Added a bitexactness test for the noise suppressor.
This CL also extracts part of the functionality used in the bitexactness test for the high-pass filter into a separate file in order to be able to reuse that functionality in bitexactness tests for the other submodules in APM (including the bitexactness test for the noise suppressor). BUG=wertc:5336 Review URL: https://codereview.webrtc.org/1783203002 Cr-Commit-Position: refs/heads/master@{#12061}
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@ -14,52 +14,11 @@
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
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#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
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#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
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namespace webrtc {
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namespace {
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// Test to see whether two vectors are identical and report any
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// differences.
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::testing::AssertionResult AssertVectorsNotEqual(
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const char* m_expr,
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const char* n_expr,
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const std::vector<float>& output,
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const std::vector<float>& reference) {
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// Compare the output in the reference in a soft manner.
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bool equal = true;
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const float threshold = 1.0f / 32768.0f;
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for (size_t k = 0; k < reference.size(); ++k) {
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if (fabs(output[k] - reference[k]) > threshold) {
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equal = false;
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break;
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}
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}
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// If the vectors are deemed not to be similar, return a report of the
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// difference.
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if (!equal) {
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// Lambda function that produces a formatted string with the data in the
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// vector.
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auto print_vector_in_c_format = [](std::vector<float> v,
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size_t num_values_to_print) {
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std::string s = "{ ";
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for (size_t k = 0; k < num_values_to_print; ++k) {
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s += std::to_string(v[k]) + "f";
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s += (k < (num_values_to_print - 1)) ? ", " : "";
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}
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return s + " }";
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};
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return ::testing::AssertionFailure()
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<< "Actual: " << print_vector_in_c_format(output, reference.size())
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<< std::endl
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<< std::endl
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<< "Expected: "
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<< print_vector_in_c_format(reference, reference.size())
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<< std::endl;
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}
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return ::testing::AssertionSuccess();
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}
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// Process one frame of data and produce the output.
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std::vector<float> ProcessOneFrame(const std::vector<float>& frame_input,
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@ -72,8 +31,9 @@ std::vector<float> ProcessOneFrame(const std::vector<float>& frame_input,
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test::CopyVectorToAudioBuffer(stream_config, frame_input, &audio_buffer);
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high_pass_filter->ProcessCaptureAudio(&audio_buffer);
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std::vector<float> frame_output =
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test::ExtractVectorFromAudioBuffer(stream_config, &audio_buffer);
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std::vector<float> frame_output;
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test::ExtractVectorFromAudioBuffer(stream_config, &audio_buffer,
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&frame_output);
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return frame_output;
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}
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@ -122,7 +82,9 @@ void RunBitexactnessTest(int sample_rate,
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reference_frame_length);
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}
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EXPECT_PRED_FORMAT2(AssertVectorsNotEqual, output_to_verify, reference);
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const float kTolerance = 1.0f / 32768.0f;
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EXPECT_TRUE(test::BitExactFrame(reference_frame_length, num_channels,
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reference, output_to_verify, kTolerance));
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}
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// Method for forming a vector out of an array.
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@ -0,0 +1,260 @@
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/*
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* Copyright (c) 2016 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 <vector>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/array_view.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
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#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
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#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
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namespace webrtc {
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namespace {
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const int kNumFramesToProcess = 1000;
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// Process one frame of data and produce the output.
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void ProcessOneFrame(int sample_rate_hz,
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AudioBuffer* capture_buffer,
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NoiseSuppressionImpl* noise_suppressor) {
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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capture_buffer->SplitIntoFrequencyBands();
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}
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noise_suppressor->AnalyzeCaptureAudio(capture_buffer);
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noise_suppressor->ProcessCaptureAudio(capture_buffer);
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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capture_buffer->MergeFrequencyBands();
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}
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}
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// Processes a specified amount of frames, verifies the results and reports
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// any errors.
