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/*
* 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 "absl/strings/string_view.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(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) 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,
absl::string_view in_file_name,
int in_sample_rate,
int payload_type,
SdpAudioFormat format);
void Teardown();
void Run();
bool Add10MsData();
protected:
AudioCodingModule* _acm;
private:
PCMFile _pcmFile;
AudioFrame _audioFrame;
TestPacketization* _packetization;
};
class Receiver {
public:
Receiver();
virtual ~Receiver() {}
void Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
absl::string_view out_file_name,
size_t channels,
int file_num);
void Teardown();
void Run();
virtual bool IncomingPacket();
bool PlayoutData();
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;
RTPHeader _rtpHeader;
size_t _realPayloadSizeBytes;
size_t _payloadSizeBytes;
uint32_t _nextTime;
};
class EncodeDecodeTest {
public:
EncodeDecodeTest();
void Perform();
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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