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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_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Define coding parameter as
// <channels, bit_rate, file_name, extension, if_save_output>.
typedef std::tr1::tuple<size_t, int, std::string, std::string, bool>
coding_param;
class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
protected:
AudioCodecSpeedTest(int block_duration_ms,
int input_sampling_khz,
int output_sampling_khz);
virtual void SetUp();
virtual void TearDown();
// EncodeABlock(...) does the following:
// 1. encodes a block of audio, saved in |in_data|,
// 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size,
// 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
// 4. return the cost of time (in millisecond) spent on actual encoding.
virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes) = 0;
// DecodeABlock(...) does the following:
// 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
// (in bytes),
// 2. save the decoded audio in |out_data|,
// 3. return the cost of time (in millisecond) spent on actual decoding.
virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
int16_t* out_data) = 0;
// Encoding and decode an audio of |audio_duration| (in seconds) and
// record the runtime for encoding and decoding separately.
void EncodeDecode(size_t audio_duration);
int block_duration_ms_;
int input_sampling_khz_;
int output_sampling_khz_;
// Number of samples-per-channel in a frame.
size_t input_length_sample_;
// Expected output number of samples-per-channel in a frame.
size_t output_length_sample_;
rtc::scoped_ptr<int16_t[]> in_data_;
rtc::scoped_ptr<int16_t[]> out_data_;
size_t data_pointer_;
size_t loop_length_samples_;
rtc::scoped_ptr<uint8_t[]> bit_stream_;
// Maximum number of bytes in output bitstream for a frame of audio.
size_t max_bytes_;
size_t encoded_bytes_;
float encoding_time_ms_;
float decoding_time_ms_;
FILE* out_file_;
size_t channels_;
// Bit rate is in bit-per-second.
int bit_rate_;
std::string in_filename_;
// Determines whether to save the output to file.
bool save_out_data_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
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