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This reverts commit2c41cbae37
. Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2. Original change's description: > Revert "Wire up non-sender RTT for audio, and implement related standardized stats." > > This reverts commitfb0dca6c05
. > > Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium. > > Original change's description: > > Wire up non-sender RTT for audio, and implement related standardized stats. > > > > The implemented stats are: > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements > > > > Bug: webrtc:12951, webrtc:12714 > > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956 > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34861} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta,hbos,minyue > > Bug: webrtc:12951, webrtc:12714 > Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Olga Sharonova <olka@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#34897} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12951, webrtc:12714 Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34930}
197 lines
7.9 KiB
C++
197 lines
7.9 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 MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
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#define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
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#include <memory>
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#include <set>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "modules/include/module.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
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#include "test/gmock.h"
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namespace webrtc {
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class MockRtpRtcpInterface : public RtpRtcpInterface {
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public:
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MOCK_METHOD(void,
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IncomingRtcpPacket,
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(const uint8_t* incoming_packet, size_t packet_length),
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(override));
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MOCK_METHOD(void, SetRemoteSSRC, (uint32_t ssrc), (override));
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MOCK_METHOD(void, SetLocalSsrc, (uint32_t ssrc), (override));
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MOCK_METHOD(void, SetMaxRtpPacketSize, (size_t size), (override));
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MOCK_METHOD(size_t, MaxRtpPacketSize, (), (const, override));
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MOCK_METHOD(void,
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RegisterSendPayloadFrequency,
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(int payload_type, int frequency),
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(override));
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MOCK_METHOD(int32_t,
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DeRegisterSendPayload,
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(int8_t payload_type),
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(override));
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MOCK_METHOD(void, SetExtmapAllowMixed, (bool extmap_allow_mixed), (override));
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MOCK_METHOD(void,
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RegisterRtpHeaderExtension,
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(absl::string_view uri, int id),
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(override));
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MOCK_METHOD(int32_t,
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DeregisterSendRtpHeaderExtension,
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(RTPExtensionType type),
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(override));
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MOCK_METHOD(void,
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DeregisterSendRtpHeaderExtension,
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(absl::string_view uri),
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(override));
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MOCK_METHOD(bool, SupportsPadding, (), (const, override));
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MOCK_METHOD(bool, SupportsRtxPayloadPadding, (), (const, override));
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MOCK_METHOD(uint32_t, StartTimestamp, (), (const, override));
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MOCK_METHOD(void, SetStartTimestamp, (uint32_t timestamp), (override));
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MOCK_METHOD(uint16_t, SequenceNumber, (), (const, override));
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MOCK_METHOD(void, SetSequenceNumber, (uint16_t seq), (override));
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MOCK_METHOD(void, SetRtpState, (const RtpState& rtp_state), (override));
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MOCK_METHOD(void, SetRtxState, (const RtpState& rtp_state), (override));
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MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
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MOCK_METHOD(RtpState, GetRtpState, (), (const, override));
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MOCK_METHOD(RtpState, GetRtxState, (), (const, override));
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MOCK_METHOD(uint32_t, SSRC, (), (const, override));
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MOCK_METHOD(void, SetRid, (const std::string& rid), (override));
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MOCK_METHOD(void, SetMid, (const std::string& mid), (override));
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MOCK_METHOD(void, SetCsrcs, (const std::vector<uint32_t>& csrcs), (override));
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MOCK_METHOD(void, SetRtxSendStatus, (int modes), (override));
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MOCK_METHOD(int, RtxSendStatus, (), (const, override));
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MOCK_METHOD(absl::optional<uint32_t>, RtxSsrc, (), (const, override));
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MOCK_METHOD(void, SetRtxSendPayloadType, (int, int), (override));
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MOCK_METHOD(absl::optional<uint32_t>, FlexfecSsrc, (), (const, override));
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MOCK_METHOD(int32_t, SetSendingStatus, (bool sending), (override));
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MOCK_METHOD(bool, Sending, (), (const, override));
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MOCK_METHOD(void, SetSendingMediaStatus, (bool sending), (override));
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MOCK_METHOD(bool, SendingMedia, (), (const, override));
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MOCK_METHOD(bool, IsAudioConfigured, (), (const, override));
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MOCK_METHOD(void, SetAsPartOfAllocation, (bool), (override));
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MOCK_METHOD(RtpSendRates, GetSendRates, (), (const, override));
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MOCK_METHOD(bool,
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OnSendingRtpFrame,
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(uint32_t, int64_t, int, bool),
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(override));
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MOCK_METHOD(bool,
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TrySendPacket,
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(RtpPacketToSend * packet, const PacedPacketInfo& pacing_info),
