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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_OPUS_TEST_H_
#define MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
#include <math.h>
#include <cstddef>
#include <cstdint>
#include <memory>
#include "api/neteq/neteq.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/TestStereo.h"
namespace webrtc {
class OpusTest {
public:
OpusTest();
~OpusTest();
void Perform();
private:
void Run(TestPackStereo* channel,
size_t channels,
int bitrate,
size_t frame_length,
int percent_loss = 0);
void OpenOutFile(int test_number);
std::unique_ptr<NetEq> neteq_;
acm2::ResamplerHelper resampler_helper_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;
PCMFile out_file_;
PCMFile out_file_standalone_;
int counter_;
uint8_t payload_type_;
uint32_t rtp_timestamp_;
PushResampler<int16_t> resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
WebRtcOpusDecInst* opus_mono_decoder_;
WebRtcOpusDecInst* opus_stereo_decoder_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
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