#include <math.h>
#include <stdio.h>
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/test/testsupport/fileutils.h"
|
| | webrtc::test::DEFINE_string (in_filename, ResourcePath("audio_coding/speech_mono_16kHz", "pcm"), "Filename for input audio (specify sample rate with --input_sample_rate ," "and channels with --channels).") |
| |
| | webrtc::test::DEFINE_int32 (input_sample_rate, 16000, "Sample rate of input file in Hz.") |
| |
| | webrtc::test::DEFINE_int32 (channels, 1, "Number of channels in input audio.") |
| |
| | webrtc::test::DEFINE_string (out_filename, OutputPath()+"neteq_quality_test_out.pcm", "Name of output audio file.") |
| |
| | webrtc::test::DEFINE_int32 (runtime_ms, 10000, "Simulated runtime (milliseconds).") |
| |
| | webrtc::test::DEFINE_int32 (packet_loss_rate, 10, "Percentile of packet loss.") |
| |
| | webrtc::test::DEFINE_int32 (random_loss_mode, 1, "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot loss.") |
| |
| | webrtc::test::DEFINE_int32 (burst_length, 30, "Burst length in milliseconds, only valid for Gilbert Elliot loss.") |
| |
| | webrtc::test::DEFINE_double (drift_factor, 0.0, "Time drift factor.") |
| |
