xref: /aosp_15_r20/external/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
12 #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
13 
14 #include "modules/audio_coding/neteq/packet_buffer.h"
15 #include "test/gmock.h"
16 
17 namespace webrtc {
18 
19 class MockPacketBuffer : public PacketBuffer {
20  public:
MockPacketBuffer(size_t max_number_of_packets,const TickTimer * tick_timer)21   MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
22       : PacketBuffer(max_number_of_packets, tick_timer) {}
~MockPacketBuffer()23   ~MockPacketBuffer() override { Die(); }
24   MOCK_METHOD(void, Die, ());
25   MOCK_METHOD(void, Flush, (StatisticsCalculator * stats), (override));
26   MOCK_METHOD(void,
27               PartialFlush,
28               (int target_level_ms,
29                size_t sample_rate,
30                size_t last_decoded_length,
31                StatisticsCalculator* stats),
32               (override));
33   MOCK_METHOD(bool, Empty, (), (const, override));
34   MOCK_METHOD(int,
35               InsertPacket,
36               (Packet && packet,
37                StatisticsCalculator* stats,
38                size_t last_decoded_length,
39                size_t sample_rate,
40                int target_level_ms,
41                const DecoderDatabase& decoder_database),
42               (override));
43   MOCK_METHOD(int,
44               InsertPacketList,
45               (PacketList * packet_list,
46                const DecoderDatabase& decoder_database,
47                absl::optional<uint8_t>* current_rtp_payload_type,
48                absl::optional<uint8_t>* current_cng_rtp_payload_type,
49                StatisticsCalculator* stats,
50                size_t last_decoded_length,
51                size_t sample_rate,
52                int target_level_ms),
53               (override));
54   MOCK_METHOD(int,
55               NextTimestamp,
56               (uint32_t * next_timestamp),
57               (const, override));
58   MOCK_METHOD(int,
59               NextHigherTimestamp,
60               (uint32_t timestamp, uint32_t* next_timestamp),
61               (const, override));
62   MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override));
63   MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override));
64   MOCK_METHOD(int,
65               DiscardNextPacket,
66               (StatisticsCalculator * stats),
67               (override));
68   MOCK_METHOD(void,
69               DiscardOldPackets,
70               (uint32_t timestamp_limit,
71                uint32_t horizon_samples,
72                StatisticsCalculator* stats),
73               (override));
74   MOCK_METHOD(void,
75               DiscardAllOldPackets,
76               (uint32_t timestamp_limit, StatisticsCalculator* stats),
77               (override));
78   MOCK_METHOD(size_t, NumPacketsInBuffer, (), (const, override));
79 };
80 
81 }  // namespace webrtc
82 #endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
83