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void RunBitexactnessTest(int sample_rate_hz,
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size_t num_channels,
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NoiseSuppressionImpl::Level level,
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float speech_probability_reference,
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rtc::ArrayView<const float> noise_estimate_reference,
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rtc::ArrayView<const float> output_reference) {
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rtc::CriticalSection crit_capture;
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NoiseSuppressionImpl noise_suppressor(&crit_capture);
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noise_suppressor.Initialize(num_channels, sample_rate_hz);
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noise_suppressor.Enable(true);
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noise_suppressor.set_level(level);
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int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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const StreamConfig capture_config(sample_rate_hz, num_channels, false);
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AudioBuffer capture_buffer(
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames());
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test::InputAudioFile capture_file(
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test::GetApmCaptureTestVectorFileName(sample_rate_hz));
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std::vector<float> capture_input(samples_per_channel * num_channels);
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for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&capture_file, capture_input);
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test::CopyVectorToAudioBuffer(capture_config, capture_input,
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&capture_buffer);
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ProcessOneFrame(sample_rate_hz, &capture_buffer, &noise_suppressor);
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}
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// Extract test results.
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std::vector<float> capture_output;
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test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
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&capture_output);
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float speech_probability = noise_suppressor.speech_probability();
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std::vector<float> noise_estimate = noise_suppressor.NoiseEstimate();
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const float kTolerance = 1.0f / 32768.0f;
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EXPECT_FLOAT_EQ(speech_probability_reference, speech_probability);
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EXPECT_TRUE(test::BitExactVector(noise_estimate_reference, noise_estimate,
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kTolerance));
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// Compare the output with the reference. Only the first values of the output
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// from last frame processed are compared in order not having to specify all
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// preceeding frames as testvectors. As the algorithm being tested has a
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// memory, testing only the last frame implicitly also tests the preceeding
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// frames.
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EXPECT_TRUE(test::BitExactFrame(
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capture_config.num_frames(), capture_config.num_channels(),
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output_reference, capture_output, kTolerance));
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}
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} // namespace
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TEST(NoiseSuppresionBitExactnessTest, Mono8kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.797542f, 6.488125f, 14.995160f};
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const float kOutputReference[] = {0.003510f, 0.004517f, 0.004669f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.797542f, 6.488125f, 14.995160f};
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const float kOutputReference[] = {0.003510f, 0.004517f, 0.004669f};
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#else
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const float kSpeechProbabilityReference = 0.73421317f;
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const float kNoiseEstimateReference[] = {0.035866f, 0.100382f, 0.229889f};
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const float kOutputReference[] = {0.003263f, 0.004402f, 0.004537f};
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#endif
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RunBitexactnessTest(8000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.475060f, 6.130507f, 14.030761f};
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const float kOutputReference[] = {0.003449f, 0.004334f, 0.004303f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.475060f, 6.130507f, 14.030761f};
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const float kOutputReference[] = {0.003449f, 0.004334f, 0.004303f};
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#else
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const float kSpeechProbabilityReference = 0.71672988f;
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const float kNoiseEstimateReference[] = {0.065653f, 0.198662f, 0.477870f};
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const float kOutputReference[] = {0.003574f, 0.004494f, 0.004499f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono32kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.480526f, 6.169749f, 14.102388f};
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const float kOutputReference[] = {0.001679f, 0.002411f, 0.002594f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.480526f, 6.169749f, 14.102388f};
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const float kOutputReference[] = {0.001679f, 0.002411f, 0.002594f};
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#else
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const float kSpeechProbabilityReference = 0.67999554f;
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const float kNoiseEstimateReference[] = {0.065606f, 0.215971f, 0.455931f};
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const float kOutputReference[] = {0.001221f, 0.001984f, 0.002228f};
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#endif
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RunBitexactnessTest(32000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono48kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.504498f, 6.068024f, 13.058871f};
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const float kOutputReference[] = {-0.013185f, -0.012769f, -0.012023f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.504498f, 6.068024f, 13.058871f};
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const float kOutputReference[] = {-0.013185f, -0.012769f, -0.012023f};
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#else
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const float kSpeechProbabilityReference = 0.70645678f;
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const float kNoiseEstimateReference[] = {0.066186f, 0.210660f, 0.402548f};
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const float kOutputReference[] = {-0.013062f, -0.012657f, -0.011934f};
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#endif
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RunBitexactnessTest(48000, 1, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Stereo16kHzLow) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {9.757937f, 12.392158f, 11.317673f};
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const float kOutputReference[] = {-0.011108f, -0.007904f, -0.012390f,
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-0.002441f, 0.000855f, -0.003204f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {10.079447f, 11.849465f, 10.667051f};
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const float kOutputReference[] = {-0.011108f, -0.007904f, -0.012390f,
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-0.002472f, 0.000916f, -0.003235f};
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#else
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const float kSpeechProbabilityReference = 0.67230678f;
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const float kNoiseEstimateReference[] = {0.298195f, 0.345745f, 0.320528f};
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const float kOutputReference[] = {-0.011459f, -0.008110f, -0.012728f,
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-0.002399f, 0.001018f, -0.003189f};
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#endif
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RunBitexactnessTest(16000, 2, NoiseSuppression::Level::kLow,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzModerate) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {1.004436f, 3.711453f, 9.602631f};
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const float kOutputReference[] = {0.004669f, 0.005524f, 0.005432f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {1.095946f, 3.351643f, 8.167248f};
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const float kOutputReference[] = {0.004669f, 0.005615f, 0.005585f};
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#else
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const float kSpeechProbabilityReference = 0.70897013f;
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const float kNoiseEstimateReference[] = {0.066269f, 0.199999f, 0.476885f};
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const float kOutputReference[] = {0.004513f, 0.005590f, 0.005614f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kModerate,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzHigh) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {1.023022f, 3.759059f, 9.614030f};
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const float kOutputReference[] = {0.004639f, 0.005402f, 0.005310f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {1.114510f, 3.410356f, 8.262188f};
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const float kOutputReference[] = {0.004547f, 0.005432f, 0.005402f};
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#else
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const float kSpeechProbabilityReference = 0.70106733f;
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const float kNoiseEstimateReference[] = {0.067901f, 0.204835f, 0.481723f};
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const float kOutputReference[] = {0.004394f, 0.005406f, 0.005416f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kHigh,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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TEST(NoiseSuppresionBitExactnessTest, Mono16kHzVeryHigh) {
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#if defined(WEBRTC_ARCH_ARM64)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.614974f, 6.041980f, 14.029047f};
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const float kOutputReference[] = {0.004273f, 0.005127f, 0.005188f};
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#elif defined(WEBRTC_ARCH_ARM)
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const float kSpeechProbabilityReference = -4.0f;
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const float kNoiseEstimateReference[] = {2.614974f, 6.041980f, 14.029047f};
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const float kOutputReference[] = {0.004273f, 0.005127f, 0.005188f};
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#else
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const float kSpeechProbabilityReference = 0.70281971f;
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const float kNoiseEstimateReference[] = {0.068797f, 0.205191f, 0.481312f};
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const float kOutputReference[] = {0.004321f, 0.005247f, 0.005263f};
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#endif
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RunBitexactnessTest(16000, 1, NoiseSuppression::Level::kVeryHigh,
