From f3e4ceee47d747c8868d919c179ecc640b9541f0 Mon Sep 17 00:00:00 2001 From: "pbos@webrtc.org" Date: Wed, 31 Jul 2013 15:17:19 +0000 Subject: [PATCH] Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ BUG=163 R=pwestin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1904005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4444 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../rtp_rtcp/interface/rtp_rtcp_defines.h | 16 +- .../source/forward_error_correction.cc | 16 ++ .../source/forward_error_correction.h | 15 +- webrtc/modules/rtp_rtcp/source/rtcp_sender.cc | 2 + webrtc/modules/rtp_rtcp/source/rtcp_sender.h | 2 + .../modules/rtp_rtcp/source/rtp_format_vp8.cc | 2 + .../modules/rtp_rtcp/source/rtp_format_vp8.h | 2 + .../rtp_rtcp/source/rtp_header_parser.cc | 7 +- .../rtp_rtcp/source/rtp_payload_registry.cc | 28 +-- .../rtp_rtcp/source/rtp_receiver_audio.cc | 2 + .../rtp_rtcp/source/rtp_receiver_audio.h | 24 +- .../rtp_rtcp/source/rtp_receiver_strategy.cc | 9 + .../rtp_rtcp/source/rtp_receiver_strategy.h | 6 +- .../rtp_rtcp/source/rtp_receiver_video.h | 19 +- .../modules/rtp_rtcp/source/rtp_rtcp_impl.h | 224 +++++++++--------- webrtc/modules/rtp_rtcp/source/rtp_sender.h | 23 +- 16 files changed, 218 insertions(+), 179 deletions(-) diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h index 500ab53a3..6f1fa39e7 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h @@ -264,24 +264,24 @@ class NullRtpFeedback : public RtpFeedback { const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int frequency, const uint8_t channels, - const uint32_t rate) { + const uint32_t rate) OVERRIDE { return 0; } - virtual void OnPacketTimeout(const int32_t id) {} + virtual void OnPacketTimeout(const int32_t id) OVERRIDE {} virtual void OnReceivedPacket(const int32_t id, - const RtpRtcpPacketType packetType) {} + const RtpRtcpPacketType packetType) OVERRIDE {} virtual void OnPeriodicDeadOrAlive(const int32_t id, - const RTPAliveType alive) {} + const RTPAliveType alive) OVERRIDE {} virtual void OnIncomingSSRCChanged(const int32_t id, - const uint32_t SSRC) {} + const uint32_t SSRC) OVERRIDE {} virtual void OnIncomingCSRCChanged(const int32_t id, const uint32_t CSRC, - const bool added) {} + const bool added) OVERRIDE {} }; // Null object version of RtpData. @@ -291,7 +291,7 @@ class NullRtpData : public RtpData { virtual int32_t OnReceivedPayloadData( const uint8_t* payloadData, const uint16_t payloadSize, - const WebRtcRTPHeader* rtpHeader) { + const WebRtcRTPHeader* rtpHeader) OVERRIDE { return 0; } }; @@ -304,7 +304,7 @@ class NullRtpAudioFeedback : public RtpAudioFeedback { virtual void OnPlayTelephoneEvent(const int32_t id, const uint8_t event, const uint16_t lengthMs, - const uint8_t volume) {} + const uint8_t volume) OVERRIDE {} }; } // namespace webrtc diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc index b88507de0..d2635e8f6 100644 --- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc +++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc @@ -40,6 +40,16 @@ enum { kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets }; +int32_t ForwardErrorCorrection::Packet::AddRef() { return ++ref_count_; } + +int32_t ForwardErrorCorrection::Packet::Release() { + int32_t ref_count; + ref_count = --ref_count_; + if (ref_count == 0) + delete this; + return ref_count; +} + // Used to link media packets to their protecting FEC packets. // // TODO(holmer): Refactor into a proper class. @@ -66,6 +76,12 @@ bool ForwardErrorCorrection::SortablePacket::LessThan( return IsNewerSequenceNumber(second->seq_num, first->seq_num); } +ForwardErrorCorrection::ReceivedPacket::ReceivedPacket() {} +ForwardErrorCorrection::ReceivedPacket::~ReceivedPacket() {} + +ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() {} +ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() {} + ForwardErrorCorrection::ForwardErrorCorrection(int32_t id) : id_(id), generated_fec_packets_(kMaxMediaPackets), diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h index 1e7695fc4..8910fe477 100644 --- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h +++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h @@ -43,16 +43,11 @@ class ForwardErrorCorrection { virtual ~Packet() {} // Add a reference. - virtual int32_t AddRef() { return ++ref_count_; } + virtual int32_t AddRef(); // Release a reference. Will delete the object if the reference count // reaches