From 62bafae6618fe3aefbd18657062abc98a40c3375 Mon Sep 17 00:00:00 2001 From: "pbos@webrtc.org" Date: Tue, 8 Jul 2014 12:10:51 +0000 Subject: [PATCH] Some refactoring inside rtp_rtcp/. Renaming ModuleRTPUtility -> RtpUtility. Renaming RTPHeaderParser -> RtpHeaderParser. Making RtpHeaderParser accept size_t instead of int for packet length. Making RtpUtility::RtpHeaderParser accept size_t for packet length. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../rtp_rtcp/interface/rtp_header_parser.h | 5 +- .../rtp_rtcp/interface/rtp_payload_registry.h | 25 ++-- .../rtp_rtcp/source/H264/rtp_sender_h264.cc | 19 ++- .../rtp_rtcp/source/fec_receiver_impl.cc | 2 +- .../rtp_rtcp/source/fec_test_helper.cc | 6 +- .../source/forward_error_correction.cc | 22 ++- .../source/mock/mock_rtp_payload_strategy.h | 28 ++-- .../modules/rtp_rtcp/source/producer_fec.cc | 2 +- .../source/receive_statistics_impl.cc | 11 +- webrtc/modules/rtp_rtcp/source/rtcp_packet.cc | 6 +- webrtc/modules/rtp_rtcp/source/rtcp_sender.cc | 125 +++++++++--------- .../rtp_rtcp/source/rtp_fec_unittest.cc | 13 +- .../rtp_rtcp/source/rtp_header_parser.cc | 16 ++- .../rtp_rtcp/source/rtp_payload_registry.cc | 112 ++++++++-------- .../source/rtp_payload_registry_unittest.cc | 31 ++--- .../rtp_rtcp/source/rtp_receiver_audio.cc | 4 +- .../rtp_rtcp/source/rtp_receiver_audio.h | 2 +- .../rtp_rtcp/source/rtp_receiver_impl.cc | 8 +- .../rtp_rtcp/source/rtp_receiver_video.cc | 25 ++-- .../rtp_rtcp/source/rtp_rtcp_impl_unittest.cc | 5 +- webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 74 +++++------ webrtc/modules/rtp_rtcp/source/rtp_sender.h | 2 +- .../rtp_rtcp/source/rtp_sender_audio.cc | 16 +-- .../rtp_rtcp/source/rtp_sender_audio.h | 13 +- .../rtp_rtcp/source/rtp_sender_unittest.cc | 36 ++--- .../rtp_rtcp/source/rtp_sender_video.cc | 12 +- .../rtp_rtcp/source/rtp_sender_video.h | 9 +- webrtc/modules/rtp_rtcp/source/rtp_utility.cc | 29 ++-- webrtc/modules/rtp_rtcp/source/rtp_utility.h | 13 +- .../rtp_rtcp/source/rtp_utility_unittest.cc | 23 ++-- .../modules/rtp_rtcp/test/testAPI/test_api.h | 4 +- .../rtp_rtcp/test/testAPI/test_api_video.cc | 8 +- .../modules/rtp_rtcp/test/testFec/test_fec.cc | 12 +- webrtc/modules/utility/source/rtp_dump_impl.h | 2 +- .../main/test/pcap_file_reader.cc | 3 +- .../main/test/pcap_file_reader_unittest.cc | 2 +- webrtc/test/rtp_rtcp_observer.h | 4 +- webrtc/video/call.cc | 5 +- webrtc/video/call_perf_tests.cc | 2 +- webrtc/video/end_to_end_tests.cc | 17 ++- webrtc/video/full_stack.cc | 4 +- webrtc/video/rampup_tests.cc | 7 +- webrtc/video/video_send_stream_tests.cc | 14 +- .../auto_test/automated/vie_network_test.cc | 2 +- .../auto_test/standard/rtp_rtcp_extensions.cc | 4 +- 45 files changed, 384 insertions(+), 400 deletions(-) diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h index a13f5b802b..2809996b25 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h @@ -23,13 +23,14 @@ class RtpHeaderParser { virtual ~RtpHeaderParser() {} // Returns true if the packet is an RTCP packet, false otherwise. - static bool IsRtcp(const uint8_t* packet, int length); + static bool IsRtcp(const uint8_t* packet, size_t length); // Parses the packet and stores the parsed packet in |header|. Returns true on // success, false otherwise. // This method is thread-safe in the sense that it can parse multiple packets // at once. - virtual bool Parse(const uint8_t* packet, int length, + virtual bool Parse(const uint8_t* packet, + size_t length, RTPHeader* header) const = 0; // Registers an RTP header extension and binds it to |id|. diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h index 965f4b0242..327ea165d5 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h @@ -25,17 +25,15 @@ class RTPPayloadStrategy { virtual bool CodecsMustBeUnique() const = 0; - virtual bool PayloadIsCompatible( - const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate) const = 0; + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const = 0; - virtual void UpdatePayloadRate( - ModuleRTPUtility::Payload* payload, - const uint32_t rate) const = 0; + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const = 0; - virtual ModuleRTPUtility::Payload* CreatePayloadType( + virtual RtpUtility::Payload* CreatePayloadType( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, @@ -43,7 +41,7 @@ class RTPPayloadStrategy { const uint32_t rate) const = 0; virtual int GetPayloadTypeFrequency( - const ModuleRTPUtility::Payload& payload) const = 0; + const RtpUtility::Payload& payload) const = 0; static RTPPayloadStrategy* CreateStrategy(const bool handling_audio); @@ -99,9 +97,8 @@ class RTPPayloadRegistry { int GetPayloadTypeFrequency(uint8_t payload_type) const; - bool PayloadTypeToPayload( - const uint8_t payload_type, - ModuleRTPUtility::Payload*& payload) const; + bool PayloadTypeToPayload(const uint8_t payload_type, + RtpUtility::Payload*& payload) const; void ResetLastReceivedPayloadTypes() { CriticalSectionScoped cs(crit_sect_.get()); @@ -151,7 +148,7 @@ class RTPPayloadRegistry { bool IsRtxInternal(const RTPHeader& header) const; scoped_ptr crit_sect_; - ModuleRTPUtility::PayloadTypeMap payload_type_map_; + RtpUtility::PayloadTypeMap payload_type_map_; scoped_ptr rtp_payload_strategy_; int8_t red_payload_type_; int8_t ulpfec_payload_type_; diff --git a/webrtc/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc b/webrtc/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc index d62f50b4c0..6560209cbf 100644 --- a/webrtc/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc +++ b/webrtc/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc @@ -238,9 +238,14 @@ RTPSenderH264::SendH264FillerData(const WebRtcRTPHeader* rtpHeader, dataBuffer[0] = static_cast(0x80); // version 2 dataBuffer[1] = rtpHeader->header.payloadType; - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc); + RtpUtility::AssignUWord16ToBuffer( + dataBuffer + 2, + _rtpSender.IncrementSequenceNumber()); // get the current + // SequenceNumber and add by 1 + // after returning + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, + rtpHeader->header.timestamp); + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, rtpHeader->header.ssrc); // set filler NALU type dataBuffer[12] = 12; // NRI field = 0, type 12 @@ -361,8 +366,12 @@ RTPSenderH264::SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader, // _sequenceNumber will not work for re-ordering by NACK from original sender // engine responsible for this - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning - //ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc); + RtpUtility::AssignUWord16ToBuffer( + dataBuffer + 2, + _rtpSender.IncrementSequenceNumber()); // get the current + // SequenceNumber and add by 1 + // after returning + // RtpUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc); // how do we know it's the last relayed packet in a frame? // 1) packets arrive in order, the engine manages that diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc index 0d6c174a3b..e795841f1c 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc @@ -171,7 +171,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket( payload_data_length - REDHeaderLength); received_packet->pkt->length = payload_data_length - REDHeaderLength; received_packet->ssrc = - ModuleRTPUtility::BufferToUWord32(&incoming_rtp_packet[8]); + RtpUtility::BufferToUWord32(&incoming_rtp_packet[8]); } else { // copy the RTP header diff --git a/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc b/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc index 176954f942..0ffd5bf4fa 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc +++ b/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc @@ -86,9 +86,9 @@ void FrameGenerator::BuildRtpHeader(uint8_t* data, const RTPHeader* header) { data[0] = 0x80; // Version 2. data[1] = header->payloadType; data[1] |= (header->markerBit ? kRtpMarkerBitMask : 0); - ModuleRTPUtility::AssignUWord16ToBuffer(data + 2, header->sequenceNumber); - ModuleRTPUtility::AssignUWord32ToBuffer(data + 4, header->timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(data + 8, header->ssrc); + RtpUtility::AssignUWord16ToBuffer(data + 2, header->sequenceNumber); + RtpUtility::AssignUWord32ToBuffer(data + 4, header->timestamp); + RtpUtility::AssignUWord32ToBuffer(data + 8, header->ssrc); } } // 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 eeead40745..b02ea08652 100644 --- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc +++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc @@ -230,7 +230,7 @@ void ForwardErrorCorrection::GenerateFecBitStrings( Packet* media_packet = *media_list_it; // Assign network-ordered media payload length. - ModuleRTPUtility::AssignUWord16ToBuffer( + RtpUtility::AssignUWord16ToBuffer( media_payload_length, media_packet->length - kRtpHeaderSize); fec_packet_length = media_packet->length + fec_rtp_offset; @@ -432,7 +432,7 @@ void ForwardErrorCorrection::GenerateFecUlpHeaders( // -- ULP header -- // Copy the payload size to the protection length field. // (We protect the entire packet.) - ModuleRTPUtility::AssignUWord16ToBuffer( + RtpUtility::AssignUWord16ToBuffer( &generated_fec_packets_[i].data[10], generated_fec_packets_[i].length - kFecHeaderSize - ulp_header_size); @@ -537,7 +537,7 @@ void ForwardErrorCorrection::InsertFECPacket( fec_packet->ssrc = rx_packet->ssrc; const uint16_t seq_num_base = - ModuleRTPUtility::BufferToUWord16(&fec_packet->pkt->data[2]); + RtpUtility::BufferToUWord16(&fec_packet->pkt->data[2]); const uint16_t maskSizeBytes = (fec_packet->pkt->data[0] & 0x40) ? kMaskSizeLBitSet : kMaskSizeLBitClear; // L bit set? @@ -650,7 +650,7 @@ void ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet, // Copy FEC payload, skipping the ULP header. memcpy(&recovered->pkt->data[kRtpHeaderSize], &fec_packet->pkt->data[kFecHeaderSize + ulp_header_size], - ModuleRTPUtility::BufferToUWord16(protection_length)); + RtpUtility::BufferToUWord16(protection_length)); // Copy the length recovery field. memcpy(recovered->length_recovery, &fec_packet->pkt->data[8], 2); // Copy the first 2 bytes of the FEC header. @@ -658,8 +658,7 @@ void ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet, // Copy the 5th to 8th bytes of the FEC header. memcpy(&recovered->pkt->data[4], &fec_packet->pkt->data[4], 4); // Set