1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
|
/*
* 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.
*/
#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
using ::std::tr1::get;
namespace webrtc {
AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
int input_sampling_khz,
int output_sampling_khz)
: block_duration_ms_(block_duration_ms),
input_sampling_khz_(input_sampling_khz),
output_sampling_khz_(output_sampling_khz),
input_length_sample_(block_duration_ms_ * input_sampling_khz_),
output_length_sample_(block_duration_ms_ * output_sampling_khz_),
data_pointer_(0),
loop_length_samples_(0),
max_bytes_(0),
encoded_bytes_(0),
encoding_time_ms_(0.0),
decoding_time_ms_(0.0),
out_file_(NULL) {
}
void AudioCodecSpeedTest::SetUp() {
channels_ = get<0>(GetParam());
bit_rate_ = get<1>(GetParam());
in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
save_out_data_ = get<4>(GetParam());
FILE* fp = fopen(in_filename_.c_str(), "rb");
assert(fp != NULL);
// Obtain file size.
fseek(fp, 0, SEEK_END);
loop_length_samples_ = ftell(fp) / sizeof(int16_t);
rewind(fp);
// Allocate memory to contain the whole file.
in_data_.reset(new int16_t[loop_length_samples_ +
input_length_sample_ * channels_]);
data_pointer_ = 0;
// Copy the file into the buffer.
ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
loop_length_samples_);
fclose(fp);
// Add an extra block length of samples to the end of the array, starting
// over again from the beginning of the array. This is done to simplify
// the reading process when reading over the end of the loop.
memcpy(&in_data_[loop_length_samples_], &in_data_[0],
input_length_sample_ * channels_ * sizeof(int16_t));
max_bytes_ =
static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
out_data_.reset(new int16_t[output_length_sample_ * channels_]);
bit_stream_.reset(new uint8_t[max_bytes_]);
if (save_out_data_) {
std::string out_filename =
::testing::UnitTest::GetInstance()->current_test_info()->name();
// Erase '/'
size_t found;
while ((found = out_filename.find('/')) != std::string::npos)
out_filename.replace(found, 1, "_");
out_filename = test::OutputPath() + out_filename + ".pcm";
out_file_ = fopen(out_filename.c_str(), "wb");
assert(out_file_ != NULL);
printf("Output to be saved in %s.\n", out_filename.c_str());
}
}
void AudioCodecSpeedTest::TearDown() {
if (save_out_data_) {
fclose(out_file_);
}
}
void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
size_t time_now_ms = 0;
float time_ms;
printf("Coding %d kHz-sampled %d-channel audio at %d bps ...\n",
input_sampling_khz_, channels_, bit_rate_);
while (time_now_ms < audio_duration_sec * 1000) {
// Encode & decode.
time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0],
max_bytes_, &encoded_bytes_);
encoding_time_ms_ += time_ms;
time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
decoding_time_ms_ += time_ms;
if (save_out_data_) {
fwrite(&out_data_[0], sizeof(int16_t),
output_length_sample_ * channels_, out_file_);
}
data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
loop_length_samples_;
time_now_ms += block_duration_ms_;
}
printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
(encoding_time_ms_ / audio_duration_sec) / 10.0,
(decoding_time_ms_ / audio_duration_sec) / 10.0);
}
} // namespace webrtc
|
