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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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace webrtc {
class MockPacketBuffer : public PacketBuffer {
public:
MockPacketBuffer(size_t max_number_of_packets)
: PacketBuffer(max_number_of_packets) {}
virtual ~MockPacketBuffer() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Flush,
void());
MOCK_CONST_METHOD0(Empty,
bool());
MOCK_METHOD1(InsertPacket,
int(Packet* packet));
MOCK_METHOD4(InsertPacketList,
int(PacketList* packet_list,
const DecoderDatabase& decoder_database,
uint8_t* current_rtp_payload_type,
uint8_t* current_cng_rtp_payload_type));
MOCK_CONST_METHOD1(NextTimestamp,
int(uint32_t* next_timestamp));
MOCK_CONST_METHOD2(NextHigherTimestamp,
int(uint32_t timestamp, uint32_t* next_timestamp));
MOCK_CONST_METHOD0(NextRtpHeader,
const RTPHeader*());
MOCK_METHOD1(GetNextPacket,
Packet*(int* discard_count));
MOCK_METHOD0(DiscardNextPacket,
int());
MOCK_METHOD2(DiscardOldPackets,
int(uint32_t timestamp_limit, uint32_t horizon_samples));
MOCK_METHOD1(DiscardAllOldPackets,
int(uint32_t timestamp_limit));
MOCK_CONST_METHOD0(NumPacketsInBuffer,
int());
MOCK_METHOD1(IncrementWaitingTimes,
void(int));
MOCK_CONST_METHOD0(current_memory_bytes,
int());
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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