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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}
186 lines
7 KiB
C++
186 lines
7 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/crypto/frame_decryptor_interface.h"
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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_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override));
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MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
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MOCK_METHOD(void,
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RegisterReceiverCongestionControlObjects,
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(PacketRouter*),
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(override));
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MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
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MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
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MOCK_METHOD(NetworkStatistics,
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GetNetworkStatistics,
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(bool),
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(const, override));
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MOCK_METHOD(AudioDecodingCallStats,
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GetDecodingCallStatistics,
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(),
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(const, override));
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MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override));
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MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override));
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MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override));
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MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override));
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MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override));
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MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override));
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MOCK_METHOD(void,
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ReceivedRTCPPacket,
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(const uint8_t*, size_t length),
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(override));
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MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override));
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MOCK_METHOD(AudioMixer::Source::AudioFrameInfo,
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GetAudioFrameWithInfo,
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(int sample_rate_hz, AudioFrame*),
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(override));
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MOCK_METHOD(int, PreferredSampleRate, (), (const, override));
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MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override));
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MOCK_METHOD(void,
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SetAssociatedSendChannel,
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(const voe::ChannelSendInterface*),
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(override));
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MOCK_METHOD(bool,
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GetPlayoutRtpTimestamp,
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(uint32_t*, int64_t*),
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(const, override));
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MOCK_METHOD(void,
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SetEstimatedPlayoutNtpTimestampMs,
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(int64_t ntp_timestamp_ms, int64_t time_ms),
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(override));
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MOCK_METHOD(absl::optional<int64_t>,
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GetCurrentEstimatedPlayoutNtpTimestampMs,
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(int64_t now_ms),
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(const, override));
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MOCK_METHOD(absl::optional<Syncable::Info>,
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GetSyncInfo,
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(),
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(const, override));
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MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override));
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MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override));
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MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override));
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MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>),
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GetReceiveCodec,
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(),
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(const, override));
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MOCK_METHOD(void,
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SetReceiveCodecs,
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((const std::map<int, SdpAudioFormat>& codecs)),
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(override));
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MOCK_METHOD(void, StartPlayout, (), (override));
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MOCK_METHOD(void, StopPlayout, (), (override));
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MOCK_METHOD(
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void,
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SetDepacketizerToDecoderFrameTransformer,
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(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
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(override));
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MOCK_METHOD(
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void,
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SetFrameDecryptor,
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(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
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(override));
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MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override));
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MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override));
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};
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class MockChannelSend : public voe::ChannelSendInterface {
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public:
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MOCK_METHOD(void,
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SetEncoder,
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(int payload_type, std::unique_ptr<AudioEncoder> encoder),
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(override));
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MOCK_METHOD(
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void,
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ModifyEncoder,
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(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier),
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(override));
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MOCK_METHOD(void,
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CallEncoder,
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(rtc::FunctionView<void(AudioEncoder*)> modifier),
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(override));
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MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override));
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MOCK_METHOD(void,
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SetSendAudioLevelIndicationStatus,
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(bool enable, int id),
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(override));
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MOCK_METHOD(void,
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RegisterSenderCongestionControlObjects,
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(RtpTransportControllerSendInterface*, RtcpBandwidthObserver*),
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(override));
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MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override));
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MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override));
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MOCK_METHOD(std::vector<ReportBlock>,
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GetRemoteRTCPReportBlocks,
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(),
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(const, override));
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MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override));
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MOCK_METHOD(void,
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RegisterCngPayloadType,
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(int payload_type, int payload_frequency),
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(override));
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MOCK_METHOD(void,
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SetSendTelephoneEventPayloadType,
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(int payload_type, int payload_frequency),
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(override));
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MOCK_METHOD(bool,
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SendTelephoneEventOutband,
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(int event, int duration_ms),
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(override));
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MOCK_METHOD(void,
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OnBitrateAllocation,
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(BitrateAllocationUpdate update),
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(override));
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MOCK_METHOD(void, SetInputMute, (bool muted), (override));
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MOCK_METHOD(void,
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ReceivedRTCPPacket,
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(const uint8_t*, size_t length),
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(override));
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MOCK_METHOD(void,
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ProcessAndEncodeAudio,
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(std::unique_ptr<AudioFrame>),
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(override));
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MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
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MOCK_METHOD(int, GetBitrate, (), (const, override));
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MOCK_METHOD(int64_t, GetRTT, (), (const, override));
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MOCK_METHOD(void, StartSend, (), (override));
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MOCK_METHOD(void, StopSend, (), (override));
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MOCK_METHOD(void,
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SetFrameEncryptor,
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(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor),
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(override));
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MOCK_METHOD(
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void,
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SetEncoderToPacketizerFrameTransformer,
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(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
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(override));
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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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