9c55f0f957
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
59 lines
1.9 KiB
C++
59 lines
1.9 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
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#include "gmock/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)
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: PacketBuffer(max_number_of_packets) {}
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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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MOCK_METHOD1(InsertPacket,
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int(Packet* packet));
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MOCK_METHOD4(InsertPacketList,
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int(PacketList* packet_list,
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const DecoderDatabase& decoder_database,
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uint8_t* current_rtp_payload_type,
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uint8_t* current_cng_rtp_payload_type));
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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(NextRtpHeader,
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const RTPHeader*());
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MOCK_METHOD1(GetNextPacket,
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Packet*(int* discard_count));
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MOCK_METHOD0(DiscardNextPacket,
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int());
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MOCK_METHOD1(DiscardOldPackets,
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int(uint32_t timestamp_limit));
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MOCK_CONST_METHOD0(NumPacketsInBuffer,
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int());
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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 // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
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