ee92b664b3
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret. While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL. I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD(). BUG=issue2173 R=minyue@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1961004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
111 lines
2.7 KiB
C++
111 lines
2.7 KiB
C++
/*
|
|
* 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
|
|
|
|
#include <stdio.h>
|
|
|
|
#include "ACMTest.h"
|
|
#include "audio_coding_module.h"
|
|
#include "RTPFile.h"
|
|
#include "PCMFile.h"
|
|
#include "typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
#define MAX_INCOMING_PAYLOAD 8096
|
|
|
|
// TestPacketization callback which writes the encoded payloads to file
|
|
class TestPacketization : public AudioPacketizationCallback {
|
|
public:
|
|
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
|
|
~TestPacketization();
|
|
virtual int32_t SendData(const FrameType frameType, const uint8_t payloadType,
|
|
const uint32_t timeStamp, const uint8_t* payloadData,
|
|
const uint16_t payloadSize,
|
|
const RTPFragmentationHeader* fragmentation);
|
|
|
|
private:
|
|
static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
|
|
int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
|
|
RTPStream* _rtpStream;
|
|
int32_t _frequency;
|
|
int16_t _seqNo;
|
|
};
|
|
|
|
class Sender {
|
|
public:
|
|
Sender();
|
|
void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
|
|
void Teardown();
|
|
void Run();
|
|
bool Add10MsData();
|
|
|
|
//for auto_test and logging
|
|
uint8_t testMode;
|
|
uint8_t codeId;
|
|
|
|
private:
|
|
AudioCodingModule* _acm;
|
|
PCMFile _pcmFile;
|
|
AudioFrame _audioFrame;
|
|
TestPacketization* _packetization;
|
|
};
|
|
|
|
class Receiver {
|
|
public:
|
|
Receiver();
|
|
void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
|
|
void Teardown();
|
|
void Run();
|
|
bool IncomingPacket();
|
|
bool PlayoutData();
|
|
|
|
//for auto_test and logging
|
|
uint8_t codeId;
|
|
uint8_t testMode;
|
|
|
|
private:
|
|
AudioCodingModule* _acm;
|
|
RTPStream* _rtpStream;
|
|
PCMFile _pcmFile;
|
|
int16_t* _playoutBuffer;
|
|
uint16_t _playoutLengthSmpls;
|
|
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
|
|
uint16_t _payloadSizeBytes;
|
|
uint16_t _realPayloadSizeBytes;
|
|
int32_t _frequency;
|
|
bool _firstTime;
|
|
WebRtcRTPHeader _rtpInfo;
|
|
uint32_t _nextTime;
|
|
};
|
|
|
|
class EncodeDecodeTest : public ACMTest {
|
|
public:
|
|
EncodeDecodeTest();
|
|
EncodeDecodeTest(int testMode);
|
|
virtual void Perform();
|
|
|
|
uint16_t _playoutFreq;
|
|
uint8_t _testMode;
|
|
|
|
private:
|
|
void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
|
|
|
|
protected:
|
|
Sender _sender;
|
|
Receiver _receiver;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif
|