717d147ebb
BUG=1811 TEST=vie_auto_test --automated, voe_auto_test --automated, trybots R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1768004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
411 lines
13 KiB
C++
411 lines
13 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
|
|
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
|
|
|
|
#include <list>
|
|
#include <vector>
|
|
|
|
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
|
|
#ifdef MATLAB
|
|
class MatlabPlot;
|
|
#endif
|
|
|
|
namespace webrtc {
|
|
|
|
class ModuleRtpRtcpImpl : public RtpRtcp {
|
|
public:
|
|
explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
|
|
|
|
virtual ~ModuleRtpRtcpImpl();
|
|
|
|
// Returns the number of milliseconds until the module want a worker thread to
|
|
// call Process.
|
|
virtual int32_t TimeUntilNextProcess();
|
|
|
|
// Process any pending tasks such as timeouts.
|
|
virtual int32_t Process();
|
|
|
|
// Receiver part.
|
|
|
|
// Called when we receive an RTCP packet.
|
|
virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
|
|
uint16_t incoming_packet_length);
|
|
|
|
virtual void SetRemoteSSRC(const uint32_t ssrc);
|
|
|
|
// Sender part.
|
|
|
|
virtual int32_t RegisterSendPayload(const CodecInst& voice_codec);
|
|
|
|
virtual int32_t RegisterSendPayload(const VideoCodec& video_codec);
|
|
|
|
virtual int32_t DeRegisterSendPayload(const int8_t payload_type);
|
|
|
|
virtual int8_t SendPayloadType() const;
|
|
|
|
// Register RTP header extension.
|
|
virtual int32_t RegisterSendRtpHeaderExtension(
|
|
const RTPExtensionType type,
|
|
const uint8_t id);
|
|
|
|
virtual int32_t DeregisterSendRtpHeaderExtension(
|
|
const RTPExtensionType type);
|
|
|
|
// Get start timestamp.
|
|
virtual uint32_t StartTimestamp() const;
|
|
|
|
// Configure start timestamp, default is a random number.
|
|
virtual int32_t SetStartTimestamp(const uint32_t timestamp);
|
|
|
|
virtual uint16_t SequenceNumber() const;
|
|
|
|
// Set SequenceNumber, default is a random number.
|
|
virtual int32_t SetSequenceNumber(const uint16_t seq);
|
|
|
|
virtual uint32_t SSRC() const;
|
|
|
|
// Configure SSRC, default is a random number.
|
|
virtual int32_t SetSSRC(const uint32_t ssrc);
|
|
|
|
virtual int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const;
|
|
|
|
virtual int32_t SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
|
|
const uint8_t arr_length);
|
|
|
|
virtual int32_t SetCSRCStatus(const bool include);
|
|
|
|
virtual uint32_t PacketCountSent() const;
|
|
|
|
virtual int CurrentSendFrequencyHz() const;
|
|
|
|
virtual uint32_t ByteCountSent() const;
|
|
|
|
virtual int32_t SetRTXSendStatus(const RtxMode mode,
|
|
const bool set_ssrc,
|
|
const uint32_t ssrc);
|
|
|
|
virtual int32_t RTXSendStatus(RtxMode* mode, uint32_t* ssrc,
|
|
int* payloadType) const;
|
|
|
|
|
|
virtual void SetRtxSendPayloadType(int payload_type);
|
|
|
|
// Sends kRtcpByeCode when going from true to false.
|
|
virtual int32_t SetSendingStatus(const bool sending);
|
|
|
|
virtual bool Sending() const;
|
|
|
|
// Drops or relays media packets.
|
|
virtual int32_t SetSendingMediaStatus(const bool sending);
|
|
|
|
virtual bool SendingMedia() const;
|
|
|
|
// Used by the codec module to deliver a video or audio frame for
|
|
// packetization.
