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Diffstat (limited to 'third_party/libwebrtc/audio/mock_voe_channel_proxy.h')
| -rw-r--r-- | third_party/libwebrtc/audio/mock_voe_channel_proxy.h | 186 |
1 files changed, 186 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/mock_voe_channel_proxy.h b/third_party/libwebrtc/audio/mock_voe_channel_proxy.h new file mode 100644 index 0000000000..a02bee38ad --- /dev/null +++ b/third_party/libwebrtc/audio/mock_voe_channel_proxy.h @@ -0,0 +1,186 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ +#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ + +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "api/crypto/frame_decryptor_interface.h" +#include "api/test/mock_frame_encryptor.h" +#include "audio/channel_receive.h" +#include "audio/channel_send.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "test/gmock.h" + +namespace webrtc { +namespace test { + +class MockChannelReceive : public voe::ChannelReceiveInterface { + public: + MOCK_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override)); + MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override)); + MOCK_METHOD(void, + RegisterReceiverCongestionControlObjects, + (PacketRouter*), + (override)); + MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override)); + MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override)); + MOCK_METHOD(NetworkStatistics, + GetNetworkStatistics, + (bool), + (const, override)); + MOCK_METHOD(AudioDecodingCallStats, + GetDecodingCallStatistics, + (), + (const, override)); + MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override)); + MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override)); + MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override)); + MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override)); + MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override)); + MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override)); + MOCK_METHOD(void, + ReceivedRTCPPacket, + (const uint8_t*, size_t length), + (override)); + MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override)); + MOCK_METHOD(AudioMixer::Source::AudioFrameInfo, + GetAudioFrameWithInfo, + (int sample_rate_hz, AudioFrame*), + (override)); + MOCK_METHOD(int, PreferredSampleRate, (), (const, override)); + MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override)); + MOCK_METHOD(void, + SetAssociatedSendChannel, + (const voe::ChannelSendInterface*), + (override)); + MOCK_METHOD(bool, + GetPlayoutRtpTimestamp, + (uint32_t*, int64_t*), + (const, override)); + MOCK_METHOD(void, + SetEstimatedPlayoutNtpTimestampMs, + (int64_t ntp_timestamp_ms, int64_t time_ms), + (override)); + MOCK_METHOD(absl::optional<int64_t>, + GetCurrentEstimatedPlayoutNtpTimestampMs, + (int64_t now_ms), + (const, override)); + MOCK_METHOD(absl::optional<Syncable::Info>, + GetSyncInfo, + (), + (const, override)); + MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override)); + MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override)); + MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override)); + MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>), + GetReceiveCodec, + (), + (const, override)); + MOCK_METHOD(void, + SetReceiveCodecs, + ((const std::map<int, SdpAudioFormat>& codecs)), + (override)); + MOCK_METHOD(void, StartPlayout, (), (override)); + MOCK_METHOD(void, StopPlayout, (), (override)); + MOCK_METHOD( + void, + SetDepacketizerToDecoderFrameTransformer, + (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer), + (override)); + MOCK_METHOD( + void, + SetFrameDecryptor, + (rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor), + (override)); + MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override)); + MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override)); +}; + +class MockChannelSend : public voe::ChannelSendInterface { + public: + MOCK_METHOD(void, + SetEncoder, + (int payload_type, std::unique_ptr<AudioEncoder> encoder), + (override)); + MOCK_METHOD( + void, + ModifyEncoder, + (rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier), + (override)); + MOCK_METHOD(void, + CallEncoder, + (rtc::FunctionView<void(AudioEncoder*)> modifier), + (override)); + MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override)); + MOCK_METHOD(void, + SetSendAudioLevelIndicationStatus, + (bool enable, int id), + (override)); + MOCK_METHOD(void, + RegisterSenderCongestionControlObjects, + (RtpTransportControllerSendInterface*, RtcpBandwidthObserver*), + (override)); + MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override)); + MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override)); + MOCK_METHOD(std::vector<ReportBlock>, + GetRemoteRTCPReportBlocks, + (), + (const, override)); + MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override)); + MOCK_METHOD(void, + RegisterCngPayloadType, + (int payload_type, int payload_frequency), + (override)); + MOCK_METHOD(void, + SetSendTelephoneEventPayloadType, + (int payload_type, int payload_frequency), + (override)); + MOCK_METHOD(bool, + SendTelephoneEventOutband, + (int event, int duration_ms), + (override)); + MOCK_METHOD(void, + OnBitrateAllocation, + (BitrateAllocationUpdate update), + (override)); + MOCK_METHOD(void, SetInputMute, (bool muted), (override)); + MOCK_METHOD(void, + ReceivedRTCPPacket, + (const uint8_t*, size_t length), + (override)); + MOCK_METHOD(void, + ProcessAndEncodeAudio, + (std::unique_ptr<AudioFrame>), + (override)); + MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override)); + MOCK_METHOD(int, GetTargetBitrate, (), (const, override)); + MOCK_METHOD(int64_t, GetRTT, (), (const, override)); + MOCK_METHOD(void, StartSend, (), (override)); + MOCK_METHOD(void, StopSend, (), (override)); + MOCK_METHOD(void, + SetFrameEncryptor, + (rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor), + (override)); + MOCK_METHOD( + void, + SetEncoderToPacketizerFrameTransformer, + (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer), + (override)); +}; +} // namespace test +} // namespace webrtc + +#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ |