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platform-external-webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
Henrik Kjellander ff761fba82 modules: more interface -> include renames 
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

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/*
* 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/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.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/test/testsupport/gtest_prod_util.h"
namespace webrtc {
class ModuleRtpRtcpImpl : public RtpRtcp {
public:
explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
// Returns the number of milliseconds until the module want a worker thread to
// call Process.
int64_t TimeUntilNextProcess() override;
// Process any pending tasks such as timeouts.
int32_t Process() override;
// Receiver part.
// Called when we receive an RTCP packet.
int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
size_t incoming_packet_length) override;
void SetRemoteSSRC(uint32_t ssrc) override;
// Sender part.
int32_t RegisterSendPayload(const CodecInst& voice_codec) override;
int32_t RegisterSendPayload(const VideoCodec& video_codec) override;
int32_t DeRegisterSendPayload(int8_t payload_type) override;
int8_t SendPayloadType() const;
// Register RTP header extension.
int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
uint8_t id) override;
int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
// Get start timestamp.
uint32_t StartTimestamp() const override;
// Configure start timestamp, default is a random number.
void SetStartTimestamp(uint32_t timestamp) override;
uint16_t SequenceNumber() const override;
// Set SequenceNumber, default is a random number.
void SetSequenceNumber(uint16_t seq) override;
bool SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) override;
bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) override;
uint32_t SSRC() const override;
// Configure SSRC, default is a random number.
void SetSSRC(uint32_t ssrc) override;
void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
RTCPSender::FeedbackState GetFeedbackState();
int CurrentSendFrequencyHz() const;
void SetRtxSendStatus(int mode) override;
int RtxSendStatus() const override;
void SetRtxSsrc(uint32_t ssrc) override;
void SetRtxSendPayloadType(int payload_type,
int associated_payload_type) override;
std::pair<int, int> RtxSendPayloadType() const override;
// Sends kRtcpByeCode when going from true to false.
int32_t SetSendingStatus(bool sending) override;
bool Sending() const override;
// Drops or relays media packets.
void SetSendingMediaStatus(bool sending) override;
bool SendingMedia() const override;
// Used by the codec module to deliver a video or audio frame for
// packetization.
int32_t SendOutgoingData(FrameType frame_type,
int8_t payload_type,
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation = NULL,
const RTPVideoHeader* rtp_video_hdr = NULL) override;
bool TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission) override;
// Returns the number of padding bytes actually sent, which can be more or
// less than |bytes|.
size_t TimeToSendPadding(size_t bytes) override;
// RTCP part.
// Get RTCP status.
RtcpMode RTCP() const override;
// Configure RTCP status i.e on/off.
void SetRTCPStatus(RtcpMode method) override;
// Set RTCP CName.
int32_t SetCNAME(const char* c_name) override;
// Get remote CName.
int32_t RemoteCNAME(uint32_t remote_ssrc,
char c_name[RTCP_CNAME_SIZE]) const override;
// Get remote NTP.
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 override;
int32_t AddMixedCNAME(uint32_t ssrc, const char* c_name) override;
int32_t RemoveMixedCNAME(uint32_t ssrc) override;
// Get RoundTripTime.
int32_t RTT(uint32_t remote_ssrc,
int64_t* rtt,
int64_t* avg_rtt,
int64_t* min_rtt,
int64_t* max_rtt) const override;
// Force a send of an RTCP packet.
// Normal SR and RR are triggered via the process function.
int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
int32_t SendCompoundRTCP(
const std::set<RTCPPacketType>& rtcpPacketTypes) override;
// Statistics of the amount of data sent and received.
int32_t DataCountersRTP(size_t* bytes_sent,
uint32_t* packets_sent) const override;
void GetSendStreamDataCounters(
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const override;
void GetRtpPacketLossStats(
bool outgoing,
uint32_t ssrc,
struct RtpPacketLossStats* loss_stats) const override;
// Get received RTCP report, sender info.
int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) override;
// Get received RTCP report, report block.
int32_t RemoteRTCPStat(
std::vector<RTCPReportBlock>* receive_blocks) const override;
// (REMB) Receiver Estimated Max Bitrate.
bool REMB() const override;
void SetREMBStatus(bool enable) override;
void SetREMBData(uint32_t bitrate,
const std::vector<uint32_t>& ssrcs) override;
// (TMMBR) Temporary Max Media Bit Rate.