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kSpeechProbabilityReference, kNoiseEstimateReference,
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kOutputReference);
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}
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} // namespace webrtc
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@ -10,10 +10,12 @@
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#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
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#include <string.h>
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namespace webrtc {
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namespace test {
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void SetupFrame(StreamConfig stream_config,
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void SetupFrame(const StreamConfig& stream_config,
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std::vector<float*>* frame,
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std::vector<float>* frame_samples) {
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frame_samples->resize(stream_config.num_channels() *
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@ -25,30 +27,28 @@ void SetupFrame(StreamConfig stream_config,
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}
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void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
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const std::vector<float>& source,
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rtc::ArrayView<const float> source,
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AudioBuffer* destination) {
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std::vector<float*> input;
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std::vector<float> input_samples;
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SetupFrame(stream_config, &input, &input_samples);
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RTC_DCHECK_EQ(input_samples.size(), source.size());
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input_samples = source;
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RTC_CHECK_EQ(input_samples.size(), source.size());
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memcpy(input_samples.data(), source.data(),
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source.size() * sizeof(source[0]));
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destination->CopyFrom(&input[0], stream_config);
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}
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std::vector<float> ExtractVectorFromAudioBuffer(
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const StreamConfig& stream_config,
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AudioBuffer* source) {
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void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
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AudioBuffer* source,
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std::vector<float>* destination) {
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std::vector<float*> output;
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std::vector<float> output_samples;
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||||
SetupFrame(stream_config, &output, &output_samples);
|
||||
SetupFrame(stream_config, &output, destination);
|
||||
|
||||
source->CopyTo(stream_config, &output[0]);
|
||||
|
||||
return output_samples;
|
||||
}
|
||||
|
||||
} // namespace test
|
||||
|
||||
@ -12,6 +12,7 @@
|
||||
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
|
||||
|
||||
#include <vector>
|
||||
#include "webrtc/base/array_view.h"
|
||||
#include "webrtc/modules/audio_processing/audio_buffer.h"
|
||||
#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
@ -20,13 +21,13 @@ namespace test {
|
||||
|
||||
// Copies a vector into an audiobuffer.
|
||||
void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
|
||||
const std::vector<float>& source,
|
||||
rtc::ArrayView<const float> source,
|
||||
AudioBuffer* destination);
|
||||
|
||||
// Extracts a vector from an audiobuffer.
|
||||
std::vector<float> ExtractVectorFromAudioBuffer(
|
||||
const StreamConfig& stream_config,
|
||||
AudioBuffer* source);
|
||||
void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
|
||||
AudioBuffer* source,
|
||||
std::vector<float>* destination);
|
||||
|
||||
} // namespace test
|
||||
} // namespace webrtc
|
||||
|
||||
145
webrtc/modules/audio_processing/test/bitexactness_tools.cc
Normal file
145
webrtc/modules/audio_processing/test/bitexactness_tools.cc
Normal file
@ -0,0 +1,145 @@
|
||||
/*
|
||||
* Copyright (c) 2016 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_processing/test/bitexactness_tools.h"
|
||||
|
||||
#include <math.h>
|
||||
#include <algorithm>
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/base/array_view.h"
|
||||
#include "webrtc/test/testsupport/fileutils.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
||||
std::string GetApmRenderTestVectorFileName(int sample_rate_hz) {
|
||||
switch (sample_rate_hz) {
|
||||
case 8000:
|
||||
return ResourcePath("far8_stereo", "pcm");
|
||||
case 16000:
|
||||
return ResourcePath("far16_stereo", "pcm");
|
||||
case 32000:
|
||||
return ResourcePath("far32_stereo", "pcm");
|
||||
case 48000:
|
||||
return ResourcePath("far48_stereo", "pcm");
|
||||
default:
|
||||
RTC_DCHECK(false);
|
||||
}
|
||||
return "";
|
||||
}
|
||||
|
||||
std::string GetApmCaptureTestVectorFileName(int sample_rate_hz) {
|
||||
switch (sample_rate_hz) {
|
||||
case 8000:
|
||||
return ResourcePath("near8_stereo", "pcm");
|
||||
case 16000:
|
||||
return ResourcePath("near16_stereo", "pcm");
|
||||
case 32000:
|
||||
return ResourcePath("near32_stereo", "pcm");
|
||||
case 48000:
|
||||
return ResourcePath("near48_stereo", "pcm");
|
||||
default:
|
||||
RTC_DCHECK(false);
|
||||
}
|
||||
return "";
|
||||
}
|
||||
|
||||
void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
|
||||
size_t num_channels,
|
||||
InputAudioFile* stereo_pcm_file,
|
||||
rtc::ArrayView<float> data) {
|
||||
RTC_DCHECK_EQ(data.size(), samples_per_channel * num_channels);
|
||||
std::vector<int16_t> read_samples(samples_per_channel * 2);
|
||||
stereo_pcm_file->Read(samples_per_channel * 2, read_samples.data());
|
||||
|
||||
// Convert samples to float and discard any channels not needed.