zero. - virtual int32_t Release() { - int32_t ref_count; - ref_count = --ref_count_; - if (ref_count == 0) delete this; - return ref_count; - } + virtual int32_t Release(); uint16_t length; // Length of packet in bytes. uint8_t data[IP_PACKET_SIZE]; // Packet data. @@ -90,6 +85,9 @@ class ForwardErrorCorrection { // TODO(holmer): Refactor into a proper class. class ReceivedPacket : public SortablePacket { public: + ReceivedPacket(); + ~ReceivedPacket(); + uint32_t ssrc; // SSRC of the current frame. Must be set for FEC // packets, but not required for media packets. bool is_fec; // Set to true if this is an FEC packet and false @@ -102,6 +100,9 @@ class ForwardErrorCorrection { // TODO(holmer): Refactor into a proper class. class RecoveredPacket : public SortablePacket { public: + RecoveredPacket(); + ~RecoveredPacket(); + bool was_recovered; // Will be true if this packet was recovered by // the FEC. Otherwise it was a media packet passed in // through the received packet list. diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index 55dbd0e85..c7bd5ac6a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc @@ -31,6 +31,8 @@ NACKStringBuilder::NACKStringBuilder() : // Empty. } +NACKStringBuilder::~NACKStringBuilder() {} + void NACKStringBuilder::PushNACK(uint16_t nack) { if (_count == 0) diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h index 9be2f3f9b..5a78f577d 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h @@ -32,6 +32,8 @@ class NACKStringBuilder { public: NACKStringBuilder(); + ~NACKStringBuilder(); + void PushNACK(uint16_t nack); std::string GetResult(); diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc index ec94fae12..4eb4cd6b9 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc @@ -67,6 +67,8 @@ RtpFormatVp8::RtpFormatVp8(const uint8_t* payload_data, part_info_.fragmentationOffset[0] = 0; } +RtpFormatVp8::~RtpFormatVp8() {} + int RtpFormatVp8::NextPacket(uint8_t* buffer, int* bytes_to_send, bool* last_packet) { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h index ae6ae93ec..650c0fad5 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h @@ -62,6 +62,8 @@ class RtpFormatVp8 { const RTPVideoHeaderVP8& hdr_info, int max_payload_len); + ~RtpFormatVp8(); + // Get the next payload with VP8 payload header. // max_payload_len limits the sum length of payload and VP8 payload header. // buffer is a pointer to where the output will be written. diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc index 6fcc1f18a..d04872582 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc @@ -23,11 +23,12 @@ class RtpHeaderParserImpl : public RtpHeaderParser { virtual ~RtpHeaderParserImpl() {} virtual bool Parse(const uint8_t* packet, int length, - RTPHeader* header) const; + RTPHeader* header) const OVERRIDE; - virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); + virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, + uint8_t id) OVERRIDE; - virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type); + virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) OVERRIDE; private: scoped_ptr critical_section_; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc index cb689ee42..783a11339 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -253,13 +253,13 @@ bool RTPPayloadRegistry::ReportMediaPayloadType( class RTPPayloadAudioStrategy : public RTPPayloadStrategy { public: - bool CodecsMustBeUnique() const { return true; } + virtual bool CodecsMustBeUnique() const OVERRIDE { return true; } - bool PayloadIsCompatible( + virtual bool PayloadIsCompatible( const ModuleRTPUtility::Payload& payload, const uint32_t frequency, const uint8_t channels, - const uint32_t rate) const { + const uint32_t rate) const OVERRIDE { return payload.audio && payload.typeSpecific.Audio.frequency == frequency && @@ -268,18 +268,18 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { payload.typeSpecific.Audio.rate == 0 || rate == 0); } - void UpdatePayloadRate( + virtual void UpdatePayloadRate( ModuleRTPUtility::Payload* payload, - const uint32_t rate) const { + const uint32_t rate) const OVERRIDE { payload->typeSpecific.Audio.rate = rate; } - ModuleRTPUtility::Payload* CreatePayloadType( + virtual ModuleRTPUtility::Payload* CreatePayloadType( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, const uint8_t channels, - const uint32_t rate) const { + const uint32_t rate) const OVERRIDE { ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); @@ -293,28 +293,28 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { class RTPPayloadVideoStrategy : public RTPPayloadStrategy { public: - bool CodecsMustBeUnique() const { return false; } + virtual bool CodecsMustBeUnique() const OVERRIDE { return false; } - bool PayloadIsCompatible( + virtual bool PayloadIsCompatible( const ModuleRTPUtility::Payload& payload, const uint32_t frequency, const uint8_t channels, - const uint32_t rate) const { + const uint32_t rate) const OVERRIDE { return !payload.audio; } - void UpdatePayloadRate( + virtual void UpdatePayloadRate( ModuleRTPUtility::Payload* payload, - const uint32_t rate) const { + const uint32_t rate) const OVERRIDE { payload->typeSpecific.Video.maxRate = rate; } - ModuleRTPUtility::Payload* CreatePayloadType( + virtual ModuleRTPUtility::Payload* CreatePayloadType( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, const uint8_t channels, - const uint32_t rate) const { + const uint32_t rate) const OVERRIDE { RtpVideoCodecTypes videoType = kRtpGenericVideo; if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) { videoType = kRtpVp8Video; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc index dd49ec596..130ca8a70 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -41,6 +41,8 @@ RTPReceiverAudio::RTPReceiverAudio(const int32_t id, last_payload_.Audio.channels = 1; } +RTPReceiverAudio::~RTPReceiverAudio() {} + uint32_t RTPReceiverAudio::AudioFrequency() const { CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get()); if (last_received_g722_) { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h index 67b30c020..9ac99c23c 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -30,6 +30,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy { RTPReceiverAudio(const int32_t id, RtpData* data_callback, RtpAudioFeedback* incoming_messages_callback); + virtual ~RTPReceiverAudio(); uint32_t AudioFrequency() const; @@ -48,32 +49,33 @@ class RTPReceiverAudio : public RTPReceiverStrategy { uint32_t* frequency, bool* cng_payload_type_has_changed); - int32_t ParseRtpPacket( + virtual int32_t ParseRtpPacket( WebRtcRTPHeader* rtp_header, const ModuleRTPUtility::PayloadUnion& specific_payload, const bool is_red, const uint8_t* packet, const uint16_t packet_length, const int64_t timestamp_ms, - const bool is_first_packet); + const bool is_first_packet) OVERRIDE; - int32_t GetFrequencyHz() const; + virtual int32_t GetFrequencyHz() const OVERRIDE; - RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const; + virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const + OVERRIDE; - bool ShouldReportCsrcChanges(uint8_t payload_type) const; + virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE; - int32_t OnNewPayloadTypeCreated( + virtual int32_t OnNewPayloadTypeCreated( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const int8_t payload_type, - const uint32_t frequency); + const uint32_t frequency) OVERRIDE; - int32_t InvokeOnInitializeDecoder( + virtual int32_t InvokeOnInitializeDecoder( RtpFeedback* callback, const int32_t id, const int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], - const ModuleRTPUtility::PayloadUnion& specific_payload) const; + const ModuleRTPUtility::PayloadUnion& specific_payload) const OVERRIDE; // We do not allow codecs to have multiple payload types for audio, so we // need to override the default behavior (which is to do nothing). @@ -87,10 +89,10 @@ class RTPReceiverAudio : public RTPReceiverStrategy { // We need to look out for special payload types here and sometimes reset // statistics. In addition we sometimes need to tweak the frequency. - void CheckPayloadChanged(const int8_t payload_type, + virtual void CheckPayloadChanged(const int8_t payload_type, ModuleRTPUtility::PayloadUnion* specific_payload, bool* should_reset_statistics, - bool* should_discard_changes); + bool* should_discard_changes) OVERRIDE; private: diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc index 8831e1273..509d4fe0a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc @@ -29,4 +29,13 @@ void RTPReceiverStrategy::SetLastMediaSpecificPayload( memcpy(&last_payload_, &payload, sizeof(last_payload_)); } +void RTPReceiverStrategy::CheckPayloadChanged( + const int8_t payload_type, + ModuleRTPUtility::PayloadUnion* specific_payload, + bool* should_reset_statistics, + bool* should_discard_changes) { + // Default: Keep changes