|
|
virtual int32_t SendOutgoingData(
|
|
const FrameType frame_type,
|
|
const int8_t payload_type,
|
|
const uint32_t time_stamp,
|
|
int64_t capture_time_ms,
|
|
const uint8_t* payload_data,
|
|
const uint32_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation = NULL,
|
|
const RTPVideoHeader* rtp_video_hdr = NULL);
|
|
|
|
virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
|
|
int64_t capture_time_ms);
|
|
// Returns the number of padding bytes actually sent, which can be more or
|
|
// less than |bytes|.
|
|
virtual int TimeToSendPadding(int bytes);
|
|
// RTCP part.
|
|
|
|
// Get RTCP status.
|
|
virtual RTCPMethod RTCP() const;
|
|
|
|
// Configure RTCP status i.e on/off.
|
|
virtual int32_t SetRTCPStatus(const RTCPMethod method);
|
|
|
|
// Set RTCP CName.
|
|
virtual int32_t SetCNAME(const char c_name[RTCP_CNAME_SIZE]);
|
|
|
|
// Get RTCP CName.
|
|
virtual int32_t CNAME(char c_name[RTCP_CNAME_SIZE]);
|
|
|
|
// Get remote CName.
|
|
virtual int32_t RemoteCNAME(const uint32_t remote_ssrc,
|
|
char c_name[RTCP_CNAME_SIZE]) const;
|
|
|
|
// Get remote NTP.
|
|
virtual int32_t RemoteNTP(uint32_t* received_ntp_secs,
|
|
uint32_t* received_ntp_frac,
|
|
uint32_t* rtcp_arrival_time_secs,
|
|
uint32_t* rtcp_arrival_time_frac,
|
|
uint32_t* rtcp_timestamp) const;
|
|
|
|
virtual int32_t AddMixedCNAME(const uint32_t ssrc,
|
|
const char c_name[RTCP_CNAME_SIZE]);
|
|
|
|
virtual int32_t RemoveMixedCNAME(const uint32_t ssrc);
|
|
|
|
// Get RoundTripTime.
|
|
virtual int32_t RTT(const uint32_t remote_ssrc,
|
|
uint16_t* rtt,
|
|
uint16_t* avg_rtt,
|
|
uint16_t* min_rtt,
|
|
uint16_t* max_rtt) const;
|
|
|
|
// Reset RoundTripTime statistics.
|
|
virtual int32_t ResetRTT(const uint32_t remote_ssrc);
|
|
|
|
virtual void SetRtt(uint32_t rtt);
|
|
|
|
// Force a send of an RTCP packet.
|
|
// Normal SR and RR are triggered via the process function.
|
|
virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport);
|
|
|
|
virtual int32_t ResetSendDataCountersRTP();
|
|
|
|
// Statistics of the amount of data sent and received.
|
|
virtual int32_t DataCountersRTP(uint32_t* bytes_sent,
|
|
uint32_t* packets_sent) const;
|
|
|
|
// Get received RTCP report, sender info.
|
|
virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info);
|
|
|
|
// Get received RTCP report, report block.
|
|
virtual int32_t RemoteRTCPStat(
|
|
std::vector<RTCPReportBlock>* receive_blocks) const;
|
|
|
|
// Set received RTCP report block.
|
|
virtual int32_t AddRTCPReportBlock(
|
|
const uint32_t ssrc, const RTCPReportBlock* receive_block);
|
|
|
|
virtual int32_t RemoveRTCPReportBlock(const uint32_t ssrc);
|
|
|
|
// (REMB) Receiver Estimated Max Bitrate.
|
|
virtual bool REMB() const;
|
|
|
|
virtual int32_t SetREMBStatus(const bool enable);
|
|
|
|
virtual int32_t SetREMBData(const uint32_t bitrate,
|
|
const uint8_t number_of_ssrc,
|
|
const uint32_t* ssrc);
|
|
|
|
// (IJ) Extended jitter report.