bool TMMBR() const override;
void SetTMMBRStatus(bool enable) override;
int32_t SetTMMBN(const TMMBRSet* bounding_set);
uint16_t MaxPayloadLength() const override;
uint16_t MaxDataPayloadLength() const override;
int32_t SetMaxTransferUnit(uint16_t size) override;
int32_t SetTransportOverhead(bool tcp,
bool ipv6,
uint8_t authentication_overhead = 0) override;
// (NACK) Negative acknowledgment part.
int SelectiveRetransmissions() const override;
int SetSelectiveRetransmissions(uint8_t settings) override;
// Send a Negative acknowledgment packet.
int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
// Store the sent packets, needed to answer to a negative acknowledgment
// requests.
void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
bool StorePackets() const override;
// Called on receipt of RTCP report block from remote side.
void RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) override;
RtcpStatisticsCallback* GetRtcpStatisticsCallback() override;
bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override;
// (APP) Application specific data.
int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
uint32_t name,
const uint8_t* data,
uint16_t length) override;
// (XR) VOIP metric.
int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) override;
// (XR) Receiver reference time report.
void SetRtcpXrRrtrStatus(bool enable) override;
bool RtcpXrRrtrStatus() const override;
// Audio part.
// Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG).
int32_t SetAudioPacketSize(uint16_t packet_size_samples) override;
// Send a TelephoneEvent tone using RFC 2833 (4733).
int32_t SendTelephoneEventOutband(uint8_t key,
uint16_t time_ms,
uint8_t level) override;
// Set payload type for Redundant Audio Data RFC 2198.
int32_t SetSendREDPayloadType(int8_t payload_type) override;
// Get payload type for Redundant Audio Data RFC 2198.
int32_t SendREDPayloadType(int8_t& payload_type) const override;
// Store the audio level in d_bov for header-extension-for-audio-level-
// indication.
int32_t SetAudioLevel(uint8_t level_d_bov) override;
// Video part.
int32_t SendRTCPSliceLossIndication(uint8_t picture_id) override;
// Set method for requesting a new key frame.
int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) override;
// Send a request for a keyframe.
int32_t RequestKeyFrame() override;
void SetTargetSendBitrate(uint32_t bitrate_bps) override;
void SetGenericFECStatus(bool enable,
uint8_t payload_type_red,
uint8_t payload_type_fec) override;
void GenericFECStatus(bool& enable,
uint8_t& payload_type_red,
uint8_t& payload_type_fec) override;
int32_t SetFecParameters(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params) override;
bool LastReceivedNTP(uint32_t* NTPsecs,
uint32_t* NTPfrac,
uint32_t* remote_sr) const;
bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
void BitrateSent(uint32_t* total_rate,
uint32_t* video_rate,
uint32_t* fec_rate,
uint32_t* nackRate) const override;
int64_t SendTimeOfSendReport(uint32_t send_report);
bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
// Good state of RTP receiver inform sender.
int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) override;
void RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback) override;
StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
const override;
void OnReceivedTMMBR();
// Bad state of RTP receiver request a keyframe.
void OnRequestIntraFrame();
// Received a request for a new SLI.
void OnReceivedSliceLossIndication(uint8_t picture_id);
// Received a new reference frame.
void OnReceivedReferencePictureSelectionIndication(uint64_t picture_id);
void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
void OnRequestSendReport();
protected:
bool UpdateRTCPReceiveInformationTimers();
uint32_t BitrateReceivedNow() const;
// Get remote SequenceNumber.
uint16_t RemoteSequenceNumber() const;
RTPSender rtp_sender_;
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
Clock* clock_;
private:
FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
int64_t RtcpReportInterval();
void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
void set_rtt_ms(int64_t rtt_ms);
int64_t rtt_ms() const;
bool TimeToSendFullNackList(int64_t now) const;
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_;
size_t padding_index_;
// Send side
NACKMethod nack_method_;
int64_t nack_last_time_sent_full_;
uint32_t nack_last_time_sent_full_prev_;
uint16_t nack_last_seq_number_sent_;
VideoCodec send_video_codec_;
KeyFrameRequestMethod key_frame_req_method_;
RemoteBitrateEstimator* remote_bitrate_;
RtcpRttStats* rtt_stats_;
PacketLossStats send_loss_stats_;
PacketLossStats receive_loss_stats_;
// The processed RTT from RtcpRttStats.
rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
int64_t rtt_ms_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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