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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
126 lines
4.2 KiB
C++
126 lines
4.2 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
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#include "rtc_base/format_macros.h"
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#include "test/gtest.h"
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#include "test/testsupport/fileutils.h"
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using ::std::get;
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namespace webrtc {
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AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
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int input_sampling_khz,
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int output_sampling_khz)
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: block_duration_ms_(block_duration_ms),
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input_sampling_khz_(input_sampling_khz),
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output_sampling_khz_(output_sampling_khz),
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input_length_sample_(
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static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
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output_length_sample_(
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static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
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data_pointer_(0),
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loop_length_samples_(0),
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max_bytes_(0),
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encoded_bytes_(0),
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encoding_time_ms_(0.0),
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decoding_time_ms_(0.0),
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out_file_(NULL) {}
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void AudioCodecSpeedTest::SetUp() {
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channels_ = get<0>(GetParam());
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bit_rate_ = get<1>(GetParam());
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in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
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save_out_data_ = get<4>(GetParam());
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FILE* fp = fopen(in_filename_.c_str(), "rb");
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assert(fp != NULL);
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// Obtain file size.
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fseek(fp, 0, SEEK_END);
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loop_length_samples_ = ftell(fp) / sizeof(int16_t);
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rewind(fp);
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// Allocate memory to contain the whole file.
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in_data_.reset(
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new int16_t[loop_length_samples_ + input_length_sample_ * channels_]);
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data_pointer_ = 0;
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// Copy the file into the buffer.
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ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
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loop_length_samples_);
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fclose(fp);
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// Add an extra block length of samples to the end of the array, starting
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// over again from the beginning of the array. This is done to simplify
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// the reading process when reading over the end of the loop.
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memcpy(&in_data_[loop_length_samples_], &in_data_[0],
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input_length_sample_ * channels_ * sizeof(int16_t));
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max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
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out_data_.reset(new int16_t[output_length_sample_ * channels_]);
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bit_stream_.reset(new uint8_t[max_bytes_]);
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if (save_out_data_) {
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std::string out_filename =
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::testing::UnitTest::GetInstance()->current_test_info()->name();
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// Erase '/'
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size_t found;
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while ((found = out_filename.find('/')) != std::string::npos)
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out_filename.replace(found, 1, "_");
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out_filename = test::OutputPath() + out_filename + ".pcm";
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out_file_ = fopen(out_filename.c_str(), "wb");
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assert(out_file_ != NULL);
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printf("Output to be saved in %s.\n", out_filename.c_str());
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}
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}
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void AudioCodecSpeedTest::TearDown() {
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if (save_out_data_) {
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fclose(out_file_);
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}
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}
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void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
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size_t time_now_ms = 0;
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float time_ms;
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printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
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input_sampling_khz_, channels_, bit_rate_);
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while (time_now_ms < audio_duration_sec * 1000) {
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// Encode & decode.
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time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0],
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max_bytes_, &encoded_bytes_);
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encoding_time_ms_ += time_ms;
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time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
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decoding_time_ms_ += time_ms;
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if (save_out_data_) {
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fwrite(&out_data_[0], sizeof(int16_t), output_length_sample_ * channels_,
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out_file_);
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}
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data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
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loop_length_samples_;
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time_now_ms += block_duration_ms_;
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}
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printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
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(encoding_time_ms_ / audio_duration_sec) / 10.0,
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(decoding_time_ms_ / audio_duration_sec) / 10.0);
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}
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} // namespace webrtc
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