|
|
virtual bool IJ() const;
|
|
|
|
virtual int32_t SetIJStatus(const bool enable);
|
|
|
|
// (TMMBR) Temporary Max Media Bit Rate.
|
|
virtual bool TMMBR() const;
|
|
|
|
virtual int32_t SetTMMBRStatus(const bool enable);
|
|
|
|
int32_t SetTMMBN(const TMMBRSet* bounding_set);
|
|
|
|
virtual uint16_t MaxPayloadLength() const;
|
|
|
|
virtual uint16_t MaxDataPayloadLength() const;
|
|
|
|
virtual int32_t SetMaxTransferUnit(const uint16_t size);
|
|
|
|
virtual int32_t SetTransportOverhead(
|
|
const bool tcp,
|
|
const bool ipv6,
|
|
const uint8_t authentication_overhead = 0);
|
|
|
|
// (NACK) Negative acknowledgment part.
|
|
|
|
virtual int SelectiveRetransmissions() const;
|
|
|
|
virtual int SetSelectiveRetransmissions(uint8_t settings);
|
|
|
|
// Send a Negative acknowledgment packet.
|
|
virtual int32_t SendNACK(const uint16_t* nack_list, const uint16_t size);
|
|
|
|
// Store the sent packets, needed to answer to a negative acknowledgment
|
|
// requests.
|
|
virtual int32_t SetStorePacketsStatus(
|
|
const bool enable, const uint16_t number_to_store);
|
|
|
|
// (APP) Application specific data.
|
|
virtual int32_t SetRTCPApplicationSpecificData(
|
|
const uint8_t sub_type,
|
|
const uint32_t name,
|
|
const uint8_t* data,
|
|
const uint16_t length);
|
|
|
|
// (XR) VOIP metric.
|
|
virtual int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
|
|
|
|
// Audio part.
|
|
|
|
// Set audio packet size, used to determine when it's time to send a DTMF
|
|
// packet in silence (CNG).
|
|
virtual int32_t SetAudioPacketSize(
|
|
const uint16_t packet_size_samples);
|
|
|
|
virtual bool SendTelephoneEventActive(int8_t& telephone_event) const;
|
|
|
|
// Send a TelephoneEvent tone using RFC 2833 (4733).
|
|
virtual int32_t SendTelephoneEventOutband(const uint8_t key,
|
|
const uint16_t time_ms,
|
|
const uint8_t level);
|
|
|
|
// Set payload type for Redundant Audio Data RFC 2198.
|
|
virtual int32_t SetSendREDPayloadType(const int8_t payload_type);
|
|
|
|
// Get payload type for Redundant Audio Data RFC 2198.
|
|
virtual int32_t SendREDPayloadType(int8_t& payload_type) const;
|
|
|
|
// Set status and id for header-extension-for-audio-level-indication.
|
|
virtual int32_t SetRTPAudioLevelIndicationStatus(
|
|
const bool enable, const uint8_t id);
|
|
|
|
// Get status and id for header-extension-for-audio-level-indication.
|
|
virtual int32_t GetRTPAudioLevelIndicationStatus(
|
|
bool& enable, uint8_t& id) const;
|
|
|
|
// Store the audio level in d_bov for header-extension-for-audio-level-
|
|
// indication.
|
|
virtual int32_t SetAudioLevel(const uint8_t level_d_bov);
|
|
|
|
// Video part.
|
|
|
|
virtual RtpVideoCodecTypes SendVideoCodec() const;
|
|
|
|
virtual int32_t SendRTCPSliceLossIndication(
|
|
const uint8_t picture_id);
|
|
|
|
// Set method for requestion a new key frame.
|
|
virtual int32_t SetKeyFrameRequestMethod(
|
|
const KeyFrameRequestMethod method);
|
|
|
|
// Send a request for a keyframe.
