c38eef896a
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I though that is more risky, so I'll do that in a separate patch later (perhaps we could purge the types from the whole module in one go?) BUG= TEST=Trybots, vie_ & voe_auto_test --automated Review URL: https://webrtc-codereview.appspot.com/998007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
347 lines
12 KiB
C++
347 lines
12 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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#include <cassert>
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#include <cmath>
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#include <map>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
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#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
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#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1
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namespace webrtc {
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class CriticalSectionWrapper;
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class PacedSender;
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class RTPPacketHistory;
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class RTPSenderAudio;
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class RTPSenderVideo;
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class RTPSenderInterface {
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public:
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RTPSenderInterface() {}
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virtual ~RTPSenderInterface() {}
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virtual WebRtc_UWord32 SSRC() const = 0;
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virtual WebRtc_UWord32 Timestamp() const = 0;
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virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
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const WebRtc_Word8 payloadType,
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const bool markerBit,
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const WebRtc_UWord32 captureTimeStamp,
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const bool timeStampProvided = true,
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const bool incSequenceNumber = true) = 0;
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virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
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virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
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virtual WebRtc_UWord16 SequenceNumber() const = 0;
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virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
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virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
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virtual WebRtc_UWord16 PacketOverHead() const = 0;
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virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
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virtual WebRtc_Word32 SendToNetwork(uint8_t* data_buffer,
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int payload_length,
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int rtp_header_length,
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int64_t capture_time_ms,
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StorageType storage) = 0;
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};
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class RTPSender : public Bitrate, public RTPSenderInterface {
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public:
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RTPSender(const WebRtc_Word32 id,
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const bool audio,
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RtpRtcpClock* clock,
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Transport* transport,
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RtpAudioFeedback* audio_feedback,
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PacedSender* paced_sender);
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virtual ~RTPSender();
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void ProcessBitrate();
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WebRtc_UWord16 ActualSendBitrateKbit() const;
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WebRtc_UWord32 VideoBitrateSent() const;
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WebRtc_UWord32 FecOverheadRate() const;
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WebRtc_UWord32 NackOverheadRate() const;
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void SetTargetSendBitrate(const WebRtc_UWord32 bits);
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WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
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WebRtc_Word32 RegisterPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
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WebRtc_Word8 SendPayloadType() const;
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int SendPayloadFrequency() const;
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void SetSendingStatus(const bool enabled);
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void SetSendingMediaStatus(const bool enabled);
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bool SendingMedia() const;
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// number of sent RTP packets
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WebRtc_UWord32 Packets() const;
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// number of sent RTP bytes
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WebRtc_UWord32 Bytes() const;
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void ResetDataCounters();
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WebRtc_UWord32 StartTimestamp() const;
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void SetStartTimestamp(WebRtc_UWord32 timestamp, bool force);
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WebRtc_UWord32 GenerateNewSSRC();
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void SetSSRC(const WebRtc_UWord32 ssrc);
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WebRtc_UWord16 SequenceNumber() const;
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void SetSequenceNumber(WebRtc_UWord16 seq);
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WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
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void SetCSRCStatus(const bool include);
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void SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength);
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WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
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const WebRtc_UWord16 packetOverHead);
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WebRtc_Word32 SendOutgoingData(const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation,
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VideoCodecInformation* codecInfo = NULL,
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const RTPVideoTypeHeader* rtpTypeHdr = NULL);
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WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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int64_t capture_time_ms,
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WebRtc_Word32 bytes);
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/*
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* RTP header extension
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*/
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WebRtc_Word32 SetTransmissionTimeOffset(
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const WebRtc_Word32 transmissionTimeOffset);
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WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
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const WebRtc_UWord8 id);
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WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
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WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
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WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const;
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WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
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WebRtc_UWord8* dataBuffer) const;
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bool UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
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const WebRtc_UWord16 rtp_packet_length,
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const WebRtcRTPHeader& rtp_header,
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const WebRtc_Word64 time_diff_ms) const;
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void TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms);
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/*
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* NACK
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*/
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int SelectiveRetransmissions() const;
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int SetSelectiveRetransmissions(uint8_t settings);
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void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
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const WebRtc_UWord16* nackSequenceNumbers,
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const WebRtc_UWord16 avgRTT);
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void SetStorePacketsStatus(const bool enable,
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const WebRtc_UWord16 numberToStore);
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bool StorePackets() const;
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WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
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WebRtc_UWord32 min_resend_time = 0);
