| /* |
| * 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, 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()); |
| MOCK_METHOD1(InsertPacket, |
| int(Packet* packet)); |
| MOCK_METHOD4(InsertPacketList, |
| int(PacketList* packet_list, |
| const DecoderDatabase& decoder_database, |
| rtc::Optional<uint8_t>* current_rtp_payload_type, |
| rtc::Optional<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*(size_t* 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, |
| size_t()); |
| 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_ |