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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 <gflags/gflags.h>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.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:
virtual bool Lost() OVERRIDE;
};
class UniformLoss : public LossModel {
public:
UniformLoss(double loss_rate);
virtual 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);
virtual 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_;
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,
enum NetEqDecoder decoder_type,
int channels,
std::string in_filename,
std::string out_filename);
virtual void SetUp() OVERRIDE;
virtual void TearDown() 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, int block_size_samples,
uint8_t* payload, int 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();
// Simulate(...) runs encoding / transmitting / decoding up to |end_time_ms|
// (miliseconds), the resulted audio is stored in the file with the name of
// |out_filename_|.
void Simulate(int end_time_ms);
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_;
const enum NetEqDecoder decoder_type_;
const int channels_;
const std::string in_filename_;
const std::string out_filename_;
const std::string log_filename_;
// Number of samples per channel in a frame.
const int in_size_samples_;
// Expected output number of samples per channel in a frame.
const int out_size_samples_;
size_t payload_size_bytes_;
int max_payload_bytes_;
scoped_ptr<InputAudioFile> in_file_;
FILE* out_file_;
FILE* log_file_;
scoped_ptr<RtpGenerator> rtp_generator_;
scoped_ptr<NetEq> neteq_;
scoped_ptr<LossModel> loss_model_;
scoped_ptr<int16_t[]> in_data_;
scoped_ptr<uint8_t[]> payload_;
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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