|
|
virtual int32_t RequestKeyFrame();
|
|
|
|
virtual int32_t SetCameraDelay(const int32_t delay_ms);
|
|
|
|
virtual void SetTargetSendBitrate(const uint32_t bitrate);
|
|
|
|
virtual int32_t SetGenericFECStatus(
|
|
const bool enable,
|
|
const uint8_t payload_type_red,
|
|
const uint8_t payload_type_fec);
|
|
|
|
virtual int32_t GenericFECStatus(
|
|
bool& enable,
|
|
uint8_t& payload_type_red,
|
|
uint8_t& payload_type_fec);
|
|
|
|
virtual int32_t SetFecParameters(
|
|
const FecProtectionParams* delta_params,
|
|
const FecProtectionParams* key_params);
|
|
|
|
virtual int32_t LastReceivedNTP(uint32_t& NTPsecs,
|
|
uint32_t& NTPfrac,
|
|
uint32_t& remote_sr);
|
|
|
|
virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
|
|
|
|
virtual void BitrateSent(uint32_t* total_rate,
|
|
uint32_t* video_rate,
|
|
uint32_t* fec_rate,
|
|
uint32_t* nackRate) const;
|
|
|
|
virtual uint32_t SendTimeOfSendReport(const uint32_t send_report);
|
|
|
|
// Good state of RTP receiver inform sender.
|
|
virtual int32_t SendRTCPReferencePictureSelection(
|
|
const uint64_t picture_id);
|
|
|
|
void OnReceivedTMMBR();
|
|
|
|
// Bad state of RTP receiver request a keyframe.
|
|
void OnRequestIntraFrame();
|
|
|
|
// Received a request for a new SLI.
|
|
void OnReceivedSliceLossIndication(const uint8_t picture_id);
|
|
|
|
// Received a new reference frame.
|
|
void OnReceivedReferencePictureSelectionIndication(
|
|
const uint64_t picture_id);
|
|
|
|
void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
|
|
|
|
void OnRequestSendReport();
|
|
|
|
protected:
|
|
void RegisterChildModule(RtpRtcp* module);
|
|
|
|
void DeRegisterChildModule(RtpRtcp* module);
|
|
|
|
bool UpdateRTCPReceiveInformationTimers();
|
|
|
|
uint32_t BitrateReceivedNow() const;
|
|
|
|
// Get remote SequenceNumber.
|
|
uint16_t RemoteSequenceNumber() const;
|
|
|
|
// Only for internal testing.
|
|
uint32_t LastSendReport(uint32_t& last_rtcptime);
|
|
|
|
RTPSender rtp_sender_;
|
|
|
|
RTCPSender rtcp_sender_;
|
|
RTCPReceiver rtcp_receiver_;
|
|
|
|
Clock* clock_;
|
|
|
|
private:
|
|
int64_t RtcpReportInterval();
|
|
|
|
int32_t id_;
|
|
const bool audio_;
|
|
bool collision_detected_;
|
|
int64_t last_process_time_;
|
|
int64_t last_bitrate_process_time_;
|
|
int64_t last_rtt_process_time_;
|
|
uint16_t packet_overhead_;
|
|
|
|
scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_;
|
|
scoped_ptr<CriticalSectionWrapper> critical_section_module_ptrs_feedback_;
|
|
ModuleRtpRtcpImpl* default_module_;
|
|
std::list<ModuleRtpRtcpImpl*> child_modules_;
|
|
|
|
// Send side
|
|
NACKMethod nack_method_;
|
|
uint32_t nack_last_time_sent_full_;
|
|
uint16_t nack_last_seq_number_sent_;
|
|
|
|
bool simulcast_;
|
|
VideoCodec send_video_codec_;
|
|
KeyFrameRequestMethod key_frame_req_method_;
|
|
|
|
RemoteBitrateEstimator* remote_bitrate_;
|
|
|
|
#ifdef MATLAB
|
|
MatlabPlot* plot1_;
|
|
#endif
|
|
|
|
RtcpRttObserver* rtt_observer_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
|