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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
68 lines
2.6 KiB
C++
68 lines
2.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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#include "modules/audio_coding/neteq/packet_buffer.h"
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#include "test/gmock.h"
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namespace webrtc {
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class MockPacketBuffer : public PacketBuffer {
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public:
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MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
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: PacketBuffer(max_number_of_packets, tick_timer) {}
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virtual ~MockPacketBuffer() { Die(); }
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MOCK_METHOD0(Die, void());
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MOCK_METHOD0(Flush,
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void());
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MOCK_CONST_METHOD0(Empty,
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bool());
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int InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
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return InsertPacketWrapped(&packet, stats);
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}
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// Since gtest does not properly support move-only types, InsertPacket is
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// implemented as a wrapper. You'll have to implement InsertPacketWrapped
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// instead and move from |*packet|.
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MOCK_METHOD2(InsertPacketWrapped,
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int(Packet* packet, StatisticsCalculator* stats));
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MOCK_METHOD5(InsertPacketList,
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int(PacketList* packet_list,
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const DecoderDatabase& decoder_database,
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rtc::Optional<uint8_t>* current_rtp_payload_type,
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rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
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StatisticsCalculator* stats));
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MOCK_CONST_METHOD1(NextTimestamp,
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int(uint32_t* next_timestamp));
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MOCK_CONST_METHOD2(NextHigherTimestamp,
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int(uint32_t timestamp, uint32_t* next_timestamp));
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MOCK_CONST_METHOD0(PeekNextPacket,
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const Packet*());
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MOCK_METHOD0(GetNextPacket,
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rtc::Optional<Packet>());
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MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
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MOCK_METHOD3(DiscardOldPackets,
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void(uint32_t timestamp_limit,
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uint32_t horizon_samples,
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StatisticsCalculator* stats));
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MOCK_METHOD2(DiscardAllOldPackets,
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void(uint32_t timestamp_limit, StatisticsCalculator* stats));
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MOCK_CONST_METHOD0(NumPacketsInBuffer,
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size_t());
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MOCK_METHOD1(IncrementWaitingTimes,
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void(int));
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MOCK_CONST_METHOD0(current_memory_bytes,
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int());
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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