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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h | 82 |
1 files changed, 82 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h new file mode 100644 index 0000000000..48357ea466 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h @@ -0,0 +1,82 @@ +/* + * 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_ |