8d609f6b6d
Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed:292084c376> > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed:fbcc5cb386TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
108 lines
5.4 KiB
C++
108 lines
5.4 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(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_METHOD1(EnableAudioNetworkAdaptor,
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void(const std::string& config_string));
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MOCK_METHOD0(DisableAudioNetworkAdaptor, void());
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MOCK_METHOD2(SetReceiverFrameLengthRange,
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void(int min_frame_length_ms, int max_frame_length_ms));
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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_CONST_METHOD1(GetSendCodec, bool(CodecInst* codec_inst));
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MOCK_METHOD1(SetVADStatus, bool(bool enable));
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MOCK_METHOD1(SetCodecFECStatus, bool(bool enable));
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MOCK_METHOD1(SetOpusDtx, bool(bool enable));
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MOCK_METHOD1(SetOpusMaxPlaybackRate, bool(int frequency_hz));
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MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
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MOCK_METHOD2(SetSendCNPayloadType,
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bool(int type, PayloadFrequencies frequency));
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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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