| /* |
| * Copyright (c) 2014 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 WEBRTC_TEST_MOCK_AUDIO_ENCODER_H_ |
| #define WEBRTC_TEST_MOCK_AUDIO_ENCODER_H_ |
| |
| #include <string> |
| |
| #include "webrtc/api/audio_codecs/audio_encoder.h" |
| #include "webrtc/rtc_base/array_view.h" |
| #include "webrtc/test/gmock.h" |
| |
| namespace webrtc { |
| |
| class MockAudioEncoder : public AudioEncoder { |
| public: |
| // TODO(nisse): Valid overrides commented out, because the gmock |
| // methods don't use any override declarations, and we want to avoid |
| // warnings from -Winconsistent-missing-override. See |
| // http://crbug.com/428099. |
| MockAudioEncoder(); |
| ~MockAudioEncoder(); |
| MOCK_METHOD0(Die, void()); |
| MOCK_METHOD1(Mark, void(std::string desc)); |
| MOCK_CONST_METHOD0(SampleRateHz, int()); |
| MOCK_CONST_METHOD0(NumChannels, size_t()); |
| MOCK_CONST_METHOD0(RtpTimestampRateHz, int()); |
| MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t()); |
| MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t()); |
| MOCK_CONST_METHOD0(GetTargetBitrate, int()); |
| MOCK_METHOD0(Reset, void()); |
| MOCK_METHOD1(SetFec, bool(bool enable)); |
| MOCK_METHOD1(SetDtx, bool(bool enable)); |
| MOCK_METHOD1(SetApplication, bool(Application application)); |
| MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz)); |
| MOCK_METHOD1(SetMaxBitrate, void(int max_bps)); |
| MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes)); |
| MOCK_METHOD2(OnReceivedUplinkBandwidth, |
| void(int target_audio_bitrate_bps, |
| rtc::Optional<int64_t> probing_interval_ms)); |
| MOCK_METHOD1(OnReceivedUplinkPacketLossFraction, |
| void(float uplink_packet_loss_fraction)); |
| |
| MOCK_METHOD2(EnableAudioNetworkAdaptor, |
| bool(const std::string& config_string, RtcEventLog* event_log)); |
| |
| // Note, we explicitly chose not to create a mock for the Encode method. |
| MOCK_METHOD3(EncodeImpl, |
| EncodedInfo(uint32_t timestamp, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::Buffer* encoded)); |
| |
| class FakeEncoding { |
| public: |
| // Creates a functor that will return |info| and adjust the rtc::Buffer |
| // given as input to it, so it is info.encoded_bytes larger. |
| explicit FakeEncoding(const AudioEncoder::EncodedInfo& info); |
| |
| // Shorthand version of the constructor above, for when only setting |
| // encoded_bytes in the EncodedInfo object matters. |
| explicit FakeEncoding(size_t encoded_bytes); |
| |
| AudioEncoder::EncodedInfo operator()(uint32_t timestamp, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::Buffer* encoded); |
| |
| private: |
| AudioEncoder::EncodedInfo info_; |
| }; |
| |
| class CopyEncoding { |
| public: |
| ~CopyEncoding(); |
| |
| // Creates a functor that will return |info| and append the data in the |
| // payload to the buffer given as input to it. Up to info.encoded_bytes are |
| // appended - make sure the payload is big enough! Since it uses an |
| // ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out |
| // of scope! |
| CopyEncoding(AudioEncoder::EncodedInfo info, |
| rtc::ArrayView<const uint8_t> payload); |
| |
| // Shorthand version of the constructor above, for when you wish to append |
| // the whole payload and do not care about any EncodedInfo attribute other |
| // than encoded_bytes. |
| explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload); |
| |
| AudioEncoder::EncodedInfo operator()(uint32_t timestamp, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::Buffer* encoded); |
| |
| private: |
| AudioEncoder::EncodedInfo info_; |
| rtc::ArrayView<const uint8_t> payload_; |
| }; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_TEST_MOCK_AUDIO_ENCODER_H_ |