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WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet,
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const WebRtc_UWord32 size);
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bool ProcessNACKBitRate(const WebRtc_UWord32 now);
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/*
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* RTX
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*/
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void SetRTXStatus(const bool enable,
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const bool setSSRC,
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const WebRtc_UWord32 SSRC);
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void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
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/*
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* Functions wrapping RTPSenderInterface
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*/
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virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
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const WebRtc_Word8 payloadType,
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const bool markerBit,
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const WebRtc_UWord32 captureTimeStamp,
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const bool timeStampProvided = true,
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const bool incSequenceNumber = true);
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virtual WebRtc_UWord16 RTPHeaderLength() const ;
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virtual WebRtc_UWord16 IncrementSequenceNumber();
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virtual WebRtc_UWord16 MaxPayloadLength() const;
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virtual WebRtc_UWord16 PacketOverHead() const;
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// current timestamp
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virtual WebRtc_UWord32 Timestamp() const;
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virtual WebRtc_UWord32 SSRC() const;
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virtual WebRtc_Word32 SendToNetwork(uint8_t* data_buffer,
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int payload_length,
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int rtp_header_length,
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int64_t capture_time_ms,
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StorageType storage);
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/*
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* Audio
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*/
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// Send a DTMF tone using RFC 2833 (4733)
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WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
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const WebRtc_UWord16 time_ms,
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const WebRtc_UWord8 level);
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bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
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// Set audio packet size, used to determine when it's time to send a DTMF
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// packet in silence (CNG)
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WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
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// Set status and ID for header-extension-for-audio-level-indication.
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WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
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const WebRtc_UWord8 ID);
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// Get status and ID for header-extension-for-audio-level-indication.
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WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
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WebRtc_UWord8& ID) const;
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// Store the audio level in dBov for
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// header-extension-for-audio-level-indication.
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WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
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// Set payload type for Redundant Audio Data RFC 2198
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WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
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// Get payload type for Redundant Audio Data RFC 2198
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WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
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/*
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* Video
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*/
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VideoCodecInformation* CodecInformationVideo();
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RtpVideoCodecTypes VideoCodecType() const;
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WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
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WebRtc_Word32 SendRTPIntraRequest();
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// FEC
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WebRtc_Word32 SetGenericFECStatus(const bool enable,
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const WebRtc_UWord8 payloadTypeRED,
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const WebRtc_UWord8 payloadTypeFEC);
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WebRtc_Word32 GenericFECStatus(bool& enable,
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WebRtc_UWord8& payloadTypeRED,
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WebRtc_UWord8& payloadTypeFEC) const;
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WebRtc_Word32 SetFecParameters(
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const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params);
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protected:
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WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
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RtpVideoCodecTypes& videoType);
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private:
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void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
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const WebRtc_UWord32 now);
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WebRtc_Word32 SendPaddingAccordingToBitrate(
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WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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int64_t capture_time_ms);
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WebRtc_Word32 _id;
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const bool _audioConfigured;
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RTPSenderAudio* _audio;
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RTPSenderVideo* _video;
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PacedSender* paced_sender_;
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CriticalSectionWrapper* _sendCritsect;
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Transport* _transport;
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bool _sendingMedia;
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WebRtc_UWord16 _maxPayloadLength;
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WebRtc_UWord16 _targetSendBitrate;
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WebRtc_UWord16 _packetOverHead;
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WebRtc_Word8 _payloadType;
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std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
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RtpHeaderExtensionMap _rtpHeaderExtensionMap;
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WebRtc_Word32 _transmissionTimeOffset;
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// NACK
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WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE];
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WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE];
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Bitrate _nackBitrate;
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RTPPacketHistory* _packetHistory;
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// Statistics
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WebRtc_UWord32 _packetsSent;
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WebRtc_UWord32 _payloadBytesSent;
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// RTP variables
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bool _startTimeStampForced;
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WebRtc_UWord32 _startTimeStamp;
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SSRCDatabase& _ssrcDB;
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WebRtc_UWord32 _remoteSSRC;
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bool _sequenceNumberForced;
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WebRtc_UWord16 _sequenceNumber;
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WebRtc_UWord16 _sequenceNumberRTX;
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bool _ssrcForced;
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WebRtc_UWord32 _ssrc;
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WebRtc_UWord32 _timeStamp;
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WebRtc_UWord8 _CSRCs;
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WebRtc_UWord32 _CSRC[kRtpCsrcSize];
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bool _includeCSRCs;
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bool _RTX;
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WebRtc_UWord32 _ssrcRTX;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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