and don't reset statistics. + *should_discard_changes = false; + *should_reset_statistics = false; +} } // namespace webrtc diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h index 8e8fa1d0d..be1002099 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -80,11 +80,7 @@ class RTPReceiverStrategy { const int8_t payload_type, ModuleRTPUtility::PayloadUnion* specific_payload, bool* should_reset_statistics, - bool* should_discard_changes) { - // Default: Keep changes and don't reset statistics. - *should_discard_changes = false; - *should_reset_statistics = false; - } + bool* should_discard_changes); // Stores / retrieves the last media specific payload for later reference. void GetLastMediaSpecificPayload( diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h index 520e20126..f2d2193a5 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h @@ -33,32 +33,33 @@ class RTPReceiverVideo : public RTPReceiverStrategy { virtual ~RTPReceiverVideo(); - int32_t ParseRtpPacket( + virtual int32_t ParseRtpPacket( WebRtcRTPHeader* rtp_header, const ModuleRTPUtility::PayloadUnion& specific_payload, const bool is_red, const uint8_t* packet, const uint16_t packet_length, const int64_t timestamp, - const bool is_first_packet); + const bool is_first_packet) OVERRIDE; - int32_t GetFrequencyHz() const; + virtual int32_t GetFrequencyHz() const OVERRIDE; - RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const; + virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const + OVERRIDE; - bool ShouldReportCsrcChanges(uint8_t payload_type) const; + virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE; - int32_t OnNewPayloadTypeCreated( + virtual int32_t OnNewPayloadTypeCreated( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const int8_t payload_type, - const uint32_t frequency); + const uint32_t frequency) OVERRIDE; - int32_t InvokeOnInitializeDecoder( + virtual int32_t InvokeOnInitializeDecoder( RtpFeedback* callback, const int32_t id, const int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], - const ModuleRTPUtility::PayloadUnion& specific_payload) const; + const ModuleRTPUtility::PayloadUnion& specific_payload) const OVERRIDE; virtual int32_t ReceiveRecoveredPacketCallback( WebRtcRTPHeader* rtp_header, diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h index c70f7cbed..7ad15f646 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h @@ -36,119 +36,119 @@ class ModuleRtpRtcpImpl : public RtpRtcp { // Returns the number of milliseconds until the module want a worker thread to // call Process. - virtual int32_t TimeUntilNextProcess(); + virtual int32_t TimeUntilNextProcess() OVERRIDE; // Process any pending tasks such as timeouts. - virtual int32_t Process(); + virtual int32_t Process() OVERRIDE; // Receiver part. // Configure a timeout value. virtual int32_t SetPacketTimeout(const uint32_t rtp_timeout_ms, - const uint32_t rtcp_timeout_ms); + const uint32_t rtcp_timeout_ms) OVERRIDE; // Set periodic dead or alive notification. virtual int32_t SetPeriodicDeadOrAliveStatus( const bool enable, - const uint8_t sample_time_seconds); + const uint8_t sample_time_seconds) OVERRIDE; // Get periodic dead or alive notification status. virtual int32_t PeriodicDeadOrAliveStatus( bool& enable, - uint8_t& sample_time_seconds); + uint8_t& sample_time_seconds) OVERRIDE; - virtual int32_t RegisterReceivePayload(const CodecInst& voice_codec); + virtual int32_t RegisterReceivePayload(const CodecInst& voice_codec) OVERRIDE; - virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec); + virtual int32_t RegisterReceivePayload( + const VideoCodec& video_codec) OVERRIDE; virtual int32_t ReceivePayloadType(const CodecInst& voice_codec, - int8_t* pl_type); + int8_t* pl_type) OVERRIDE; virtual int32_t ReceivePayloadType(const VideoCodec& video_codec, - int8_t* pl_type); + int8_t* pl_type) OVERRIDE; - virtual int32_t DeRegisterReceivePayload( - const int8_t payload_type); + virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) OVERRIDE; // Get the currently configured SSRC filter. - virtual int32_t SSRCFilter(uint32_t& allowed_ssrc) const; + virtual int32_t SSRCFilter(uint32_t& allowed_ssrc) const OVERRIDE; // Set a SSRC to be used as a filter for incoming RTP streams. virtual int32_t SetSSRCFilter(const bool enable, - const uint32_t allowed_ssrc); + const uint32_t allowed_ssrc) OVERRIDE; // Get last received remote timestamp. - virtual uint32_t