the SSRC field. - ModuleRTPUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8], - fec_packet->ssrc); + RtpUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8], fec_packet->ssrc); } void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) { @@ -668,12 +667,11 @@ void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) { recovered->pkt->data[0] &= 0xbf; // Clear the 2nd bit. // Set the SN field. - ModuleRTPUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2], - recovered->seq_num); + RtpUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2], + recovered->seq_num); // Recover the packet length. recovered->pkt->length = - ModuleRTPUtility::BufferToUWord16(recovered->length_recovery) + - kRtpHeaderSize; + RtpUtility::BufferToUWord16(recovered->length_recovery) + kRtpHeaderSize; } void ForwardErrorCorrection::XorPackets(const Packet* src_packet, @@ -688,8 +686,8 @@ void ForwardErrorCorrection::XorPackets(const Packet* src_packet, } // XOR with the network-ordered payload size. uint8_t media_payload_length[2]; - ModuleRTPUtility::AssignUWord16ToBuffer(media_payload_length, - src_packet->length - kRtpHeaderSize); + RtpUtility::AssignUWord16ToBuffer(media_payload_length, + src_packet->length - kRtpHeaderSize); dst_packet->length_recovery[0] ^= media_payload_length[0]; dst_packet->length_recovery[1] ^= media_payload_length[1]; diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h index ccf82e5d68..f577cbaad1 100644 --- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h +++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h @@ -21,21 +21,21 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy { MOCK_CONST_METHOD0(CodecsMustBeUnique, bool()); MOCK_CONST_METHOD4(PayloadIsCompatible, - bool(const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate)); + bool(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate)); MOCK_CONST_METHOD2(UpdatePayloadRate, - void(ModuleRTPUtility::Payload* payload, const uint32_t rate)); - MOCK_CONST_METHOD1(GetPayloadTypeFrequency, int( - const ModuleRTPUtility::Payload& payload)); - MOCK_CONST_METHOD5(CreatePayloadType, - ModuleRTPUtility::Payload*( - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate)); + void(RtpUtility::Payload* payload, const uint32_t rate)); + MOCK_CONST_METHOD1(GetPayloadTypeFrequency, + int(const RtpUtility::Payload& payload)); + MOCK_CONST_METHOD5( + CreatePayloadType, + RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate)); }; } // namespace webrtc diff --git a/webrtc/modules/rtp_rtcp/source/producer_fec.cc b/webrtc/modules/rtp_rtcp/source/producer_fec.cc index 3173d3d529..747cd89e9c 100644 --- a/webrtc/modules/rtp_rtcp/source/producer_fec.cc +++ b/webrtc/modules/rtp_rtcp/source/producer_fec.cc @@ -61,7 +61,7 @@ void RedPacket::CreateHeader(const uint8_t* rtp_header, int header_length, void RedPacket::SetSeqNum(int seq_num) { assert(seq_num >= 0 && seq_num < (1<<16)); - ModuleRTPUtility::AssignUWord16ToBuffer(&data_[2], seq_num); + RtpUtility::AssignUWord16ToBuffer(&data_[2], seq_num); } void RedPacket::AssignPayload(const uint8_t* payload, int length) { diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc index aa7c9c5715..e3bc95f79d 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -133,11 +133,12 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header, void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header, uint32_t receive_time_secs, uint32_t receive_time_frac) { - uint32_t receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( - receive_time_secs, receive_time_frac, header.payload_type_frequency); - uint32_t last_receive_time_rtp = ModuleRTPUtility::ConvertNTPTimeToRTP( - last_receive_time_secs_, last_receive_time_frac_, - header.payload_type_frequency); + uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP( + receive_time_secs, receive_time_frac, header.payload_type_frequency); + uint32_t last_receive_time_rtp = + RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_, + last_receive_time_frac_, + header.payload_type_frequency); int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) - (header.timestamp - last_received_timestamp_); diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc index f6d3bd3d71..68da3aebe6 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc @@ -61,15 +61,15 @@ void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) { buffer[(*offset)++] = value; } void AssignUWord16(uint8_t* buffer, size_t* offset, uint16_t value) { - ModuleRTPUtility::AssignUWord16ToBuffer(buffer + *offset, value); + RtpUtility::AssignUWord16ToBuffer(buffer + *offset, value); *offset += 2; } void AssignUWord24(uint8_t* buffer, size_t* offset, uint32_t value) { - ModuleRTPUtility::AssignUWord24ToBuffer(buffer + *offset, value); + RtpUtility::AssignUWord24ToBuffer(buffer + *offset, value); *offset += 3; } void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) { - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + *offset, value); + RtpUtility::AssignUWord32ToBuffer(buffer + *offset, value); *offset += 4; } diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index d73de9c424..b9ab0c1e97 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc @@ -708,24 +708,24 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state, pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // NTP - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, NTPsec); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPsec); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, NTPfrac); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPfrac); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, RTPtime); pos += 4; //sender's packet count - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, - feedback_state.packet_count_sent); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, + feedback_state.packet_count_sent); pos += 4; //sender's octet count - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, - feedback_state.byte_count_sent); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, + feedback_state.byte_count_sent); pos += 4; uint8_t numberOfReportBlocks = 0; @@ -741,7 +741,7 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state, rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; uint16_t len = uint16_t((pos/4) -1); - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len); return 0; } @@ -767,7 +767,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // CNAME = 1 @@ -802,7 +802,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { uint32_t SSRC = it->first; // Add SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, SSRC); pos += 4; // CNAME = 1 @@ -833,8 +833,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) { } // in 32-bit words minus one and we don't count the header uint16_t buffer_length = (SDESLength / 4) - 1; - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, - buffer_length); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, buffer_length); return 0; } @@ -859,7 +858,7 @@ RTCPSender::BuildRR(uint8_t* rtcpbuffer, pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; uint8_t numberOfReportBlocks = 0; @@ -874,7 +873,7 @@ RTCPSender::BuildRR(uint8_t* rtcpbuffer, rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; uint16_t len = uint16_t((pos)/4 -1); - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len); return 0; } @@ -925,8 +924,8 @@ RTCPSender::BuildExtendedJitterReport( rtcpbuffer[pos++]=(uint8_t)(1); // Add inter-arrival jitter - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, - jitterTransmissionTimeOffset); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, + jitterTransmissionTimeOffset); pos += 4; return 0; } @@ -949,11 +948,11 @@ RTCPSender::BuildPLI(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(2); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; return 0; } @@ -979,7 +978,7 @@ int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer, rtcpbuffer[pos++] = (uint8_t)(4); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // RFC 5104 4.3.1.2. Semantics @@ -990,7 +989,7 @@ int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer, rtcpbuffer[pos++] = (uint8_t)0; // Additional Feedback Control Information (FCI) - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; rtcpbuffer[pos++] = (uint8_t)(_sequenceNumberFIR); @@ -1025,18 +1024,18 @@ RTCPSender::BuildSLI(uint8_t* rtcpbuffer, int& pos, const uint8_t pictureID) rtcpbuffer[pos++]=(uint8_t)(3); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; // Add first, number & picture ID 6 bits // first = 0, 13 - bits // number = 0x1fff, 13 - bits only ones for now uint32_t sliField = (0x1fff << 6)+ (0x3f & pictureID); - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, sliField); pos += 4; return 0; } @@ -1090,11 +1089,11 @@ RTCPSender::BuildRPSI(uint8_t* rtcpbuffer, rtcpbuffer[pos++]=size; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; // calc padding length @@ -1147,11 +1146,11 @@ RTCPSender::BuildREMB(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=_lengthRembSSRC + 4; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Remote SSRC must be 0 - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, 0); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, 0); pos += 4; rtcpbuffer[pos++]='R'; @@ -1177,8 +1176,8 @@ RTCPSender::BuildREMB(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(brMantissa); for (int i = 0; i < _lengthRembSSRC; i++) - { - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]); + { + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _rembSSRC[i]); pos += 4; } return 0; @@ -1264,7 +1263,7 @@ int32_t RTCPSender::BuildTMMBR(ModuleRtpRtcpImpl* rtp_rtcp_module, rtcpbuffer[pos++]=(uint8_t)(4); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // RFC 5104 4.2.1.2. Semantics @@ -1276,7 +1275,7 @@ int32_t RTCPSender::BuildTMMBR(ModuleRtpRtcpImpl* rtp_rtcp_module, rtcpbuffer[pos++]=(uint8_t)0; // Additional Feedback Control Information (FCI) - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; uint32_t bitRate = _tmmbr_Send*1000; @@ -1324,7 +1323,7 @@ RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos) pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // RFC 5104 4.2.2.2. Semantics @@ -1342,7 +1341,7 @@ RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos) if (boundingSet->Tmmbr(n) > 0) { uint32_t tmmbrSSRC = boundingSet->Ssrc(n); - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, tmmbrSSRC); pos += 4; uint32_t bitRate = boundingSet->Tmmbr(n) * 1000; @@ -1395,11 +1394,11 @@ RTCPSender::BuildAPP(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(length); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add our application name - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _appName); pos += 4; // Add the data @@ -1433,11 +1432,11 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer, rtcpbuffer[pos++]=(uint8_t)(3); //setting it to one kNACK signal as default // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; NACKStringBuilder stringBuilder; @@ -1464,9 +1463,9 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer, } // Write the sequence number and the bitmask to the packet. assert(pos + 4 < IP_PACKET_SIZE); - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, nack); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, nack); pos += 2; - ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, bitmask); + RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, bitmask); pos += 2; numOfNackFields++; } @@ -1497,13 +1496,13 @@ RTCPSender::BuildBYE(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)(1 + _CSRCs); // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // add CSRCs for(int i = 0; i < _CSRCs; i++) { - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _CSRC[i]); pos += 4; } } else @@ -1517,7 +1516,7 @@ RTCPSender::BuildBYE(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=(uint8_t)1; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; } return 0; @@ -1546,7 +1545,7 @@ int32_t RTCPSender::BuildReceiverReferenceTime(uint8_t* buffer, buffer[pos++] = 4; // XR packet length. // Add our own SSRC. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); pos += 4; // 0 1 2 3 @@ -1566,9 +1565,9 @@ int32_t RTCPSender::BuildReceiverReferenceTime(uint8_t* buffer, buffer[pos++] = 2; // Block length. // NTP timestamp. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, ntp_sec); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, ntp_sec); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, ntp_frac); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, ntp_frac); pos += 4; return 0; @@ -1589,7 +1588,7 @@ int32_t RTCPSender::BuildDlrr(uint8_t* buffer, buffer[pos++] = 5; // XR packet length. // Add our own SSRC. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC); pos += 4; // 0 1 2 3 @@ -1614,11 +1613,11 @@ int32_t RTCPSender::BuildDlrr(uint8_t* buffer, buffer[pos++] = 3; // Block length. // NTP timestamp. - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, info.sourceSSRC); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.sourceSSRC); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, info.lastRR); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.lastRR); pos += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(buffer + pos, info.delaySinceLastRR); + RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.delaySinceLastRR); pos += 4; return 0; @@ -1644,7 +1643,7 @@ RTCPSender::BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos) pos++; // Add our own SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); pos += 4; // Add a VoIP metrics block @@ -1654,7 +1653,7 @@ RTCPSender::BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos) rtcpbuffer[pos++]=8; // Add the remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); pos += 4; rtcpbuffer[pos++] = _xrVoIPMetric.lossRate; @@ -2210,33 +2209,33 @@ int32_t RTCPSender::WriteReportBlocksToBuffer( RTCPReportBlock* reportBlock = it->second; if (reportBlock) { // Remote SSRC - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, remoteSSRC); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, remoteSSRC); position += 4; // fraction lost rtcpbuffer[position++] = reportBlock->fractionLost; // cumulative loss - ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+position, - reportBlock->cumulativeLost); + RtpUtility::AssignUWord24ToBuffer(rtcpbuffer + position, + reportBlock->cumulativeLost); position += 3; // extended highest seq_no, contain the highest sequence number received - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->extendedHighSeqNum); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->extendedHighSeqNum); position += 4; // Jitter - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->jitter); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->jitter); position += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->lastSR); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->lastSR); position += 4; - ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+position, - reportBlock->delaySinceLastSR); + RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, + reportBlock->delaySinceLastSR); position += 4; } } diff --git a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc index 993433b652..9d19fde1a6 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc @@ -866,7 +866,7 @@ void RtpFecTest::ReceivedPackets(const PacketList& packet_list, int* loss_mask, // For media packets, the sequence number and marker bit is // obtained from RTP header. These were set in ConstructMediaPackets(). received_packet->seq_num = - webrtc::ModuleRTPUtility::BufferToUWord16(&packet->data[2]); + webrtc::RtpUtility::BufferToUWord16(&packet->data[2]); } else { // The sequence number, marker bit, and ssrc number are defined in the // RTP header of the FEC packet, which is not constructed in this test. @@ -921,12 +921,11 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets, // Only push one (fake) frame to the FEC. media_packet->data[1] &= 0x7f; - webrtc::ModuleRTPUtility::AssignUWord16ToBuffer(&media_packet->data[2], - sequence_number); - webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[4], - time_stamp); - webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[8], - ssrc_); + webrtc::RtpUtility::AssignUWord16ToBuffer(&media_packet->data[2], + sequence_number); + webrtc::RtpUtility::AssignUWord32ToBuffer(&media_packet->data[4], + time_stamp); + webrtc::RtpUtility::AssignUWord32ToBuffer(&media_packet->data[8], ssrc_); // Generate random values for payload. for (int j = 12; j < media_packet->length; ++j) { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc index bb24d4dbfb..3fc26663e0 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc @@ -21,7 +21,8 @@ class RtpHeaderParserImpl : public RtpHeaderParser { RtpHeaderParserImpl(); virtual ~RtpHeaderParserImpl() {} - virtual bool Parse(const uint8_t* packet, int length, + virtual bool Parse(const uint8_t* packet, + size_t length, RTPHeader* header) const OVERRIDE; virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, @@ -31,7 +32,7 @@ class RtpHeaderParserImpl : public RtpHeaderParser { private: scoped_ptr critical_section_; - RtpHeaderExtensionMap rtp_header_extension_map_; + RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(critical_section_); }; RtpHeaderParser* RtpHeaderParser::Create() { @@ -41,14 +42,15 @@ RtpHeaderParser* RtpHeaderParser::Create() { RtpHeaderParserImpl::RtpHeaderParserImpl() : critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {} -bool RtpHeaderParser::IsRtcp(const uint8_t* packet, int length) { - ModuleRTPUtility::RTPHeaderParser rtp_parser(packet, length); +bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) { + RtpUtility::RtpHeaderParser rtp_parser(packet, length); return rtp_parser.RTCP(); } -bool RtpHeaderParserImpl::Parse(const uint8_t* packet, int length, - RTPHeader* header) const { - ModuleRTPUtility::RTPHeaderParser rtp_parser(packet, length); +bool RtpHeaderParserImpl::Parse(const uint8_t* packet, + size_t length, + RTPHeader* header) const { + RtpUtility::RtpHeaderParser rtp_parser(packet, length); memset(header, 0, sizeof(*header)); RtpHeaderExtensionMap map; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc index 26db8f3292..ec05a73bfd 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -29,7 +29,7 @@ RTPPayloadRegistry::RTPPayloadRegistry( RTPPayloadRegistry::~RTPPayloadRegistry() { while (!payload_type_map_.empty()) { - ModuleRTPUtility::PayloadTypeMap::iterator it = payload_type_map_.begin(); + RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.begin(); delete it->second; payload_type_map_.erase(it); } @@ -69,12 +69,12 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::iterator it = - payload_type_map_.find(payload_type); + RtpUtility::PayloadTypeMap::iterator it = + payload_type_map_.find(payload_type); if (it != payload_type_map_.end()) { // We already use this payload type. - ModuleRTPUtility::Payload* payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); @@ -83,7 +83,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( // Check if it's the same as we already have. // If same, ignore sending an error. if (payload_name_length == name_length && - ModuleRTPUtility::StringCompare( + RtpUtility::StringCompare( payload->name, payload_name, payload_name_length)) { if (rtp_payload_strategy_->PayloadIsCompatible(*payload, frequency, channels, rate)) { @@ -100,18 +100,18 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( payload_name, payload_name_length, frequency, channels, rate); } - ModuleRTPUtility::Payload* payload = NULL; + RtpUtility::Payload* payload = NULL; // Save the RED payload type. Used in both audio and video. - if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) { + if (RtpUtility::StringCompare(payload_name, "red", 3)) { red_payload_type_ = payload_type; - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; memset(payload, 0, sizeof(*payload)); payload->audio = false; strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); - } else if (ModuleRTPUtility::StringCompare(payload_name, "ulpfec", 3)) { + } else if (RtpUtility::StringCompare(payload_name, "ulpfec", 3)) { ulpfec_payload_type_ = payload_type; - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; memset(payload, 0, sizeof(*payload)); payload->audio = false; strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); @@ -132,7 +132,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload( int32_t RTPPayloadRegistry::DeRegisterReceivePayload( const int8_t payload_type) { CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::iterator it = + RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.find(payload_type); assert(it != payload_type_map_.end()); delete it->second; @@ -149,15 +149,14 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( const uint32_t frequency, const uint8_t channels, const uint32_t rate) { - ModuleRTPUtility::PayloadTypeMap::iterator iterator = - payload_type_map_.begin(); + RtpUtility::PayloadTypeMap::iterator iterator = payload_type_map_.begin(); for (; iterator != payload_type_map_.end(); ++iterator) { - ModuleRTPUtility::Payload* payload = iterator->second; + RtpUtility::Payload* payload = iterator->second; size_t name_length = strlen(payload->name); - if (payload_name_length == name_length - && ModuleRTPUtility::StringCompare(payload->name, payload_name, - payload_name_length)) { + if (payload_name_length == name_length && + RtpUtility::StringCompare( + payload->name, payload_name, payload_name_length)) { // We found the payload name in the list. // If audio, check frequency and rate. if (payload->audio) { @@ -168,7 +167,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( payload_type_map_.erase(iterator); break; } - } else if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) { + } else if (RtpUtility::StringCompare(payload_name, "red", 3)) { delete payload; payload_type_map_.erase(iterator); break; @@ -188,16 +187,15 @@ int32_t RTPPayloadRegistry::ReceivePayloadType( CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::const_iterator it = - payload_type_map_.begin(); + RtpUtility::PayloadTypeMap::const_iterator it = payload_type_map_.begin(); for (; it != payload_type_map_.end(); ++it) { - ModuleRTPUtility::Payload* payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); size_t name_length = strlen(payload->name); if (payload_name_length == name_length && - ModuleRTPUtility::StringCompare( + RtpUtility::StringCompare( payload->name, payload_name, payload_name_length)) { // Name matches. if (payload->audio) { @@ -261,9 +259,9 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet, *packet_length -= kRtxHeaderSize; // Replace the SSRC and the sequence number with the originals. - ModuleRTPUtility::AssignUWord16ToBuffer(*restored_packet + 2, - original_sequence_number); - ModuleRTPUtility::AssignUWord32ToBuffer(*restored_packet + 8, original_ssrc); + RtpUtility::AssignUWord16ToBuffer(*restored_packet + 2, + original_sequence_number); + RtpUtility::AssignUWord32ToBuffer(*restored_packet + 8, original_ssrc); CriticalSectionScoped cs(crit_sect_.get()); @@ -307,8 +305,8 @@ bool RTPPayloadRegistry::IsEncapsulated(const RTPHeader& header) const { bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const { CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::const_iterator it = - payload_type_map_.find(payload_type); + RtpUtility::PayloadTypeMap::const_iterator it = + payload_type_map_.find(payload_type); // Check that this is a registered payload type. if (it == payload_type_map_.end()) { @@ -320,7 +318,7 @@ bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type, int RTPPayloadRegistry::GetPayloadTypeFrequency( uint8_t payload_type) const { - ModuleRTPUtility::Payload* payload; + RtpUtility::Payload* payload; if (!PayloadTypeToPayload(payload_type, payload)) { return -1; } @@ -329,12 +327,12 @@ int RTPPayloadRegistry::GetPayloadTypeFrequency( } bool RTPPayloadRegistry::PayloadTypeToPayload( - const uint8_t payload_type, - ModuleRTPUtility::Payload*& payload) const { + const uint8_t payload_type, + RtpUtility::Payload*& payload) const { CriticalSectionScoped cs(crit_sect_.get()); - ModuleRTPUtility::PayloadTypeMap::const_iterator it = - payload_type_map_.find(payload_type); + RtpUtility::PayloadTypeMap::const_iterator it = + payload_type_map_.find(payload_type); // Check that this is a registered payload type. if (it == payload_type_map_.end()) { @@ -365,11 +363,10 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { public: virtual bool CodecsMustBeUnique() const OVERRIDE { return true; } - virtual bool PayloadIsCompatible( - const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate) const OVERRIDE { + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const OVERRIDE { return payload.audio && payload.typeSpecific.Audio.frequency == frequency && @@ -378,19 +375,18 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { payload.typeSpecific.Audio.rate == 0 || rate == 0); } - virtual void UpdatePayloadRate( - ModuleRTPUtility::Payload* payload, - const uint32_t rate) const OVERRIDE { + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const OVERRIDE { payload->typeSpecific.Audio.rate = rate; } - virtual ModuleRTPUtility::Payload* CreatePayloadType( + virtual RtpUtility::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 OVERRIDE { - ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; + RtpUtility::Payload* payload = new RtpUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); assert(frequency >= 1000); @@ -401,8 +397,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { return payload; } - int GetPayloadTypeFrequency( - const ModuleRTPUtility::Payload& payload) const { + int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const { return payload.typeSpecific.Audio.frequency; } }; @@ -411,39 +406,37 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy { public: virtual bool CodecsMustBeUnique() const OVERRIDE { return false; } - virtual bool PayloadIsCompatible( - const ModuleRTPUtility::Payload& payload, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate) const OVERRIDE { + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const OVERRIDE { return !payload.audio; } - virtual void UpdatePayloadRate( - ModuleRTPUtility::Payload* payload, - const uint32_t rate) const OVERRIDE { + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const OVERRIDE { payload->typeSpecific.Video.maxRate = rate; } - virtual ModuleRTPUtility::Payload* CreatePayloadType( + virtual RtpUtility::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 OVERRIDE { RtpVideoCodecTypes videoType = kRtpVideoGeneric; - if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) { + if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { videoType = kRtpVideoVp8; - } else if (ModuleRTPUtility::StringCompare(payloadName, "H264", 4)) { + } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { videoType = kRtpVideoH264; - } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) { + } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { videoType = kRtpVideoGeneric; - } else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6)) { + } else if (RtpUtility::StringCompare(payloadName, "ULPFEC", 6)) { videoType = kRtpVideoNone; } else { videoType = kRtpVideoGeneric; } - ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; + RtpUtility::Payload* payload = new RtpUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); @@ -453,8 +446,7 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy { return payload; } - int GetPayloadTypeFrequency( - const ModuleRTPUtility::Payload& payload) const { + int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const { return kVideoPayloadTypeFrequency; } }; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc index c03ffcd1f3..2dacbdd142 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc @@ -36,18 +36,19 @@ class RtpPayloadRegistryTest : public ::testing::Test { } protected: - ModuleRTPUtility::Payload* ExpectReturnOfTypicalAudioPayload( - uint8_t payload_type, uint32_t rate) { + RtpUtility::Payload* ExpectReturnOfTypicalAudioPayload(uint8_t payload_type, + uint32_t rate) { bool audio = true; - ModuleRTPUtility::Payload returned_payload = { "name", audio, { - // Initialize the audio struct in this case. - { kTypicalFrequency, kTypicalChannels, rate } - }}; + RtpUtility::Payload returned_payload = { + "name", + audio, + {// Initialize the audio struct in this case. + {kTypicalFrequency, kTypicalChannels, rate}}}; // Note: we return a new payload since the payload registry takes ownership // of the created object. - ModuleRTPUtility::Payload* returned_payload_on_heap = - new ModuleRTPUtility::Payload(returned_payload); + RtpUtility::Payload* returned_payload_on_heap = + new RtpUtility::Payload(returned_payload); EXPECT_CALL(*mock_payload_strategy_, CreatePayloadType(kTypicalPayloadName, payload_type, kTypicalFrequency, @@ -62,7 +63,7 @@ class RtpPayloadRegistryTest : public ::testing::Test { TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) { uint8_t payload_type = 97; - ModuleRTPUtility::Payload* returned_payload_on_heap = + RtpUtility::Payload* returned_payload_on_heap = ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate); bool new_payload_created = false; @@ -72,7 +73,7 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) { EXPECT_TRUE(new_payload_created) << "A new payload WAS created."; - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type, retrieved_payload)); @@ -99,7 +100,7 @@ TEST_F(RtpPayloadRegistryTest, DoesNotCreateNewPayloadTypeIfRed) { ASSERT_EQ(red_type_of_the_day, rtp_payload_registry_->red_payload_type()); - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(red_type_of_the_day, retrieved_payload)); EXPECT_FALSE(retrieved_payload->audio); @@ -111,7 +112,7 @@ TEST_F(RtpPayloadRegistryTest, uint8_t payload_type = 97; bool ignored = false; - ModuleRTPUtility::Payload* first_payload_on_heap = + RtpUtility::Payload* first_payload_on_heap = ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate); EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload( kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels, @@ -121,7 +122,7 @@ TEST_F(RtpPayloadRegistryTest, kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels, kTypicalRate, &ignored)) << "Adding same codec twice = bad."; - ModuleRTPUtility::Payload* second_payload_on_heap = + RtpUtility::Payload* second_payload_on_heap = ExpectReturnOfTypicalAudioPayload(payload_type - 1, kTypicalRate); EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload( kTypicalPayloadName, payload_type - 1, kTypicalFrequency, @@ -129,7 +130,7 @@ TEST_F(RtpPayloadRegistryTest, "With a different payload type is fine though."; // Ensure both payloads are preserved. - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type, retrieved_payload)); EXPECT_EQ(first_payload_on_heap, retrieved_payload); @@ -168,7 +169,7 @@ TEST_F(RtpPayloadRegistryTest, kTypicalPayloadName, payload_type - 