fcf79cca7b
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp Partial implementation: currently only populated when a/v sync is enabled. Bug: webrtc:7065 Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29581}
133 lines
6.0 KiB
C++
133 lines
6.0 KiB
C++
/*
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* Copyright (c) 2015 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 AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "api/test/mock_frame_encryptor.h"
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#include "audio/channel_receive.h"
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#include "audio/channel_send.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockChannelReceive : public voe::ChannelReceiveInterface {
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public:
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MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
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MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
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void(PacketRouter* packet_router));
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MOCK_METHOD0(ResetReceiverCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallReceiveStatistics());
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MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
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MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
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MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
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MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
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MOCK_CONST_METHOD0(GetTotalOutputDuration, double());
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MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
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MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink));
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MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
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MOCK_METHOD2(ReceivedRTCPPacket, void(const uint8_t* packet, size_t length));
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MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
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MOCK_METHOD2(GetAudioFrameWithInfo,
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AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
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AudioFrame* audio_frame));
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MOCK_CONST_METHOD0(PreferredSampleRate, int());
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MOCK_METHOD1(SetAssociatedSendChannel,
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void(const voe::ChannelSendInterface* send_channel));
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MOCK_CONST_METHOD2(GetPlayoutRtpTimestamp,
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bool(uint32_t* rtp_timestamp, int64_t* time_ms));
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MOCK_METHOD2(SetEstimatedPlayoutNtpTimestampMs,
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void(int64_t ntp_timestamp_ms, int64_t time_ms));
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MOCK_CONST_METHOD1(GetCurrentEstimatedPlayoutNtpTimestampMs,
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absl::optional<int64_t>(int64_t now_ms));
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MOCK_CONST_METHOD0(GetSyncInfo, absl::optional<Syncable::Info>());
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MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
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MOCK_METHOD1(SetBaseMinimumPlayoutDelayMs, bool(int delay_ms));
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MOCK_CONST_METHOD0(GetBaseMinimumPlayoutDelayMs, int());
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MOCK_CONST_METHOD0(GetReceiveCodec,
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absl::optional<std::pair<int, SdpAudioFormat>>());
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MOCK_METHOD1(SetReceiveCodecs,
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void(const std::map<int, SdpAudioFormat>& codecs));
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MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
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MOCK_METHOD0(StartPlayout, void());
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MOCK_METHOD0(StopPlayout, void());
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};
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class MockChannelSend : public voe::ChannelSendInterface {
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public:
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// GMock doesn't like move-only types, like std::unique_ptr.
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virtual void SetEncoder(int payload_type,
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std::unique_ptr<AudioEncoder> encoder) {
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return SetEncoderForMock(payload_type, &encoder);
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}
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MOCK_METHOD2(SetEncoderForMock,
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void(int payload_type, std::unique_ptr<AudioEncoder>* encoder));
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MOCK_METHOD1(
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ModifyEncoder,
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void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
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MOCK_METHOD1(CallEncoder,
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void(rtc::FunctionView<void(AudioEncoder*)> modifier));
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MOCK_METHOD3(SetRid,
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void(const std::string& rid,
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int extension_id,
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int repaired_extension_id));
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MOCK_METHOD2(SetMid, void(const std::string& mid, int extension_id));
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MOCK_METHOD1(SetRTCP_CNAME, void(absl::string_view c_name));
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MOCK_METHOD1(SetExtmapAllowMixed, void(bool extmap_allow_mixed));
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MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
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MOCK_METHOD2(RegisterSenderCongestionControlObjects,
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void(RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer));
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MOCK_METHOD0(ResetSenderCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallSendStatistics());
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MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
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MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
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MOCK_METHOD2(RegisterCngPayloadType,
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void(int payload_type, int payload_frequency));
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MOCK_METHOD2(SetSendTelephoneEventPayloadType,
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void(int payload_type, int payload_frequency));
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MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
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MOCK_METHOD1(OnBitrateAllocation, void(BitrateAllocationUpdate update));
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MOCK_METHOD1(SetInputMute, void(bool muted));
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MOCK_METHOD2(ReceivedRTCPPacket, void(const uint8_t* packet, size_t length));
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// GMock doesn't like move-only types, like std::unique_ptr.
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virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
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ProcessAndEncodeAudioForMock(&audio_frame);
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}
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MOCK_METHOD1(ProcessAndEncodeAudioForMock,
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void(std::unique_ptr<AudioFrame>* audio_frame));
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MOCK_METHOD1(SetTransportOverhead,
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void(size_t transport_overhead_per_packet));
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MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
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MOCK_CONST_METHOD0(GetBitrate, int());
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MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
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MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
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void(float recoverable_packet_loss_rate));
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MOCK_CONST_METHOD0(GetRTT, int64_t());
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MOCK_METHOD0(StartSend, void());
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MOCK_METHOD0(StopSend, void());
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MOCK_METHOD1(
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SetFrameEncryptor,
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void(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor));
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};
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} // namespace test
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} // namespace webrtc
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#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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