RemoteTimestamp() const; + virtual uint32_t RemoteTimestamp() const OVERRIDE; // Get the local time of the last received remote timestamp. - virtual int64_t LocalTimeOfRemoteTimeStamp() const; + virtual int64_t LocalTimeOfRemoteTimeStamp() const OVERRIDE; // Get the current estimated remote timestamp. - virtual int32_t EstimatedRemoteTimeStamp( - uint32_t& timestamp) const; + virtual int32_t EstimatedRemoteTimeStamp(uint32_t& timestamp) const OVERRIDE; - virtual uint32_t RemoteSSRC() const; + virtual uint32_t RemoteSSRC() const OVERRIDE; - virtual int32_t RemoteCSRCs( - uint32_t arr_of_csrc[kRtpCsrcSize]) const; + virtual int32_t RemoteCSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const + OVERRIDE; virtual int32_t SetRTXReceiveStatus(const bool enable, - const uint32_t ssrc); + const uint32_t ssrc) OVERRIDE; virtual int32_t RTXReceiveStatus(bool* enable, uint32_t* ssrc, - int* payloadType) const; + int* payloadType) const OVERRIDE; - virtual void SetRtxReceivePayloadType(int payload_type); + virtual void SetRtxReceivePayloadType(int payload_type) OVERRIDE; // Called when we receive an RTP packet. - virtual int32_t IncomingRtpPacket(const uint8_t* incoming_packet, - const uint16_t packet_length, - const RTPHeader& parsed_rtp_header); + virtual int32_t IncomingRtpPacket( + const uint8_t* incoming_packet, + const uint16_t packet_length, + const RTPHeader& parsed_rtp_header) OVERRIDE; // Called when we receive an RTCP packet. virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, - uint16_t incoming_packet_length); + uint16_t incoming_packet_length) OVERRIDE; // Sender part. - virtual int32_t RegisterSendPayload(const CodecInst& voice_codec); + virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) OVERRIDE; - virtual int32_t RegisterSendPayload(const VideoCodec& video_codec); + virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) OVERRIDE; - virtual int32_t DeRegisterSendPayload(const int8_t payload_type); + virtual int32_t DeRegisterSendPayload(const int8_t payload_type) OVERRIDE; virtual int8_t SendPayloadType() const; // Register RTP header extension. virtual int32_t RegisterSendRtpHeaderExtension( const RTPExtensionType type, - const uint8_t id); + const uint8_t id) OVERRIDE; virtual int32_t DeregisterSendRtpHeaderExtension( - const RTPExtensionType type); + const RTPExtensionType type) OVERRIDE; // Get start timestamp. - virtual uint32_t StartTimestamp() const; + virtual uint32_t StartTimestamp() const OVERRIDE; // Configure start timestamp, default is a random number. - virtual int32_t SetStartTimestamp(const uint32_t timestamp); + virtual int32_t SetStartTimestamp(const uint32_t timestamp) OVERRIDE; - virtual uint16_t SequenceNumber() const; + virtual uint16_t SequenceNumber() const OVERRIDE; // Set SequenceNumber, default is a random number. - virtual int32_t SetSequenceNumber(const uint16_t seq); + virtual int32_t SetSequenceNumber(const uint16_t seq) OVERRIDE; - virtual uint32_t SSRC() const; + virtual uint32_t SSRC() const OVERRIDE; // Configure SSRC, default is a random number. - virtual int32_t SetSSRC(const uint32_t ssrc); + virtual int32_t SetSSRC(const uint32_t ssrc) OVERRIDE; - virtual int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const; + virtual int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const OVERRIDE; virtual int32_t SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize], - const uint8_t arr_length); + const uint8_t arr_length) OVERRIDE; - virtual int32_t SetCSRCStatus(const bool include); + virtual int32_t SetCSRCStatus(const bool include) OVERRIDE; virtual uint32_t PacketCountSent() const; @@ -158,23 +158,23 @@ class ModuleRtpRtcpImpl : public RtpRtcp { virtual int32_t SetRTXSendStatus(const RtxMode mode, const bool set_ssrc, - const uint32_t ssrc); + const uint32_t ssrc) OVERRIDE; virtual int32_t RTXSendStatus(RtxMode* mode, uint32_t* ssrc, - int* payloadType) const; + int* payloadType) const OVERRIDE; - virtual void SetRtxSendPayloadType(int payload_type); + virtual void SetRtxSendPayloadType(int payload_type) OVERRIDE; // Sends kRtcpByeCode when going from true to false. - virtual int32_t SetSendingStatus(const bool sending); + virtual int32_t SetSendingStatus(const bool sending) OVERRIDE; - virtual bool Sending() const; + virtual bool Sending() const OVERRIDE; // Drops or relays media packets. - virtual int32_t SetSendingMediaStatus(const bool sending); + virtual