blob: 94a0c564da49fa3a94fa47717a975473bf58f8bd [file] [log] [blame]
/*
* Copyright (c) 2011 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 TEST_STEREO_H
#define TEST_STEREO_H
#include "ACMTest.h"
#include "Channel.h"
#include "PCMFile.h"
namespace webrtc {
class TestPackStereo : public AudioPacketizationCallback
{
public:
TestPackStereo();
~TestPackStereo();
void RegisterReceiverACM(AudioCodingModule* acm);
virtual WebRtc_Word32 SendData(const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const RTPFragmentationHeader* fragmentation);
WebRtc_UWord16 GetPayloadSize();
WebRtc_UWord32 GetTimeStampDiff();
void ResetPayloadSize();
void SetCodecType(int codecType);
private:
AudioCodingModule* _receiverACM;
WebRtc_Word16 _seqNo;
WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
WebRtc_UWord32 _timeStampDiff;
WebRtc_UWord32 _lastInTimestamp;
WebRtc_UWord64 _totalBytes;
WebRtc_UWord16 _payloadSize;
WebRtc_UWord16 _noChannels;
int _codecType;
};
class TestStereo : public ACMTest
{
public:
TestStereo(int testMode);
~TestStereo();
void Perform();
private:
// The default value of '-1' indicates that the registration is based only on codec name
// and a sampling frequncy matching is not required. This is useful for codecs which support
// several sampling frequency.
WebRtc_Word16 RegisterSendCodec(char side,
char* codecName,
WebRtc_Word32 sampFreqHz,
int rate,
int packSize,
int channels);
void Run(TestPackStereo* channel, int in_channels, int out_channels);
void OpenOutFile(WebRtc_Word16 testNumber);
void DisplaySendReceiveCodec();
WebRtc_Word32 SendData(
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const RTPFragmentationHeader* fragmentation);
int _testMode;
AudioCodingModule* _acmA;
AudioCodingModule* _acmB;
TestPackStereo* _channelA2B;
PCMFile _in_file_stereo;
PCMFile _in_file_mono;
PCMFile _outFileB;
WebRtc_Word16 _testCntr;
WebRtc_UWord16 _packSizeSamp;
WebRtc_UWord16 _packSizeBytes;
int _counter;
int _codecType;
};
} // namespace webrtc
#endif