20a4b3fb2a
These are the changes made to WebRtcVoiceEngine and surrounding code. It still contains some things that are inelegant, like how AudioCodecSpec and AudioFormatInfo is ferried around in SendCodecSpec. This should probably be resolved before landing. There are also a few test still that are disabled. They should be removed or fixed, as the case may be. I've put this CL up to get a better overview of the changes made and how reviewable they are. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2705093002 Cr-Commit-Position: refs/heads/master@{#17904}
98 lines
4.8 KiB
C++
98 lines
4.8 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 WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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#define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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#include <string>
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#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "webrtc/test/gmock.h"
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#include "webrtc/voice_engine/channel_proxy.h"
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namespace webrtc {
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namespace test {
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class MockVoEChannelProxy : public voe::ChannelProxy {
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public:
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// GTest doesn't like move-only types, like std::unique_ptr
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bool 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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bool(int payload_type,
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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(SetRTCPStatus, void(bool enable));
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
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MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
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MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
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MOCK_METHOD1(EnableReceiveTransportSequenceNumber, 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_METHOD1(RegisterReceiverCongestionControlObjects,
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void(PacketRouter* packet_router));
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MOCK_METHOD0(ResetSenderCongestionControlObjects, void());
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MOCK_METHOD0(ResetReceiverCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
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MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
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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(GetSpeechOutputLevel, int());
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MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
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MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
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MOCK_METHOD2(SetSendTelephoneEventPayloadType, bool(int payload_type,
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int payload_frequency));
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MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
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MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
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// TODO(solenberg): Talk the compiler into accepting this mock method:
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// MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink));
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MOCK_METHOD1(SetInputMute, void(bool muted));
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MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
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MOCK_METHOD0(DeRegisterExternalTransport, void());
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MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
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MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
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MOCK_CONST_METHOD0(GetAudioDecoderFactory,
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const rtc::scoped_refptr<AudioDecoderFactory>&());
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MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
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MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
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MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
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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(NeededFrequency, int());
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MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
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MOCK_METHOD1(AssociateSendChannel,
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void(const ChannelProxy& send_channel_proxy));
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MOCK_METHOD0(DisassociateSendChannel, void());
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MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp,
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RtpReceiver** rtp_receiver));
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MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
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MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
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MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
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MOCK_METHOD1(SetReceiveCodecs,
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void(const std::map<int, SdpAudioFormat>& codecs));
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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(GetSources, std::vector<RtpSource>());
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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