int32_t SetSendingMediaStatus(const bool sending) OVERRIDE; - virtual bool SendingMedia() const; + virtual bool SendingMedia() const OVERRIDE; // Used by the codec module to deliver a video or audio frame for // packetization. @@ -186,78 +186,78 @@ class ModuleRtpRtcpImpl : public RtpRtcp { const uint8_t* payload_data, const uint32_t payload_size, const RTPFragmentationHeader* fragmentation = NULL, - const RTPVideoHeader* rtp_video_hdr = NULL); + const RTPVideoHeader* rtp_video_hdr = NULL) OVERRIDE; virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, - int64_t capture_time_ms); + int64_t capture_time_ms) OVERRIDE; // Returns the number of padding bytes actually sent, which can be more or // less than |bytes|. - virtual int TimeToSendPadding(int bytes); + virtual int TimeToSendPadding(int bytes) OVERRIDE; // RTCP part. // Get RTCP status. - virtual RTCPMethod RTCP() const; + virtual RTCPMethod RTCP() const OVERRIDE; // Configure RTCP status i.e on/off. - virtual int32_t SetRTCPStatus(const RTCPMethod method); + virtual int32_t SetRTCPStatus(const RTCPMethod method) OVERRIDE; // Set RTCP CName. - virtual int32_t SetCNAME(const char c_name[RTCP_CNAME_SIZE]); + virtual int32_t SetCNAME(const char c_name[RTCP_CNAME_SIZE]) OVERRIDE; // Get RTCP CName. - virtual int32_t CNAME(char c_name[RTCP_CNAME_SIZE]); + virtual int32_t CNAME(char c_name[RTCP_CNAME_SIZE]) OVERRIDE; // Get remote CName. virtual int32_t RemoteCNAME(const uint32_t remote_ssrc, - char c_name[RTCP_CNAME_SIZE]) const; + char c_name[RTCP_CNAME_SIZE]) const OVERRIDE; // Get remote NTP. virtual int32_t RemoteNTP(uint32_t* received_ntp_secs, uint32_t* received_ntp_frac, uint32_t* rtcp_arrival_time_secs, uint32_t* rtcp_arrival_time_frac, - uint32_t* rtcp_timestamp) const; + uint32_t* rtcp_timestamp) const OVERRIDE; virtual int32_t AddMixedCNAME(const uint32_t ssrc, - const char c_name[RTCP_CNAME_SIZE]); + const char c_name[RTCP_CNAME_SIZE]) OVERRIDE; - virtual int32_t RemoveMixedCNAME(const uint32_t ssrc); + virtual int32_t RemoveMixedCNAME(const uint32_t ssrc) OVERRIDE; // Get RoundTripTime. virtual int32_t RTT(const uint32_t remote_ssrc, uint16_t* rtt, uint16_t* avg_rtt, uint16_t* min_rtt, - uint16_t* max_rtt) const; + uint16_t* max_rtt) const OVERRIDE; // Reset RoundTripTime statistics. - virtual int32_t ResetRTT(const uint32_t remote_ssrc); + virtual int32_t ResetRTT(const uint32_t remote_ssrc) OVERRIDE; - virtual void SetRtt(uint32_t rtt); + virtual void SetRtt(uint32_t rtt) OVERRIDE; // Force a send of an RTCP packet. // Normal SR and RR are triggered via the process function. - virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport); + virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) OVERRIDE; // Statistics of our locally created statistics of the received RTP stream. virtual int32_t StatisticsRTP(uint8_t* fraction_lost, uint32_t* cum_lost, uint32_t* ext_max, uint32_t* jitter, - uint32_t* max_jitter = NULL) const; + uint32_t* max_jitter = NULL) const OVERRIDE; // Reset RTP statistics. - virtual int32_t ResetStatisticsRTP(); + virtual int32_t ResetStatisticsRTP() OVERRIDE; - virtual int32_t ResetReceiveDataCountersRTP(); + virtual int32_t ResetReceiveDataCountersRTP() OVERRIDE; - virtual int32_t ResetSendDataCountersRTP(); + virtual int32_t ResetSendDataCountersRTP() OVERRIDE; // Statistics of the amount of data sent and received. virtual int32_t DataCountersRTP(uint32_t* bytes_sent, uint32_t* packets_sent, uint32_t* bytes_received, - uint32_t* packets_received) const; + uint32_t* packets_received) const OVERRIDE; virtual int32_t ReportBlockStatistics( uint8_t* fraction_lost, @@ -267,118 +267,120 @@ class ModuleRtpRtcpImpl : public RtpRtcp { uint32_t* jitter_transmission_time_offset); // Get received RTCP report, sender info. - virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info); + virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) OVERRIDE; // Get received RTCP report, report block. virtual int32_t RemoteRTCPStat( - std::vector* receive_blocks) const; + std::vector* receive_blocks) const OVERRIDE; // Set received RTCP report block. virtual int32_t AddRTCPReportBlock( - const uint32_t ssrc, const RTCPReportBlock* receive_block); + const uint32_t ssrc, const RTCPReportBlock* receive_block) OVERRIDE; - virtual int32_t RemoveRTCPReportBlock(const uint32_t ssrc); + virtual int32_t RemoveRTCPReportBlock(const uint32_t ssrc) OVERRIDE; // (REMB) Receiver