1, kTypicalFrequency, kTypicalChannels, kTypicalRate, &ignored)); - ModuleRTPUtility::Payload* retrieved_payload = NULL; + RtpUtility::Payload* retrieved_payload = NULL; EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload( payload_type, retrieved_payload)) << "The first payload should be " "deregistered because the only thing that differs is payload type."; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc index c8104cc373..05eefbe086 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -159,10 +159,10 @@ int32_t RTPReceiverAudio::OnNewPayloadTypeCreated( uint32_t frequency) { CriticalSectionScoped lock(crit_sect_.get()); - if (ModuleRTPUtility::StringCompare(payload_name, "telephone-event", 15)) { + if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) { telephone_event_payload_type_ = payload_type; } - if (ModuleRTPUtility::StringCompare(payload_name, "cn", 2)) { + if (RtpUtility::StringCompare(payload_name, "cn", 2)) { // we can have three CNG on 8000Hz, 16000Hz and 32000Hz if (frequency == 8000) { cng_nb_payload_type_ = payload_type; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h index 0ffd4bf4be..4fb7256d48 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -83,7 +83,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy, // 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). void PossiblyRemoveExistingPayloadType( - ModuleRTPUtility::PayloadTypeMap* payload_type_map, + RtpUtility::PayloadTypeMap* payload_type_map, const char payload_name[RTP_PAYLOAD_NAME_SIZE], size_t payload_name_length, uint32_t frequency, diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc index d92618f2d5..7493488db8 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -22,10 +22,10 @@ namespace webrtc { -using ModuleRTPUtility::GetCurrentRTP; -using ModuleRTPUtility::Payload; -using ModuleRTPUtility::RTPPayloadParser; -using ModuleRTPUtility::StringCompare; +using RtpUtility::GetCurrentRTP; +using RtpUtility::Payload; +using RtpUtility::RTPPayloadParser; +using RtpUtility::StringCompare; RtpReceiver* RtpReceiver::CreateVideoReceiver( int id, Clock* clock, diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc index 261599163e..c058ed5daa 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -129,12 +129,11 @@ int32_t RTPReceiverVideo::BuildRTPheader( if (rtp_header->header.markerBit) { data_buffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1 } - ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + 2, - rtp_header->header.sequenceNumber); - ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 4, - rtp_header->header.timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 8, - rtp_header->header.ssrc); + RtpUtility::AssignUWord16ToBuffer(data_buffer + 2, + rtp_header->header.sequenceNumber); + RtpUtility::AssignUWord32ToBuffer(data_buffer + 4, + rtp_header->header.timestamp); + RtpUtility::AssignUWord32ToBuffer(data_buffer + 8, rtp_header->header.ssrc); int32_t rtp_header_length = 12; @@ -146,8 +145,7 @@ int32_t RTPReceiverVideo::BuildRTPheader( } uint8_t* ptr = &data_buffer[rtp_header_length]; for (uint32_t i = 0; i < rtp_header->header.numCSRCs; ++i) { - ModuleRTPUtility::AssignUWord32ToBuffer(ptr, - rtp_header->header.arrOfCSRCs[i]); + RtpUtility::AssignUWord32ToBuffer(ptr, rtp_header->header.arrOfCSRCs[i]); ptr += 4; } data_buffer[0] = (data_buffer[0] & 0xf0) | rtp_header->header.numCSRCs; @@ -160,8 +158,8 @@ int32_t RTPReceiverVideo::BuildRTPheader( int32_t RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_data_length) { - ModuleRTPUtility::RTPPayload parsed_packet; - ModuleRTPUtility::RTPPayloadParser rtp_payload_parser( + RtpUtility::RTPPayload parsed_packet; + RtpUtility::RTPPayloadParser rtp_payload_parser( kRtpVideoVp8, payload_data, payload_data_length); if (!rtp_payload_parser.Parse(parsed_packet)) @@ -170,11 +168,12 @@ int32_t RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header, if (parsed_packet.info.VP8.dataLength == 0) return 0; - rtp_header->frameType = (parsed_packet.frameType == ModuleRTPUtility::kIFrame) - ? kVideoFrameKey : kVideoFrameDelta; + rtp_header->frameType = (parsed_packet.frameType == RtpUtility::kIFrame) + ? kVideoFrameKey + : kVideoFrameDelta; RTPVideoHeaderVP8* to_header = &rtp_header->type.Video.codecHeader.VP8; - ModuleRTPUtility::RTPPayloadVP8* from_header = &parsed_packet.info.VP8; + RtpUtility::RTPPayloadVP8* from_header = &parsed_packet.info.VP8; rtp_header->type.Video.isFirstPacket = from_header->beginningOfPartition && (from_header->partitionID == 0); 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 eb76cfe7c9..930778c3b0 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -232,8 +232,9 @@ class RtpSendingTestTransport : public Transport { virtual int SendPacket(int channel, const void* data, int length) { RTPHeader header; scoped_ptr parser(RtpHeaderParser::Create()); - EXPECT_TRUE( - parser->Parse(static_cast(data), length, &header)); + EXPECT_TRUE(parser->Parse(static_cast(data), + static_cast(length), + &header)); bytes_received_[header.ssrc] += length; ++packets_received_[header.ssrc]; return length; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc index 919eb029c9..dc9f4b1d19 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc @@ -119,7 +119,7 @@ RTPSender::~RTPSender() { SSRCDatabase::ReturnSSRCDatabase(); delete send_critsect_; while (!payload_type_map_.empty()) { - std::map::iterator it = + std::map::iterator it = payload_type_map_.begin(); delete it->second; payload_type_map_.erase(it); @@ -224,17 +224,17 @@ int32_t RTPSender::RegisterPayload( assert(payload_name); CriticalSectionScoped cs(send_critsect_); - std::map::iterator it = + std::map::iterator it = payload_type_map_.find(payload_number); if (payload_type_map_.end() != it) { // We already use this payload type. - ModuleRTPUtility::Payload *payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); // Check if it's the same as we already have. - if (ModuleRTPUtility::StringCompare(payload->name, payload_name, - RTP_PAYLOAD_NAME_SIZE - 1)) { + if (RtpUtility::StringCompare( + payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) { if (audio_configured_ && payload->audio && payload->typeSpecific.Audio.frequency == frequency && (payload->typeSpecific.Audio.rate == rate || @@ -250,7 +250,7 @@ int32_t RTPSender::RegisterPayload( return -1; } int32_t ret_val = -1; - ModuleRTPUtility::Payload *payload = NULL; + RtpUtility::Payload* payload = NULL; if (audio_configured_) { ret_val = audio_->RegisterAudioPayload(payload_name, payload_number, frequency, channels, rate, payload); @@ -268,13 +268,13 @@ int32_t RTPSender::DeRegisterSendPayload( const int8_t payload_type) { CriticalSectionScoped lock(send_critsect_); - std::map::iterator it = + std::map::iterator it = payload_type_map_.find(payload_type); if (payload_type_map_.end() == it) { return -1; } - ModuleRTPUtility::Payload *payload = it->second; + RtpUtility::Payload* payload = it->second; delete payload; payload_type_map_.erase(it); return 0; @@ -376,14 +376,14 @@ int32_t RTPSender::CheckPayloadType(const int8_t payload_type, } return 0; } - std::map::iterator it = + std::map::iterator it = payload_type_map_.find(payload_type); if (it == payload_type_map_.end()) { LOG(LS_WARNING) << "Payload type " << payload_type << " not registered."; return -1; } payload_type_ = payload_type; - ModuleRTPUtility::Payload *payload = it->second; + RtpUtility::Payload* payload = it->second; assert(payload); if (!payload->audio && !audio_configured_) { video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType); @@ -465,7 +465,7 @@ int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) { } if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false)) return -1; - ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length); + RtpUtility::RtpHeaderParser rtp_parser(buffer, length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); bytes_left -= length - rtp_header.headerLength; @@ -614,7 +614,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) { } if (paced_sender_) { - ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length); + RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length); RTPHeader header; if (!rtp_parser.Parse(header)) { assert(false); @@ -813,7 +813,7 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, bool is_retransmit) { uint8_t *buffer_to_send_ptr = buffer; - ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length); + RtpUtility::RtpHeaderParser rtp_parser(buffer, length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); TRACE_EVENT_INSTANT2("webrtc_rtp", "PrepareAndSendPacket", @@ -925,13 +925,13 @@ int RTPSender::TimeToSendPadding(int bytes) { return bytes_sent; } -// TODO(pwestin): send in the RTPHeaderParser to avoid parsing it again. +// TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again. int32_t RTPSender::SendToNetwork( uint8_t *buffer, int payload_length, int rtp_header_length, int64_t capture_time_ms, StorageType storage, PacedSender::Priority priority) { - ModuleRTPUtility::RTPHeaderParser rtp_parser( - buffer, payload_length + rtp_header_length); + RtpUtility::RtpHeaderParser rtp_parser(buffer, + payload_length + rtp_header_length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); @@ -1046,9 +1046,9 @@ int RTPSender::CreateRTPHeader( if (marker_bit) { header[1] |= kRtpMarkerBitMask; // Marker bit is set. } - ModuleRTPUtility::AssignUWord16ToBuffer(header + 2, sequence_number); - ModuleRTPUtility::AssignUWord32ToBuffer(header + 4, timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(header + 8, ssrc); + RtpUtility::AssignUWord16ToBuffer(header + 2, sequence_number); + RtpUtility::AssignUWord32ToBuffer(header + 4, timestamp); + RtpUtility::AssignUWord32ToBuffer(header + 8, ssrc); int32_t rtp_header_length = 12; // Add the CSRCs if any. @@ -1060,7 +1060,7 @@ int RTPSender::CreateRTPHeader( } uint8_t *ptr = &header[rtp_header_length]; for (int i = 0; i < num_csrcs; ++i) { - ModuleRTPUtility::AssignUWord32ToBuffer(ptr, csrcs[i]); + RtpUtility::AssignUWord32ToBuffer(ptr, csrcs[i]); ptr += 4; } header[0] = (header[0] & 0xf0) | num_csrcs; @@ -1123,8 +1123,7 @@ uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const { const uint32_t kHeaderLength = kRtpOneByteHeaderLength; // Add extension ID (0xBEDE). - ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer, - kRtpOneByteHeaderExtensionId); + RtpUtility::AssignUWord16ToBuffer(data_buffer, kRtpOneByteHeaderExtensionId); // Add extensions. uint16_t total_block_length = 0; @@ -1157,8 +1156,8 @@ uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const { } // Set header length (in number of Word32, header excluded). assert(total_block_length % 4 == 0); - ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + kPosLength, - total_block_length / 4); + RtpUtility::AssignUWord16ToBuffer(data_buffer + kPosLength, + total_block_length / 4); // Total added length. return kHeaderLength + total_block_length; } @@ -1192,8 +1191,8 @@ uint8_t RTPSender::BuildTransmissionTimeOffsetExtension( size_t pos = 0; const uint8_t len = 2; data_buffer[pos++] = (id << 4) + len; - ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos, - transmission_time_offset_); + RtpUtility::AssignUWord24ToBuffer(data_buffer + pos, + transmission_time_offset_); pos += 3; assert(pos == kTransmissionTimeOffsetLength); return kTransmissionTimeOffsetLength; @@ -1261,8 +1260,7 @@ uint8_t RTPSender::BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const { size_t pos = 0; const uint8_t len = 2; data_buffer[pos++] = (id << 4) + len; - ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos, - absolute_send_time_); + RtpUtility::AssignUWord24ToBuffer(data_buffer + pos, absolute_send_time_); pos += 3; assert(pos == kAbsoluteSendTimeLength); return kAbsoluteSendTimeLength; @@ -1310,8 +1308,8 @@ void RTPSender::UpdateTransmissionTimeOffset( return; } // Update transmission offset field (converting to a 90 kHz timestamp). - ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, - time_diff_ms * 90); // RTP timestamp. + RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, + time_diff_ms * 90); // RTP timestamp. } bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet, @@ -1398,14 +1396,14 @@ void RTPSender::UpdateAbsoluteSendTime( } // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit // fractional part). - ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, - ((now_ms << 18) / 1000) & 0x00ffffff); + RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, + ((now_ms << 18) / 1000) & 0x00ffffff); } void RTPSender::SetSendingStatus(bool enabled) { if (enabled) { uint32_t frequency_hz = SendPayloadFrequency(); - uint32_t RTPtime = ModuleRTPUtility::GetCurrentRTP(clock_, frequency_hz); + uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz); // Will be ignored if it's already configured via API. SetStartTimestamp(RTPtime, false); @@ -1630,8 +1628,8 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length, CriticalSectionScoped cs(send_critsect_); uint8_t* data_buffer_rtx = buffer_rtx; // Add RTX header. - ModuleRTPUtility::RTPHeaderParser rtp_parser( - reinterpret_cast(buffer), *length); + RtpUtility::RtpHeaderParser rtp_parser( + reinterpret_cast(buffer), *length); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); @@ -1648,15 +1646,15 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length, // Replace sequence number. uint8_t *ptr = data_buffer_rtx + 2; - ModuleRTPUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++); + RtpUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++); // Replace SSRC. ptr += 6; - ModuleRTPUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_); + RtpUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_); // Add OSN (original sequence number). ptr = data_buffer_rtx + rtp_header.headerLength; - ModuleRTPUtility::AssignUWord16ToBuffer(ptr, rtp_header.sequenceNumber); + RtpUtility::AssignUWord16ToBuffer(ptr, rtp_header.sequenceNumber); ptr += 2; // Add original payload data. diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h index 78d3a2a8a0..d7ebc7ddd1 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h @@ -359,7 +359,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer { uint16_t packet_over_head_; int8_t payload_type_ GUARDED_BY(send_critsect_); - std::map payload_type_map_; + std::map payload_type_map_; RtpHeaderExtensionMap rtp_header_extension_map_; int32_t transmission_time_offset_; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc index 6b3e2276ee..99c008513f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -89,10 +89,10 @@ int32_t RTPSenderAudio::RegisterAudioPayload( const uint32_t frequency, const uint8_t channels, const uint32_t rate, - ModuleRTPUtility::Payload*& payload) { + RtpUtility::Payload*& payload) { CriticalSectionScoped cs(_sendAudioCritsect); - if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) { + if (RtpUtility::StringCompare(payloadName, "cn", 2)) { // we can have multiple CNG payload types if (frequency == 8000) { _cngNBPayloadType = payloadType; @@ -110,14 +110,14 @@ int32_t RTPSenderAudio::RegisterAudioPayload( return -1; } } - if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) { + if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) { // Don't add it to the list // we dont want to allow send with a DTMF payloadtype _dtmfPayloadType = payloadType; return 0; // The default timestamp rate is 8000 Hz, but other rates may be defined. } - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; payload->typeSpecific.Audio.frequency = frequency; payload->typeSpecific.Audio.channels = channels; payload->typeSpecific.Audio.rate = rate; @@ -388,8 +388,8 @@ int32_t RTPSenderAudio::SendAudio( return -1; } uint32_t REDheader = (timestampOffset << 10) + blockLength; - ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength, - REDheader); + RtpUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength, + REDheader); rtpHeaderLength += 3; dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0]; @@ -436,7 +436,7 @@ int32_t RTPSenderAudio::SendAudio( // Update audio level extension, if included. { uint16_t packetSize = payloadSize + rtpHeaderLength; - ModuleRTPUtility::RTPHeaderParser rtp_parser(dataBuffer, packetSize); + RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); RTPHeader rtp_header; rtp_parser.Parse(rtp_header); _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header, @@ -558,7 +558,7 @@ RTPSenderAudio::SendTelephoneEventPacket(const bool ended, // First byte is Event number, equals key number dtmfbuffer[12] = _dtmfKey; dtmfbuffer[13] = E|R|volume; - ModuleRTPUtility::AssignUWord16ToBuffer(dtmfbuffer+14, duration); + RtpUtility::AssignUWord16ToBuffer(dtmfbuffer + 14, duration); _sendAudioCritsect->Leave(); TRACE_EVENT_INSTANT2("webrtc_rtp", diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h index 732199c17a..d3f67e5edb 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h @@ -26,13 +26,12 @@ public: RTPSender* rtpSender); virtual ~RTPSenderAudio(); - int32_t RegisterAudioPayload( - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate, - ModuleRTPUtility::Payload*& payload); + int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate, + RtpUtility::Payload*& payload); int32_t SendAudio(const FrameType frameType, const int8_t payloadType, diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc index 18482890f7..9ba60b1bf1 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -204,7 +204,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) { EXPECT_EQ(kRtpHeaderSize, length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; const bool valid_rtp_header = rtp_parser.Parse(rtp_header, NULL); @@ -235,7 +235,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; @@ -276,7 +276,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; @@ -306,7 +306,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; @@ -344,7 +344,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; // Updating audio level is done in RTPSenderAudio, so simulate it here. @@ -394,7 +394,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) { length); // Verify - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); webrtc::RTPHeader rtp_header; // Updating audio level is done in RTPSenderAudio, so simulate it here. @@ -471,8 +471,8 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) { EXPECT_EQ(1, transport_.packets_sent_); EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); // Parse sent packet. - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser( - transport_.last_sent_packet_, rtp_length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + rtp_length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, @@ -533,8 +533,8 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) { EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_); // Parse sent packet. - webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser( - transport_.last_sent_packet_, rtp_length); + webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + rtp_length); webrtc::RTPHeader rtp_header; RtpHeaderExtensionMap map; map.Register(kRtpExtensionTransmissionTimeOffset, @@ -744,8 +744,8 @@ TEST_F(RtpSenderTest, SendGenericVideo) { 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser.Parse(rtp_header)); @@ -770,8 +770,8 @@ TEST_F(RtpSenderTest, SendGenericVideo) { 1234, 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser2(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); ASSERT_TRUE(rtp_parser.Parse(rtp_header)); payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_); @@ -1018,8 +1018,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) { 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser.Parse(rtp_header)); @@ -1047,8 +1047,8 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { 4321, payload, sizeof(payload), NULL)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_, - transport_.last_sent_packet_len_); + RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, + transport_.last_sent_packet_len_); webrtc::RTPHeader rtp_header; ASSERT_TRUE(rtp_parser.Parse(rtp_header)); diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc index e4138ff556..ea5f7a7e3a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -84,20 +84,20 @@ int32_t RTPSenderVideo::RegisterVideoPayload( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t maxBitRate, - ModuleRTPUtility::Payload*& payload) { + RtpUtility::Payload*& payload) { CriticalSectionScoped cs(_sendVideoCritsect); RtpVideoCodecTypes videoType = kRtpVideoGeneric; - if (ModuleRTPUtility::StringCompare(payloadName, "VP8",3)) { + if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { videoType = kRtpVideoVp8; - } else if (ModuleRTPUtility::StringCompare(payloadName, "H264", 4)) { + } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { videoType = kRtpVideoH264; - } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) { + } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { videoType = kRtpVideoGeneric; } else { videoType = kRtpVideoGeneric; } - payload = new ModuleRTPUtility::Payload; + payload = new RtpUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); payload->typeSpecific.Video.videoCodecType = videoType; @@ -213,7 +213,7 @@ RTPSenderVideo::SendRTPIntraRequest() data[2] = 0; data[3] = 1; // length - ModuleRTPUtility::AssignUWord32ToBuffer(data+4, _rtpSender.SSRC()); + RtpUtility::AssignUWord32ToBuffer(data + 4, _rtpSender.SSRC()); TRACE_EVENT_INSTANT1("webrtc_rtp", "Video::IntraRequest", diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h index daa730e8c2..82bd1de862 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h @@ -39,11 +39,10 @@ public: uint16_t FECPacketOverhead() const; - int32_t RegisterVideoPayload( - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t maxBitRate, - ModuleRTPUtility::Payload*& payload); + int32_t RegisterVideoPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t maxBitRate, + RtpUtility::Payload*& payload); int32_t SendVideo(const RtpVideoCodecTypes videoType, const FrameType frameType, diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc index c1f3c64274..95389b46af 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc @@ -66,7 +66,7 @@ ReceiveStatistics* NullObjectReceiveStatistics() { return &null_receive_statistics; } -namespace ModuleRTPUtility { +namespace RtpUtility { enum { kRtcpExpectedVersion = 2, @@ -215,16 +215,16 @@ void RTPPayload::SetType(RtpVideoCodecTypes videoType) { } } -RTPHeaderParser::RTPHeaderParser(const uint8_t* rtpData, - const uint32_t rtpDataLength) - : _ptrRTPDataBegin(rtpData), - _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) { +RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData, + const size_t rtpDataLength) + : _ptrRTPDataBegin(rtpData), + _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) { } -RTPHeaderParser::~RTPHeaderParser() { +RtpHeaderParser::~RtpHeaderParser() { } -bool RTPHeaderParser::RTCP() const { +bool RtpHeaderParser::RTCP() const { // 72 to 76 is reserved for RTP // 77 to 79 is not reserver but they are not assigned we will block them // for RTCP 200 SR == marker bit + 72 @@ -299,7 +299,7 @@ bool RTPHeaderParser::RTCP() const { return RTCP; } -bool RTPHeaderParser::ParseRtcp(RTPHeader* header) const { +bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const { assert(header != NULL); const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin; @@ -328,7 +328,7 @@ bool RTPHeaderParser::ParseRtcp(RTPHeader* header) const { return true; } -bool RTPHeaderParser::Parse(RTPHeader& header, +bool RtpHeaderParser::Parse(RTPHeader& header, RtpHeaderExtensionMap* ptrExtensionMap) const { const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin; if (length < kRtpMinParseLength) { @@ -441,7 +441,7 @@ bool RTPHeaderParser::Parse(RTPHeader& header, return true; } -void RTPHeaderParser::ParseOneByteExtensionHeader( +void RtpHeaderParser::ParseOneByteExtensionHeader( RTPHeader& header, const RtpHeaderExtensionMap* ptrExtensionMap, const uint8_t* ptrRTPDataExtensionEnd, @@ -552,10 +552,9 @@ void RTPHeaderParser::ParseOneByteExtensionHeader( } } -uint8_t RTPHeaderParser::ParsePaddingBytes( - const uint8_t* ptrRTPDataExtensionEnd, - const uint8_t* ptr) const { - +uint8_t RtpHeaderParser::ParsePaddingBytes( + const uint8_t* ptrRTPDataExtensionEnd, + const uint8_t* ptr) const { uint8_t num_zero_bytes = 0; while (ptrRTPDataExtensionEnd - ptr > 0) { if (*ptr != 0) { @@ -768,6 +767,6 @@ int RTPPayloadParser::ParseVP8TIDAndKeyIdx(RTPPayloadVP8* vp8, return 0; } -} // namespace ModuleRTPUtility +} // namespace RtpUtility } // namespace webrtc diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h index 732301f6fb..ef50570d28 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h @@ -28,8 +28,7 @@ RtpFeedback* NullObjectRtpFeedback(); RtpAudioFeedback* NullObjectRtpAudioFeedback(); ReceiveStatistics* NullObjectReceiveStatistics(); -namespace ModuleRTPUtility -{ +namespace RtpUtility { // January 1970, in NTP seconds. const uint32_t NTP_JAN_1970 = 2208988800UL; @@ -92,12 +91,10 @@ namespace ModuleRTPUtility */ uint32_t BufferToUWord32(const uint8_t* dataBuffer); - class RTPHeaderParser - { + class RtpHeaderParser { public: - RTPHeaderParser(const uint8_t* rtpData, - const uint32_t rtpDataLength); - ~RTPHeaderParser(); + RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength); + ~RtpHeaderParser(); bool RTCP() const; bool ParseRtcp(RTPHeader* header) const; @@ -207,7 +204,7 @@ namespace ModuleRTPUtility const RtpVideoCodecTypes _videoType; }; -} // namespace ModuleRTPUtility + } // namespace RtpUtility } // namespace webrtc diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc index d33eaf4c84..27170d5526 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc @@ -8,9 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ - /* - * This file conatins unit tests for the ModuleRTPUtility. + * This file conatins unit tests for the RtpUtility. */ #include "testing/gtest/include/gtest/gtest.h" @@ -20,9 +19,9 @@ namespace webrtc { -using ModuleRTPUtility::RTPPayloadParser; -using ModuleRTPUtility::RTPPayload; -using ModuleRTPUtility::RTPPayloadVP8; +using RtpUtility::RTPPayloadParser; +using RtpUtility::RTPPayload; +using RtpUtility::RTPPayloadVP8; // Payload descriptor // 0 1 2 3 4 5 6 7 @@ -81,7 +80,7 @@ TEST(ParseVP8Test, BasicHeader) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 4 /*PartID*/); @@ -102,7 +101,7 @@ TEST(ParseVP8Test, PictureID) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 1 /*N*/, 0 /*S*/, 0 /*PartID*/); @@ -141,7 +140,7 @@ TEST(ParseVP8Test, Tl0PicIdx) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kIFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 0 /*PartID*/); @@ -164,7 +163,7 @@ TEST(ParseVP8Test, TIDAndLayerSync) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/); @@ -188,7 +187,7 @@ TEST(ParseVP8Test, KeyIdx) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/); @@ -214,7 +213,7 @@ TEST(ParseVP8Test, MultipleExtensions) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kPFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/); @@ -263,7 +262,7 @@ TEST(ParseVP8Test, TestWithPacketizer) { RTPPayload parsedPacket; ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket)); - EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType); + EXPECT_EQ(RtpUtility::kIFrame, parsedPacket.frameType); EXPECT_EQ(kRtpVideoVp8, parsedPacket.type); VerifyBasicHeader(parsedPacket.info.VP8, diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h index 8061ce018c..1c6b88385e 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h @@ -52,7 +52,9 @@ class LoopBackTransport : public webrtc::Transport { } RTPHeader header; scoped_ptr parser(RtpHeaderParser::Create()); - if (!parser->Parse(static_cast(data), len, &header)) { + if (!parser->Parse(static_cast(data), + static_cast(len), + &header)) { return -1; } PayloadUnion payload_specific; diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc index 94d1e52ef5..4c4944d9f8 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc @@ -83,11 +83,9 @@ class RtpRtcpVideoTest : public ::testing::Test { uint32_t sequence_number) { dataBuffer[0] = static_cast(0x80); // version 2 dataBuffer[1] = static_cast(kPayloadType); - ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + 2, - sequence_number); - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp); - ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer + 8, - 0x1234); // SSRC. + RtpUtility::AssignUWord16ToBuffer(dataBuffer + 2, sequence_number); + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp); + RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, 0x1234); // SSRC. int32_t rtpHeaderLength = 12; return rtpHeaderLength; } diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc index 83978e47b0..bdb53afebc 100644 --- a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc +++ b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc @@ -259,12 +259,10 @@ TEST(FecTest, FecTest) { // Only push one (fake) frame to the FEC. mediaPacket->data[1] &= 0x7f; - ModuleRTPUtility::AssignUWord16ToBuffer(&mediaPacket->data[2], - seqNum); - ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[4], - timeStamp); - ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[8], - ssrc); + RtpUtility::AssignUWord16ToBuffer(&mediaPacket->data[2], seqNum); + RtpUtility::AssignUWord32ToBuffer(&mediaPacket->data[4], + timeStamp); + RtpUtility::AssignUWord32ToBuffer(&mediaPacket->data[8], ssrc); // Generate random values for payload for (int32_t j = 12; j < mediaPacket->length; ++j) { mediaPacket->data[j] = static_cast(rand() % 256); @@ -303,7 +301,7 @@ TEST(FecTest, FecTest) { memcpy(receivedPacket->pkt->data, mediaPacket->data, mediaPacket->length); receivedPacket->seq_num = - ModuleRTPUtility::BufferToUWord16(&mediaPacket->data[2]); + RtpUtility::BufferToUWord16(&mediaPacket->data[2]); receivedPacket->is_fec = false; } mediaPacketIdx++; diff --git a/webrtc/modules/utility/source/rtp_dump_impl.h b/webrtc/modules/utility/source/rtp_dump_impl.h index 04ae7dfca2..ff3f07cec7 100644 --- a/webrtc/modules/utility/source/rtp_dump_impl.h +++ b/webrtc/modules/utility/source/rtp_dump_impl.h @@ -35,7 +35,7 @@ private: inline uint16_t RtpDumpHtons(uint16_t x) const; // Return true if the packet starts with a valid RTCP header. - // Note: See ModuleRTPUtility::RTPHeaderParser::RTCP() for details on how + // Note: See RtpUtility::RtpHeaderParser::RTCP() for details on how // to determine if the packet is an RTCP packet. bool RTCP(const uint8_t* packet) const; diff --git a/webrtc/modules/video_coding/main/test/pcap_file_reader.cc b/webrtc/modules/video_coding/main/test/pcap_file_reader.cc index 3d4e2659df..68c856652d 100644 --- a/webrtc/modules/video_coding/main/test/pcap_file_reader.cc +++ b/webrtc/modules/video_coding/main/test/pcap_file_reader.cc @@ -268,8 +268,7 @@ class PcapFileReaderImpl : public RtpPacketSourceInterface { } TRY(Read(read_buffer_, marker.payload_length)); - ModuleRTPUtility::RTPHeaderParser