| /* | |
| * 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, (StatisticsCalculator * stats), (override)); | |
| MOCK_METHOD(void, | |
| PartialFlush, | |
| (int target_level_ms, | |
| size_t sample_rate, | |
| size_t last_decoded_length, | |
| StatisticsCalculator* stats), | |
| (override)); | |
| MOCK_METHOD(bool, Empty, (), (const, override)); | |
| MOCK_METHOD(int, | |
| InsertPacket, | |
| (Packet && packet, | |
| StatisticsCalculator* stats, | |
| size_t last_decoded_length, | |
| size_t sample_rate, | |
| int target_level_ms, | |
| const DecoderDatabase& decoder_database), | |
| (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, | |
| size_t last_decoded_length, | |
| size_t sample_rate, | |
| int target_level_ms), | |
| (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_ |