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(override));
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MOCK_METHOD(void,
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SetFecProtectionParams,
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(const FecProtectionParams& delta_params,
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const FecProtectionParams& key_params),
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(override));
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MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
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FetchFecPackets,
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(),
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(override));
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MOCK_METHOD(void,
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OnPacketsAcknowledged,
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(rtc::ArrayView<const uint16_t>),
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(override));
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MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
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GeneratePadding,
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(size_t target_size_bytes),
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(override));
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MOCK_METHOD(std::vector<RtpSequenceNumberMap::Info>,
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GetSentRtpPacketInfos,
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(rtc::ArrayView<const uint16_t> sequence_numbers),
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(const, override));
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MOCK_METHOD(size_t, ExpectedPerPacketOverhead, (), (const, override));
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MOCK_METHOD(void, OnPacketSendingThreadSwitched, (), (override));
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MOCK_METHOD(RtcpMode, RTCP, (), (const, override));
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MOCK_METHOD(void, SetRTCPStatus, (RtcpMode method), (override));
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MOCK_METHOD(int32_t,
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SetCNAME,
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(const char cname[RTCP_CNAME_SIZE]),
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(override));
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MOCK_METHOD(int32_t,
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RemoteNTP,
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(uint32_t * received_ntp_secs,
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uint32_t* received_ntp_frac,
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uint32_t* rtcp_arrival_time_secs,
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uint32_t* rtcp_arrival_time_frac,
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uint32_t* rtcp_timestamp),
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(const, override));
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MOCK_METHOD(int32_t,
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RTT,
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(uint32_t remote_ssrc,
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int64_t* rtt,
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int64_t* avg_rtt,
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int64_t* min_rtt,
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int64_t* max_rtt),
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(const, override));
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MOCK_METHOD(int64_t, ExpectedRetransmissionTimeMs, (), (const, override));
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MOCK_METHOD(int32_t, SendRTCP, (RTCPPacketType packet_type), (override));
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MOCK_METHOD(void,
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GetSendStreamDataCounters,
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(StreamDataCounters*, StreamDataCounters*),
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(const, override));
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MOCK_METHOD(std::vector<ReportBlockData>,
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GetLatestReportBlockData,
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(),
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(const, override));
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MOCK_METHOD(absl::optional<SenderReportStats>,
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GetSenderReportStats,
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(),
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(const, override));
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MOCK_METHOD(absl::optional<NonSenderRttStats>,
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GetNonSenderRttStats,
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(),
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(const, override));
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MOCK_METHOD(void,
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SetRemb,
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(int64_t bitrate, std::vector<uint32_t> ssrcs),
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(override));
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MOCK_METHOD(void, UnsetRemb, (), (override));
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MOCK_METHOD(int32_t,
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SendNACK,
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(const uint16_t* nack_list, uint16_t size),
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(override));
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MOCK_METHOD(void,
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SendNack,
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(const std::vector<uint16_t>& sequence_numbers),
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(override));
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MOCK_METHOD(void,
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SetStorePacketsStatus,
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(bool enable, uint16_t number_to_store),
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(override));
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MOCK_METHOD(void,
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SendCombinedRtcpPacket,
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(std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets),
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(override));
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MOCK_METHOD(int32_t,
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SendLossNotification,
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(uint16_t last_decoded_seq_num,
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uint16_t last_received_seq_num,
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bool decodability_flag,
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bool buffering_allowed),
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(override));
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MOCK_METHOD(void,
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SetVideoBitrateAllocation,
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(const VideoBitrateAllocation&),
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(override));
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MOCK_METHOD(RTPSender*, RtpSender, (), (override));
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MOCK_METHOD(const RTPSender*, RtpSender, (), (const, override));
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
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