rtp_parser(read_buffer_, - marker.payload_length); + RtpUtility::RtpHeaderParser rtp_parser(read_buffer_, marker.payload_length); if (rtp_parser.RTCP()) { rtp_parser.ParseRtcp(&marker.rtp_header); packets_.push_back(marker); diff --git a/webrtc/modules/video_coding/main/test/pcap_file_reader_unittest.cc b/webrtc/modules/video_coding/main/test/pcap_file_reader_unittest.cc index 1810071461..c6f1d511c1 100644 --- a/webrtc/modules/video_coding/main/test/pcap_file_reader_unittest.cc +++ b/webrtc/modules/video_coding/main/test/pcap_file_reader_unittest.cc @@ -55,7 +55,7 @@ class TestPcapFileReader : public ::testing::Test { EXPECT_GE(kBufferSize, length); length = kBufferSize; - ModuleRTPUtility::RTPHeaderParser rtp_header_parser(data, length); + RtpUtility::RtpHeaderParser rtp_header_parser(data, length); webrtc::RTPHeader header; if (!rtp_header_parser.RTCP() && rtp_header_parser.Parse(header, NULL)) { pps[header.ssrc]++; diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h index 670a29def2..11531b3b2d 100644 --- a/webrtc/test/rtp_rtcp_observer.h +++ b/webrtc/test/rtp_rtcp_observer.h @@ -129,7 +129,7 @@ class RtpRtcpObserver { private: virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { - EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, static_cast(length))); + EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length)); Action action; { CriticalSectionScoped lock(crit_); @@ -146,7 +146,7 @@ class RtpRtcpObserver { } virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { - EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, static_cast(length))); + EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length)); Action action; { CriticalSectionScoped lock(crit_); diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc index e45b549b16..95e1c7b7a3 100644 --- a/webrtc/video/call.cc +++ b/webrtc/video/call.cc @@ -100,8 +100,6 @@ class Call : public webrtc::Call, public PacketReceiver { std::map send_ssrcs_ GUARDED_BY(send_lock_); scoped_ptr send_lock_; - scoped_ptr rtp_header_parser_; - scoped_ptr overuse_observer_proxy_; VideoSendStream::RtpStateMap suspended_send_ssrcs_; @@ -133,7 +131,6 @@ Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config) : config_(config), receive_lock_(RWLockWrapper::CreateRWLock()), send_lock_(RWLockWrapper::CreateRWLock()), - rtp_header_parser_(RtpHeaderParser::Create()), video_engine_(video_engine), base_channel_id_(-1) { assert(video_engine != NULL); @@ -339,7 +336,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(const uint8_t* packet, PacketReceiver::DeliveryStatus Call::DeliverPacket(const uint8_t* packet, size_t length) { - if (RtpHeaderParser::IsRtcp(packet, static_cast(length))) + if (RtpHeaderParser::IsRtcp(packet, length)) return DeliverRtcp(packet, length); return DeliverRtp(packet, length); diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc index 18246168bc..62d2adcbab 100644 --- a/webrtc/video/call_perf_tests.cc +++ b/webrtc/video/call_perf_tests.cc @@ -354,7 +354,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, virtual Action OnSendRtp(const uint8_t* packet, size_t length) { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); if (!rtp_start_timestamp_set_) { // Calculate the rtp timestamp offset in order to calculate the real diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index 748eb71528..8946f8902c 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -254,8 +254,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE( - rtp_parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header)); // Never drop retransmitted packets. if (dropped_packets_.find(header.sequenceNumber) != @@ -342,7 +341,7 @@ TEST_F(EndToEndTest, DISABLED_CanReceiveFec) { virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE EXCLUSIVE_LOCKS_REQUIRED(crit_) { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_EQ(kRedPayloadType, header.payloadType); int encapsulated_payload_type = @@ -445,7 +444,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); if (header.timestamp == retransmitted_timestamp_) { EXPECT_EQ(retransmission_ssrc_, header.ssrc); @@ -632,7 +631,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); // Drop all retransmitted packets to force a PLI. if (header.timestamp <= highest_dropped_timestamp_) @@ -724,7 +723,7 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) { private: virtual DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE { - if (RtpHeaderParser::IsRtcp(packet, static_cast(length))) { + if (RtpHeaderParser::IsRtcp(packet, length)) { return receiver_->DeliverPacket(packet, length); } else { DeliveryStatus delivery_status = @@ -1188,7 +1187,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_TRUE(valid_ssrcs_[header.ssrc]) << "Received unknown SSRC: " << header.ssrc; @@ -1557,7 +1556,7 @@ TEST_F(EndToEndTest, RedundantPayloadsTransmittedOnAllSsrcs) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); if (!registered_rtx_ssrc_[header.ssrc]) return SEND_PACKET; @@ -1642,7 +1641,7 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); const uint32_t ssrc = header.ssrc; const uint16_t sequence_number = header.sequenceNumber; const uint32_t timestamp = header.timestamp; diff --git a/webrtc/video/full_stack.cc b/webrtc/video/full_stack.cc index e2a8e86351..284efe2037 100644 --- a/webrtc/video/full_stack.cc +++ b/webrtc/video/full_stack.cc @@ -104,7 +104,7 @@ class VideoAnalyzer : public PacketReceiver, size_t length) OVERRIDE { scoped_ptr parser(RtpHeaderParser::Create()); RTPHeader header; - parser->Parse(packet, static_cast(length), &header); + parser->Parse(packet, length, &header); { CriticalSectionScoped lock(crit_.get()); recv_times_[header.timestamp - rtp_timestamp_delta_] = @@ -143,7 +143,7 @@ class VideoAnalyzer : public PacketReceiver, virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { scoped_ptr parser(RtpHeaderParser::Create()); RTPHeader header; - parser->Parse(packet, static_cast(length), &header); + parser->Parse(packet, length, &header); { CriticalSectionScoped lock(crit_.get()); diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc index 5529f92ee6..c83d370948 100644 --- a/webrtc/video/rampup_tests.cc +++ b/webrtc/video/rampup_tests.cc @@ -125,7 +125,7 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { CriticalSectionScoped lock(crit_.get()); RTPHeader header; - EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header)); receive_stats_->IncomingPacket(header, length, false); payload_registry_->SetIncomingPayloadType(header); remote_bitrate_estimator_->IncomingPacket( @@ -151,8 +151,7 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { rtx_media_ssrcs_[header.ssrc], header); length = restored_length; - EXPECT_TRUE(rtp_parser_->Parse( - restored_packet, static_cast(length), &header)); + EXPECT_TRUE(rtp_parser_->Parse(restored_packet, length, &header)); } else { rtp_rtcp_->SetRemoteSSRC(header.ssrc); } @@ -291,7 +290,7 @@ class LowRateStreamObserver : public test::DirectTransport, size_t length) OVERRIDE { CriticalSectionScoped lock(crit_.get()); RTPHeader header; - EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header)); receive_stats_->IncomingPacket(header, length, false); remote_bitrate_estimator_->IncomingPacket( clock_->TimeInMilliseconds(), static_cast(length - 12), header); diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc index 254f6ee553..08c4fafa45 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc @@ -137,7 +137,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) { virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_FALSE(header.extension.hasTransmissionTimeOffset); EXPECT_TRUE(header.extension.hasAbsoluteSendTime); @@ -178,7 +178,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); EXPECT_TRUE(header.extension.hasTransmissionTimeOffset); EXPECT_FALSE(header.extension.hasAbsoluteSendTime); @@ -323,7 +323,7 @@ TEST_F(VideoSendStreamTest, SupportsFec) { private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); // Send lossy receive reports to trigger FEC enabling. if (send_count_++ % 2 != 0) { @@ -398,7 +398,7 @@ void VideoSendStreamTest::TestNackRetransmission( private: virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); // Nack second packet after receiving the third one. if (++send_count_ == 3) { @@ -724,7 +724,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) { CriticalSectionScoped lock(crit_.get()); ++rtp_count_; RTPHeader header; - EXPECT_TRUE(parser_->Parse(packet, static_cast(length), &header)); + EXPECT_TRUE(parser_->Parse(packet, length, &header)); last_sequence_number_ = header.sequenceNumber; if (test_state_ == kBeforeSuspend) { @@ -1041,11 +1041,11 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) { private: virtual DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE { - if (RtpHeaderParser::IsRtcp(packet, static_cast(length))) + if (RtpHeaderParser::IsRtcp(packet, length)) return DELIVERY_OK; RTPHeader header; - if (!parser_->Parse(packet, static_cast(length), &header)) + if (!parser_->Parse(packet, length, &header)) return DELIVERY_PACKET_ERROR; assert(stream_ != NULL); VideoSendStream::Stats stats = stream_->GetStats(); diff --git a/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc b/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc index 1e46b3d0ee..a2d060e7de 100644 --- a/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc +++ b/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc @@ -31,7 +31,7 @@ class RtcpCollectorTransport : public webrtc::Transport { } virtual int SendRTCPPacket(int channel, const void* data, int len) { const uint8_t* buf = static_cast(data); - webrtc::ModuleRTPUtility::RTPHeaderParser parser(buf, len); + webrtc::RtpUtility::RtpHeaderParser parser(buf, len); if (parser.RTCP()) { Packet p; p.channel = channel; diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc index cf33adf187..a678b13adb 100644 --- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc +++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc @@ -31,7 +31,9 @@ class ExtensionVerifyTransport : public webrtc::Transport { virtual int SendPacket(int channel, const void* data, int len) { webrtc::RTPHeader header; - if (parser_->Parse(static_cast(data), len, &header)) { + if (parser_->Parse(reinterpret_cast(data), + static_cast(len), + &header)) { bool ok = true; if (audio_level_id_ >= 0 && !header.extension.hasAudioLevel) {