From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../codecs/tools/audio_codec_speed_test.cc | 126 +++++++++++++++++++++ 1 file changed, 126 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc') diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc new file mode 100644 index 0000000000..537e6fcede --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc @@ -0,0 +1,126 @@ +/* + * 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 "modules/audio_coding/codecs/tools/audio_codec_speed_test.h" + +#include "rtc_base/checks.h" +#include "test/gtest.h" +#include "test/testsupport/file_utils.h" + +using ::std::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_( + static_cast(block_duration_ms_ * input_sampling_khz_)), + output_length_sample_( + static_cast(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"); + RTC_DCHECK(fp); + + // 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_ = 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"); + RTC_DCHECK(out_file_); + + 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 %zu-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 -- cgit v1.2.3