t 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);
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 24cbbe4..04e09c1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -295,7 +295,7 @@
bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
- virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
+ int32_t BoundingSet(bool* tmmbr_owner, TMMBRSet* bounding_set_rec);
void BitrateSent(uint32_t* total_rate,
uint32_t* video_rate,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index f975ede..405a78d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -97,7 +97,7 @@
class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
public:
- RtpRtcpModule(SimulatedClock* clock)
+ explicit RtpRtcpModule(SimulatedClock* clock)
: receive_statistics_(ReceiveStatistics::Create(clock)) {
RtpRtcp::Configuration config;
config.audio = false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 4e91a29..17fbbac 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -326,11 +326,11 @@
return -1;
}
int32_t ret_val = 0;
- RtpUtility::Payload* payload = NULL;
+ RtpUtility::Payload* payload = nullptr;
if (audio_configured_) {
// TODO(mflodman): Change to CreateAudioPayload and make static.
ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
- frequency, channels, rate, payload);
+ frequency, channels, rate, &payload);
} else {
payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
}
@@ -455,7 +455,7 @@
}
if (audio_configured_) {
int8_t red_pl_type = -1;
- if (audio_->RED(red_pl_type) == 0) {
+ if (audio_->RED(&red_pl_type) == 0) {
// We have configured RED.
if (red_pl_type == payload_type) {
// And it's a match...
@@ -1001,7 +1001,7 @@
bool fec_enabled;
uint8_t pt_red;
uint8_t pt_fec;
- video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
+ video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
return fec_enabled &&
header.payloadType == pt_red &&
buffer[header.headerLength] == pt_fec;
@@ -1778,7 +1778,7 @@
if (!audio_configured_) {
return -1;
}
- return audio_->RED(*payload_type);
+ return audio_->RED(payload_type);
}
RtpVideoCodecTypes RTPSender::VideoCodecType() const {
@@ -1804,7 +1804,7 @@
uint8_t* payload_type_red,
uint8_t* payload_type_fec) const {
RTC_DCHECK(!audio_configured_);
- video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec);
+ video_->GenericFECStatus(enable, payload_type_red, payload_type_fec);
}
int32_t RTPSender::SetFecParameters(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 3bc861c..86407f9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -68,7 +68,7 @@
const uint32_t frequency,
const uint8_t channels,
const uint32_t rate,
- RtpUtility::Payload*& payload) {
+ RtpUtility::Payload** payload) {
if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
CriticalSectionScoped cs(_sendAudioCritsect.get());
// we can have multiple CNG payload types
@@ -96,13 +96,13 @@
return 0;
// The default timestamp rate is 8000 Hz, but other rates may be defined.
}
- payload = new RtpUtility::Payload;
- payload->typeSpecific.Audio.frequency = frequency;
- payload->typeSpecific.Audio.channels = channels;
- payload->typeSpecific.Audio.rate = rate;
- payload->audio = true;
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ *payload = new RtpUtility::Payload;
+ (*payload)->typeSpecific.Audio.frequency = frequency;
+ (*payload)->typeSpecific.Audio.channels = channels;
+ (*payload)->typeSpecific.Audio.rate = rate;
+ (*payload)->audio = true;
+ (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
+ strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
return 0;
}
@@ -384,13 +384,13 @@
}
// Get payload type for Redundant Audio Data RFC 2198
-int32_t RTPSenderAudio::RED(int8_t& payloadType) const {
+int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
CriticalSectionScoped cs(_sendAudioCritsect.get());
if (_REDPayloadType == -1) {
// not configured
return -1;
}
- payloadType = _REDPayloadType;
+ *payloadType = _REDPayloadType;
return 0;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 381bc13..a3cee5e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -31,7 +31,7 @@
uint32_t frequency,
uint8_t channels,
uint32_t rate,
- RtpUtility::Payload*& payload);
+ RtpUtility::Payload** payload);
int32_t SendAudio(FrameType frameType,
int8_t payloadType,
@@ -58,7 +58,7 @@
int32_t SetRED(int8_t payloadType);
// Get payload type for Redundant Audio Data RFC 2198
- int32_t RED(int8_t& payloadType) const;
+ int32_t RED(int8_t* payloadType) const;
protected:
int32_t SendTelephoneEventPacket(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 80967b2..1ca7831 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1281,9 +1281,9 @@
// For Telephone events, payload is not added to the registered payload list,
// it will register only the payload used for audio stream.
// Registering the payload again for audio stream with different payload name.
- strcpy(payload_name, "payload_name");
+ const char kPayloadName[] = "payload_name";
ASSERT_EQ(
- 0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000, 1, 0));
+ 0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
// DTMF event key=9, duration=500 and attenuationdB=10
rtp_sender_->SendTelephoneEvent(9, 500, 10);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index bec3623..e4604f9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -190,13 +190,13 @@
kFecMaskRandom;
}
-void RTPSenderVideo::GenericFECStatus(bool& enable,
- uint8_t& payloadTypeRED,
- uint8_t& payloadTypeFEC) const {
+void RTPSenderVideo::GenericFECStatus(bool* enable,
+ uint8_t* payloadTypeRED,
+ uint8_t* payloadTypeFEC) const {
CriticalSectionScoped cs(crit_.get());
- enable = fec_enabled_;
- payloadTypeRED = red_payload_type_;
- payloadTypeFEC = fec_payload_type_;
+ *enable = fec_enabled_;
+ *payloadTypeRED = red_payload_type_;
+ *payloadTypeFEC = fec_payload_type_;
}
size_t RTPSenderVideo::FECPacketOverhead() const {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 465240e..e59321a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -67,9 +67,9 @@
const uint8_t payloadTypeRED,
const uint8_t payloadTypeFEC);
- void GenericFECStatus(bool& enable,
- uint8_t& payloadTypeRED,
- uint8_t& payloadTypeFEC) const;
+ void GenericFECStatus(bool* enable,
+ uint8_t* payloadTypeRED,
+ uint8_t* payloadTypeFEC) const;
void SetFecParameters(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params);
