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/*
* Copyright (c) 2012 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 MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "test/gmock.h"
namespace webrtc {
class MockPacketBuffer : public PacketBuffer {
public:
MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
: PacketBuffer(max_number_of_packets, tick_timer) {}
virtual ~MockPacketBuffer() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Flush, void());
MOCK_CONST_METHOD0(Empty, bool());
int InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
return InsertPacketWrapped(&packet, stats);
}
// Since gtest does not properly support move-only types, InsertPacket is
// implemented as a wrapper. You'll have to implement InsertPacketWrapped
// instead and move from |*packet|.
MOCK_METHOD2(InsertPacketWrapped,
int(Packet* packet, StatisticsCalculator* stats));
MOCK_METHOD5(InsertPacketList,
int(PacketList* packet_list,
const DecoderDatabase& decoder_database,
absl::optional<uint8_t>* current_rtp_payload_type,
absl::optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats));
MOCK_CONST_METHOD1(NextTimestamp, int(uint32_t* next_timestamp));
MOCK_CONST_METHOD2(NextHigherTimestamp,
int(uint32_t timestamp, uint32_t* next_timestamp));
MOCK_CONST_METHOD0(PeekNextPacket, const Packet*());
MOCK_METHOD0(GetNextPacket, absl::optional<Packet>());
MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
MOCK_METHOD3(DiscardOldPackets,
void(uint32_t timestamp_limit,
uint32_t horizon_samples,
StatisticsCalculator* stats));
MOCK_METHOD2(DiscardAllOldPackets,
void(uint32_t timestamp_limit, StatisticsCalculator* stats));
MOCK_CONST_METHOD0(NumPacketsInBuffer, size_t());
MOCK_METHOD1(IncrementWaitingTimes, void(int));
MOCK_CONST_METHOD0(current_memory_bytes, int());
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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