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webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
Erik Språng 1e51a388bc Makes padding prefer video SSRCs instead of audio. 
Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.

This CL works around that by preferring to send padding on video SSRCs.

Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}
2019-12-11 16:32:14 +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 MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/include/module.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "test/gmock.h"
namespace webrtc {
class MockRtpRtcp : public RtpRtcp {
public:
MockRtpRtcp();
~MockRtpRtcp();
MOCK_METHOD2(IncomingRtcpPacket,
void(const uint8_t* incoming_packet, size_t packet_length));
MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size));
MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t());
MOCK_METHOD2(RegisterSendPayloadFrequency,
void(int payload_type, int frequency));
MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
MOCK_METHOD1(SetExtmapAllowMixed, void(bool extmap_allow_mixed));
MOCK_METHOD2(RegisterSendRtpHeaderExtension,
int32_t(RTPExtensionType type, uint8_t id));
MOCK_METHOD2(RegisterRtpHeaderExtension, void(absl::string_view uri, int id));
MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
int32_t(RTPExtensionType type));
MOCK_METHOD1(DeregisterSendRtpHeaderExtension, void(absl::string_view uri));
MOCK_CONST_METHOD0(SupportsPadding, bool());
MOCK_CONST_METHOD0(SupportsRtxPayloadPadding, bool());
MOCK_CONST_METHOD0(StartTimestamp, uint32_t());
MOCK_METHOD1(SetStartTimestamp, void(uint32_t timestamp));
MOCK_CONST_METHOD0(SequenceNumber, uint16_t());
MOCK_METHOD1(SetSequenceNumber, void(uint16_t seq));
MOCK_METHOD1(SetRtpState, void(const RtpState& rtp_state));
MOCK_METHOD1(SetRtxState, void(const RtpState& rtp_state));
MOCK_CONST_METHOD0(GetRtpState, RtpState());
MOCK_CONST_METHOD0(GetRtxState, RtpState());
MOCK_CONST_METHOD0(SSRC, uint32_t());
MOCK_METHOD1(SetSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetRid, void(const std::string& rid));
MOCK_METHOD1(SetMid, void(const std::string& mid));
MOCK_CONST_METHOD1(CSRCs, int32_t(uint32_t csrcs[kRtpCsrcSize]));
MOCK_METHOD1(SetCsrcs, void(const std::vector<uint32_t>& csrcs));
MOCK_METHOD1(SetCSRCStatus, int32_t(bool include));
MOCK_METHOD1(SetRtxSendStatus, void(int modes));
MOCK_CONST_METHOD0(RtxSendStatus, int());
MOCK_CONST_METHOD0(RtxSsrc, absl::optional<uint32_t>());
MOCK_METHOD1(SetRtxSsrc, void(uint32_t));
MOCK_METHOD2(SetRtxSendPayloadType, void(int, int));
MOCK_CONST_METHOD0(FlexfecSsrc, absl::optional<uint32_t>());
MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair<int, int>());
MOCK_METHOD1(SetSendingStatus, int32_t(bool sending));
MOCK_CONST_METHOD0(Sending, bool());
MOCK_METHOD1(SetSendingMediaStatus, void(bool sending));
MOCK_CONST_METHOD0(SendingMedia, bool());
MOCK_CONST_METHOD0(IsAudioConfigured, bool());
MOCK_METHOD1(SetAsPartOfAllocation, void(bool));
MOCK_CONST_METHOD4(BitrateSent,
void(uint32_t* total_rate,
uint32_t* video_rate,
uint32_t* fec_rate,
uint32_t* nack_rate));
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
int(uint32_t* available_bandwidth));
MOCK_METHOD4(OnSendingRtpFrame, bool(uint32_t, int64_t, int, bool));
MOCK_METHOD2(TrySendPacket,
bool(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info));