Estimated Max Bitrate. - virtual bool REMB() const; + virtual bool REMB() const OVERRIDE; - virtual int32_t SetREMBStatus(const bool enable); + virtual int32_t SetREMBStatus(const bool enable) OVERRIDE; virtual int32_t SetREMBData(const uint32_t bitrate, const uint8_t number_of_ssrc, - const uint32_t* ssrc); + const uint32_t* ssrc) OVERRIDE; // (IJ) Extended jitter report. - virtual bool IJ() const; + virtual bool IJ() const OVERRIDE; - virtual int32_t SetIJStatus(const bool enable); + virtual int32_t SetIJStatus(const bool enable) OVERRIDE; // (TMMBR) Temporary Max Media Bit Rate. - virtual bool TMMBR() const; + virtual bool TMMBR() const OVERRIDE; - virtual int32_t SetTMMBRStatus(const bool enable); + virtual int32_t SetTMMBRStatus(const bool enable) OVERRIDE; int32_t SetTMMBN(const TMMBRSet* bounding_set); - virtual uint16_t MaxPayloadLength() const; + virtual uint16_t MaxPayloadLength() const OVERRIDE; - virtual uint16_t MaxDataPayloadLength() const; + virtual uint16_t MaxDataPayloadLength() const OVERRIDE; - virtual int32_t SetMaxTransferUnit(const uint16_t size); + virtual int32_t SetMaxTransferUnit(const uint16_t size) OVERRIDE; virtual int32_t SetTransportOverhead( const bool tcp, const bool ipv6, - const uint8_t authentication_overhead = 0); + const uint8_t authentication_overhead = 0) OVERRIDE; // (NACK) Negative acknowledgment part. // Is Negative acknowledgment requests on/off? - virtual NACKMethod NACK() const; + virtual NACKMethod NACK() const OVERRIDE; // Turn negative acknowledgment requests on/off. virtual int32_t SetNACKStatus(const NACKMethod method, - int max_reordering_threshold); + int max_reordering_threshold) OVERRIDE; - virtual int SelectiveRetransmissions() const; + virtual int SelectiveRetransmissions() const OVERRIDE; - virtual int SetSelectiveRetransmissions(uint8_t settings); + virtual int SetSelectiveRetransmissions(uint8_t settings) OVERRIDE; // Send a Negative acknowledgment packet. - virtual int32_t SendNACK(const uint16_t* nack_list, const uint16_t size); + virtual int32_t SendNACK(const uint16_t* nack_list, + const uint16_t size) OVERRIDE; // Store the sent packets, needed to answer to a negative acknowledgment // requests. virtual int32_t SetStorePacketsStatus( - const bool enable, const uint16_t number_to_store); + const bool enable, const uint16_t number_to_store) OVERRIDE; // (APP) Application specific data. virtual int32_t SetRTCPApplicationSpecificData( const uint8_t sub_type, const uint32_t name, const uint8_t* data, - const uint16_t length); + const uint16_t length) OVERRIDE; // (XR) VOIP metric. - virtual int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric); + virtual int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) OVERRIDE; // Audio part. // Set audio packet size, used to determine when it's time to send a DTMF // packet in silence (CNG). virtual int32_t SetAudioPacketSize( - const uint16_t packet_size_samples); + const uint16_t packet_size_samples) OVERRIDE; // Forward DTMFs to decoder for playout. - virtual int SetTelephoneEventForwardToDecoder(bool forward_to_decoder); + virtual int SetTelephoneEventForwardToDecoder( + bool forward_to_decoder) OVERRIDE; // Is forwarding of outband telephone events turned on/off? - virtual bool TelephoneEventForwardToDecoder() const; + virtual bool TelephoneEventForwardToDecoder() const OVERRIDE; - virtual bool SendTelephoneEventActive(int8_t& telephone_event) const; + virtual bool SendTelephoneEventActive(int8_t& telephone_event) const OVERRIDE; // Send a TelephoneEvent tone using RFC 2833 (4733). virtual int32_t SendTelephoneEventOutband(const uint8_t key, const uint16_t time_ms, - const uint8_t level); + const uint8_t level) OVERRIDE; // Set payload type for Redundant Audio Data RFC 2198. - virtual int32_t SetSendREDPayloadType(const int8_t payload_type); + virtual int32_t SetSendREDPayloadType(const int8_t payload_type) OVERRIDE; // Get payload type for Redundant Audio Data RFC 2198. - virtual int32_t SendREDPayloadType(int8_t& payload_type) const; + virtual int32_t SendREDPayloadType(int8_t& payload_type) const OVERRIDE; // Set status and id for header-extension-for-audio-level-indication. virtual int32_t SetRTPAudioLevelIndicationStatus( - const bool enable, const uint8_t id); + const bool enable, const uint8_t id) OVERRIDE; // Get status and id for header-extension-for-audio-level-indication. virtual int32_t