blob: e466ea6c8bb4a58a09b431af44c7ee31c82b796e [file] [log] [blame]
/*
* 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) {}
~MockPacketBuffer() override { Die(); }
MOCK_METHOD(void, Die, ());
MOCK_METHOD(void, Flush, (), (override));
MOCK_METHOD(bool, Empty, (), (const, override));
MOCK_METHOD(int,
InsertPacket,
(Packet && packet, StatisticsCalculator* stats),
(override));
MOCK_METHOD(int,
InsertPacketList,
(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),
(override));
MOCK_METHOD(int,
NextTimestamp,
(uint32_t * next_timestamp),
(const, override));
MOCK_METHOD(int,
NextHigherTimestamp,
(uint32_t timestamp, uint32_t* next_timestamp),
(const, override));
MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override));
MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override));
MOCK_METHOD(int,
DiscardNextPacket,
(StatisticsCalculator * stats),
(override));
MOCK_METHOD(void,
DiscardOldPackets,
(uint32_t timestamp_limit,
uint32_t horizon_samples,
StatisticsCalculator* stats),
(override));
MOCK_METHOD(void,
DiscardAllOldPackets,
(uint32_t timestamp_limit, StatisticsCalculator* stats),
(override));
MOCK_METHOD(size_t, NumPacketsInBuffer, (), (const, override));
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_