MOCK_METHOD1(OnPacketsAcknowledged, void(rtc::ArrayView<const uint16_t>));
MOCK_METHOD1(
GeneratePadding,
std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes));
MOCK_METHOD2(RegisterRtcpObservers,
void(RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback));
MOCK_CONST_METHOD0(RTCP, RtcpMode());
MOCK_METHOD1(SetRTCPStatus, void(RtcpMode method));
MOCK_METHOD1(SetCNAME, int32_t(const char cname[RTCP_CNAME_SIZE]));
MOCK_CONST_METHOD2(RemoteCNAME,
int32_t(uint32_t remote_ssrc,
char cname[RTCP_CNAME_SIZE]));
MOCK_CONST_METHOD5(RemoteNTP,
int32_t(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));
MOCK_METHOD2(AddMixedCNAME,
int32_t(uint32_t ssrc, const char cname[RTCP_CNAME_SIZE]));
MOCK_METHOD1(RemoveMixedCNAME, int32_t(uint32_t ssrc));
MOCK_CONST_METHOD5(RTT,
int32_t(uint32_t remote_ssrc,
int64_t* rtt,
int64_t* avg_rtt,
int64_t* min_rtt,
int64_t* max_rtt));
MOCK_CONST_METHOD0(ExpectedRetransmissionTimeMs, int64_t());
MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packet_type));
MOCK_METHOD1(SendCompoundRTCP,
int32_t(const std::set<RTCPPacketType>& packet_types));
MOCK_CONST_METHOD2(DataCountersRTP,
int32_t(size_t* bytes_sent, uint32_t* packets_sent));
MOCK_CONST_METHOD2(GetSendStreamDataCounters,
void(StreamDataCounters*, StreamDataCounters*));
MOCK_CONST_METHOD1(RemoteRTCPStat,
int32_t(std::vector<RTCPReportBlock>* receive_blocks));
MOCK_CONST_METHOD0(GetLatestReportBlockData, std::vector<ReportBlockData>());
MOCK_METHOD4(SetRTCPApplicationSpecificData,
int32_t(uint8_t sub_type,
uint32_t name,
const uint8_t* data,
uint16_t length));
MOCK_METHOD1(SetRtcpXrRrtrStatus, void(bool enable));
MOCK_CONST_METHOD0(RtcpXrRrtrStatus, bool());
MOCK_METHOD2(SetRemb, void(int64_t bitrate, std::vector<uint32_t> ssrcs));
MOCK_METHOD0(UnsetRemb, void());
MOCK_CONST_METHOD0(TMMBR, bool());
MOCK_METHOD1(SetTMMBRStatus, void(bool enable));
MOCK_METHOD1(OnBandwidthEstimateUpdate, void(uint16_t bandwidth_kbit));
MOCK_METHOD2(SendNACK, int32_t(const uint16_t* nack_list, uint16_t size));
MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
MOCK_METHOD2(SetStorePacketsStatus,
void(bool enable, uint16_t number_to_store));
MOCK_CONST_METHOD0(StorePackets, bool());
MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
MOCK_METHOD1(RegisterRtcpCnameCallback, void(RtcpCnameCallback*));
MOCK_METHOD1(SetReportBlockDataObserver, void(ReportBlockDataObserver*));
MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
MOCK_METHOD1(SendNetworkStateEstimatePacket,
bool(const rtcp::RemoteEstimate& packet));
MOCK_METHOD1(
SendCombinedRtcpPacket,
void(std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets));
MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps));
MOCK_METHOD4(SendLossNotification,
int32_t(uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed));
MOCK_METHOD0(Process, void());
MOCK_METHOD1(SetVideoBitrateAllocation, void(const VideoBitrateAllocation&));
MOCK_METHOD0(RtpSender, RTPSender*());
MOCK_CONST_METHOD0(RtpSender, const RTPSender*());
// Members.
unsigned int remote_ssrc_;
private:
// Mocking this method is currently not required and having a default
// implementation like MOCK_METHOD0(TimeUntilNextProcess, int64_t())
// can be dangerous since it can cause a tight loop on a process thread.
virtual int64_t TimeUntilNextProcess() { return 0xffffffff; }
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
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