mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
d00ce747c7
Bug: webrtc:12338 Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34621}
93 lines
3 KiB
C++
93 lines
3 KiB
C++
/*
|
|
* 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 MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
|
|
#define MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
|
|
|
|
#include <memory>
|
|
#include <string>
|
|
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Define coding parameter as
|
|
// <channels, bit_rate, file_name, extension, if_save_output>.
|
|
typedef std::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_;
|
|
|
|
std::unique_ptr<int16_t[]> in_data_;
|
|
std::unique_ptr<int16_t[]> out_data_;
|
|
size_t data_pointer_;
|
|
size_t loop_length_samples_;
|
|
std::unique_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 // MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
|