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/*
* 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_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {
class MockAudioEncoder final : public AudioEncoder {
public:
~MockAudioEncoder() override { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD1(Mark, void(std::string desc));
MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
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());
// Note, we explicitly chose not to create a mock for the Encode method.
MOCK_METHOD4(EncodeInternal,
EncodedInfo(uint32_t timestamp,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded));
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(SetProjectedPacketLossRate, void(double fraction));
MOCK_METHOD1(SetTargetBitrate, void(int target_bps));
MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
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