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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 TEST_MOCK_AUDIO_ENCODER_H_
#define TEST_MOCK_AUDIO_ENCODER_H_
#include <string>
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "test/gmock.h"
namespace webrtc {
class MockAudioEncoder : public AudioEncoder {
public:
MockAudioEncoder();
~MockAudioEncoder();
MOCK_METHOD(int, SampleRateHz, (), (const, override));
MOCK_METHOD(size_t, NumChannels, (), (const, override));
MOCK_METHOD(int, RtpTimestampRateHz, (), (const, override));
MOCK_METHOD(size_t, Num10MsFramesInNextPacket, (), (const, override));
MOCK_METHOD(size_t, Max10MsFramesInAPacket, (), (const, override));
MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
MOCK_METHOD((absl::optional<std::pair<TimeDelta, TimeDelta>>),
GetFrameLengthRange,
(),
(const, override));
MOCK_METHOD(void, Reset, (), (override));
MOCK_METHOD(bool, SetFec, (bool enable), (override));
MOCK_METHOD(bool, SetDtx, (bool enable), (override));
MOCK_METHOD(bool, SetApplication, (Application application), (override));
MOCK_METHOD(void, SetMaxPlaybackRate, (int frequency_hz), (override));
MOCK_METHOD(void,
OnReceivedUplinkBandwidth,
(int target_audio_bitrate_bps,
absl::optional<int64_t> probing_interval_ms),
(override));
MOCK_METHOD(void,
OnReceivedUplinkPacketLossFraction,
(float uplink_packet_loss_fraction),
(override));
MOCK_METHOD(void,
OnReceivedOverhead,
(size_t overhead_bytes_per_packet),
(override));
MOCK_METHOD(bool,
EnableAudioNetworkAdaptor,
(const std::string& config_string, RtcEventLog*),
(override));
// Note, we explicitly chose not to create a mock for the Encode method.
MOCK_METHOD(EncodedInfo,
EncodeImpl,
(uint32_t timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer*),
(override));
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 // TEST_MOCK_AUDIO_ENCODER_H_
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