GetRTPAudioLevelIndicationStatus( - bool& enable, uint8_t& id) const; + bool& enable, uint8_t& id) const OVERRIDE; // Store the audio level in d_bov for header-extension-for-audio-level- // indication. - virtual int32_t SetAudioLevel(const uint8_t level_d_bov); + virtual int32_t SetAudioLevel(const uint8_t level_d_bov) OVERRIDE; // Video part. @@ -387,32 +389,32 @@ class ModuleRtpRtcpImpl : public RtpRtcp { virtual RtpVideoCodecTypes SendVideoCodec() const; virtual int32_t SendRTCPSliceLossIndication( - const uint8_t picture_id); + const uint8_t picture_id) OVERRIDE; // Set method for requestion a new key frame. virtual int32_t SetKeyFrameRequestMethod( - const KeyFrameRequestMethod method); + const KeyFrameRequestMethod method) OVERRIDE; // Send a request for a keyframe. - virtual int32_t RequestKeyFrame(); + virtual int32_t RequestKeyFrame() OVERRIDE; - virtual int32_t SetCameraDelay(const int32_t delay_ms); + virtual int32_t SetCameraDelay(const int32_t delay_ms) OVERRIDE; - virtual void SetTargetSendBitrate(const uint32_t bitrate); + virtual void SetTargetSendBitrate(const uint32_t bitrate) OVERRIDE; virtual int32_t SetGenericFECStatus( const bool enable, const uint8_t payload_type_red, - const uint8_t payload_type_fec); + const uint8_t payload_type_fec) OVERRIDE; virtual int32_t GenericFECStatus( bool& enable, uint8_t& payload_type_red, - uint8_t& payload_type_fec); + uint8_t& payload_type_fec) OVERRIDE; virtual int32_t SetFecParameters( const FecProtectionParams* delta_params, - const FecProtectionParams* key_params); + const FecProtectionParams* key_params) OVERRIDE; virtual int32_t LastReceivedNTP(uint32_t& NTPsecs, uint32_t& NTPfrac, @@ -423,7 +425,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp { virtual void BitrateSent(uint32_t* total_rate, uint32_t* video_rate, uint32_t* fec_rate, - uint32_t* nackRate) const; + uint32_t* nackRate) const OVERRIDE; virtual void SetRemoteSSRC(const uint32_t ssrc); @@ -431,7 +433,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp { // Good state of RTP receiver inform sender. virtual int32_t SendRTCPReferencePictureSelection( - const uint64_t picture_id); + const uint64_t picture_id) OVERRIDE; void OnReceivedTMMBR(); diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h index 61dc1c57e..eef2440c4 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h @@ -71,7 +71,7 @@ class RTPSender : public Bitrate, public RTPSenderInterface { void ProcessBitrate(); - uint16_t ActualSendBitrateKbit() const; + virtual uint16_t ActualSendBitrateKbit() const OVERRIDE; uint32_t VideoBitrateSent() const; uint32_t FecOverheadRate() const; @@ -79,7 +79,8 @@ class RTPSender : public Bitrate, public RTPSenderInterface { void SetTargetSendBitrate(const uint32_t bits); - uint16_t MaxDataPayloadLength() const; // with RTP and FEC headers. + virtual uint16_t MaxDataPayloadLength() const + OVERRIDE; // with RTP and FEC headers. int32_t RegisterPayload( const char payload_name[RTP_PAYLOAD_NAME_SIZE], @@ -111,7 +112,7 @@ class RTPSender : public Bitrate, public RTPSenderInterface { uint32_t GenerateNewSSRC(); void SetSSRC(const uint32_t ssrc); - uint16_t SequenceNumber() const; + virtual uint16_t SequenceNumber() const OVERRIDE; void SetSequenceNumber(uint16_t seq); int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const; @@ -196,21 +197,21 @@ class RTPSender : public Bitrate, public RTPSenderInterface { const bool marker_bit, const uint32_t capture_time_stamp, int64_t capture_time_ms, const bool time_stamp_provided = true, - const bool inc_sequence_number = true); + const bool inc_sequence_number = true) OVERRIDE; - virtual uint16_t RTPHeaderLength() const; - virtual uint16_t IncrementSequenceNumber(); - virtual uint16_t MaxPayloadLength() const; - virtual uint16_t PacketOverHead() const; + virtual uint16_t RTPHeaderLength() const OVERRIDE; + virtual uint16_t IncrementSequenceNumber() OVERRIDE; + virtual uint16_t MaxPayloadLength() const OVERRIDE; + virtual uint16_t PacketOverHead() const OVERRIDE; // Current timestamp. - virtual uint32_t Timestamp() const; - virtual uint32_t SSRC() const; + virtual uint32_t Timestamp() const OVERRIDE; + virtual uint32_t SSRC() const OVERRIDE; virtual int32_t SendToNetwork( uint8_t *data_buffer, int payload_length, int rtp_header_length, int64_t capture_time_ms, StorageType storage, - PacedSender::Priority priority); + PacedSender::Priority priority) OVERRIDE; // Audio.