webrtc/modules/audio_coding/test/EncodeDecodeTest.h
Fredrik Solenberg ec0f45be11 Revert "Remove CodecInst pt.1"
This reverts commit 056f9738bf.

Reason for revert: breaks downstream

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

TBR=solenberg@webrtc.org,kwiberg@webrtc.org

Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}
2018-12-03 15:50:51 +00:00

114 lines
2.9 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 MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
#include <stdio.h>
#include <string.h>
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/RTPFile.h"
#include "modules/include/module_common_types.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();
int32_t SendData(const FrameType frameType,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
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,
std::string in_file_name, int sample_rate, size_t channels);
void Teardown();
void Run();
bool Add10MsData();
uint8_t codeId;
protected:
AudioCodingModule* _acm;
private:
PCMFile _pcmFile;
AudioFrame _audioFrame;
TestPacketization* _packetization;
};
class Receiver {
public:
Receiver();
virtual ~Receiver() {};
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, size_t channels);
void Teardown();
void Run();
virtual bool IncomingPacket();
bool PlayoutData();
//for auto_test and logging
uint8_t codeId;
private:
PCMFile _pcmFile;
int16_t* _playoutBuffer;
uint16_t _playoutLengthSmpls;
int32_t _frequency;
bool _firstTime;
protected:
AudioCodingModule* _acm;
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
WebRtcRTPHeader _rtpInfo;
size_t _realPayloadSizeBytes;
size_t _payloadSizeBytes;
uint32_t _nextTime;
};
class EncodeDecodeTest {
public:
explicit EncodeDecodeTest(int test_mode);
void Perform();
uint16_t _playoutFreq;
private:
std::string EncodeToFile(int fileType, int codeId, int* codePars);
protected:
Sender _sender;
Receiver _receiver;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_