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/*
* Copyright (c) 2014 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_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
#include <fstream>
#include <gflags/gflags.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/typedefs.h"
using google::RegisterFlagValidator;
namespace webrtc {
namespace test {
class LossModel {
public:
virtual ~LossModel() {};
virtual bool Lost() = 0;
};
class NoLoss : public LossModel {
public:
bool Lost() override;
};
class UniformLoss : public LossModel {
public:
UniformLoss(double loss_rate);
bool Lost() override;
void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
private:
double loss_rate_;
};
class GilbertElliotLoss : public LossModel {
public:
GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
bool Lost() override;
private:
// Prob. of losing current packet, when previous packet is lost.
double prob_trans_11_;
// Prob. of losing current packet, when previous packet is not lost.
double prob_trans_01_;
bool lost_last_;
rtc::scoped_ptr<UniformLoss> uniform_loss_model_;
};
class NetEqQualityTest : public ::testing::Test {
protected:
NetEqQualityTest(int block_duration_ms,
int in_sampling_khz,
int out_sampling_khz,
NetEqDecoder decoder_type);
virtual ~NetEqQualityTest();
void SetUp() override;
// EncodeBlock(...) does the following:
// 1. encodes a block of audio, saved in |in_data| and has a length of
// |block_size_samples| (samples per channel),
// 2. save the bit stream to |payload| of |max_bytes| bytes in size,
// 3. returns the length of the payload (in bytes),
virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
uint8_t* payload, size_t max_bytes) = 0;
// PacketLost(...) determines weather a packet sent at an indicated time gets
// lost or not.
bool PacketLost();
// DecodeBlock() decodes a block of audio using the payload stored in
// |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
// audio is to be stored in |out_data_|.
int DecodeBlock();
// Transmit() uses |rtp_generator_| to generate a packet and passes it to
// |neteq_|.
int Transmit();
// Runs encoding / transmitting / decoding.
void Simulate();
// Write to log file. Usage Log() << ...
std::ofstream& Log();
NetEqDecoder decoder_type_;
const size_t channels_;
private:
int decoded_time_ms_;
int decodable_time_ms_;
double drift_factor_;
int packet_loss_rate_;
const int block_duration_ms_;
const int in_sampling_khz_;
const int out_sampling_khz_;
// Number of samples per channel in a frame.
const size_t in_size_samples_;
// Expected output number of samples per channel in a frame.
const size_t out_size_samples_;
size_t payload_size_bytes_;
size_t max_payload_bytes_;
rtc::scoped_ptr<InputAudioFile> in_file_;
rtc::scoped_ptr<AudioSink> output_;
std::ofstream log_file_;
rtc::scoped_ptr<RtpGenerator> rtp_generator_;
rtc::scoped_ptr<NetEq> neteq_;
rtc::scoped_ptr<LossModel> loss_model_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<uint8_t[]> payload_;
rtc::scoped_ptr<int16_t[]> out_data_;
WebRtcRTPHeader rtp_header_;
size_t total_payload_size_bytes_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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