100 lines
2.9 KiB
C++
100 lines
2.9 KiB
C++
/*
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* Copyright (c) 2011 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 TEST_STEREO_H
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#define TEST_STEREO_H
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#include "ACMTest.h"
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#include "Channel.h"
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#include "PCMFile.h"
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class TestPackStereo : public AudioPacketizationCallback
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{
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public:
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TestPackStereo();
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~TestPackStereo();
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void RegisterReceiverACM(AudioCodingModule* acm);
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virtual WebRtc_Word32 SendData(const FrameType frameType,
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const WebRtc_UWord8 payloadType,
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const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const RTPFragmentationHeader* fragmentation);
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WebRtc_UWord16 GetPayloadSize();
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WebRtc_UWord32 GetTimeStampDiff();
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void ResetPayloadSize();
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void SetCodecType(int codecType);
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private:
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AudioCodingModule* _receiverACM;
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WebRtc_Word16 _seqNo;
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WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
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WebRtc_UWord32 _timeStampDiff;
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WebRtc_UWord32 _lastInTimestamp;
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WebRtc_UWord64 _totalBytes;
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WebRtc_UWord16 _payloadSize;
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WebRtc_UWord16 _noChannels;
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int _codecType;
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};
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class TestStereo : public ACMTest
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{
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public:
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TestStereo(int testMode);
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~TestStereo();
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void Perform();
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private:
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// The default value of '-1' indicates that the registration is based only on codec name
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// and a sampling frequncy matching is not required. This is useful for codecs which support
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// several sampling frequency.
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WebRtc_Word16 RegisterSendCodec(char side,
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char* codecName,
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WebRtc_Word32 sampFreqHz,
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int rate,
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int packSize);
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void Run(TestPackStereo* channel);
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void OpenOutFile(WebRtc_Word16 testNumber);
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void DisplaySendReceiveCodec();
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WebRtc_Word32 SendData(
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const FrameType frameType,
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const WebRtc_UWord8 payloadType,
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const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const RTPFragmentationHeader* fragmentation);
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int _testMode;
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AudioCodingModule* _acmA;
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AudioCodingModule* _acmB;
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TestPackStereo* _channelA2B;
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PCMFile _inFileA;
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PCMFile _outFileB;
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PCMFile _inFileStereo;
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WebRtc_Word16 _testCntr;
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WebRtc_UWord16 _packSizeSamp;
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WebRtc_UWord16 _packSizeBytes;
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int _counter;
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int _codecType;
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};
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#endif
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