|
||||
for (size_t sample = 0; sample < samples_per_channel; ++sample) {
|
||||
for (size_t channel = 0; channel < num_channels; ++channel) {
|
||||
data[sample * num_channels + channel] =
|
||||
read_samples[sample * 2 + channel] / 32768.0f;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
::testing::AssertionResult BitExactFrame(size_t samples_per_channel,
|
||||
size_t num_channels,
|
||||
rtc::ArrayView<const float> reference,
|
||||
rtc::ArrayView<const float> output,
|
||||
float tolerance) {
|
||||
// Form vectors to compare the reference to. Only the first values of the
|
||||
// outputs are compared in order not having to specify all preceeding frames
|
||||
// as testvectors.
|
||||
const size_t reference_frame_length =
|
||||
rtc::CheckedDivExact(reference.size(), num_channels);
|
||||
|
||||
std::vector<float> output_to_verify;
|
||||
for (size_t channel_no = 0; channel_no < num_channels; ++channel_no) {
|
||||
output_to_verify.insert(output_to_verify.end(),
|
||||
output.begin() + channel_no * samples_per_channel,
|
||||
output.begin() + channel_no * samples_per_channel +
|
||||
reference_frame_length);
|
||||
}
|
||||
|
||||
return BitExactVector(reference, output_to_verify, tolerance);
|
||||
}
|
||||
|
||||
::testing::AssertionResult BitExactVector(rtc::ArrayView<const float> reference,
|
||||
rtc::ArrayView<const float> output,
|
||||
float tolerance) {
|
||||
// The vectors are deemed to be bitexact only if
|
||||
// a) output have a size at least as long as the reference.
|
||||
// b) the samples in the reference are bitexact with the corresponding samples
|
||||
// in the output.
|
||||
|
||||
bool equal = true;
|
||||
if (output.size() < reference.size()) {
|
||||
equal = false;
|
||||
} else {
|
||||
// Compare the first samples in the vectors.
|
||||
for (size_t k = 0; k < reference.size(); ++k) {
|
||||
if (fabs(output[k] - reference[k]) > tolerance) {
|
||||
equal = false;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if (equal) {
|
||||
return ::testing::AssertionSuccess();
|
||||
}
|
||||
|
||||
// Lambda function that produces a formatted string with the data in the
|
||||
// vector.
|
||||
auto print_vector_in_c_format = [](rtc::ArrayView<const float> v,
|
||||
size_t num_values_to_print) {
|
||||
std::string s = "{ ";
|
||||
for (size_t k = 0; k < std::min(num_values_to_print, v.size()); ++k) {
|
||||
s += std::to_string(v[k]) + "f";
|
||||
s += (k < (num_values_to_print - 1)) ? ", " : "";
|
||||
}
|
||||
return s + " }";
|
||||
};
|
||||
|
||||
// If the vectors are deemed not to be similar, return a report of the
|
||||
// difference.
|
||||
return ::testing::AssertionFailure()
|
||||
<< std::endl
|
||||
<< " Actual values : "
|
||||
<< print_vector_in_c_format(output,
|
||||
std::min(output.size(), reference.size()))
|
||||
<< std::endl
|
||||
<< " Expected values: "
|
||||
<< print_vector_in_c_format(reference, reference.size()) << std::endl;
|
||||
}
|
||||
|
||||
} // namespace test
|
||||
} // namespace webrtc
|
||||
54
webrtc/modules/audio_processing/test/bitexactness_tools.h
Normal file
54
webrtc/modules/audio_processing/test/bitexactness_tools.h
Normal file
@ -0,0 +1,54 @@
|
||||
/*
|
||||
* Copyright (c) 2016 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_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
|
||||
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
|
||||
|
||||
#include <string>
|
||||
|
||||
#include "webrtc/base/array_view.h"
|
||||
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
|
||||
#include "testing/gtest/include/gtest/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
||||
// Returns test vector to use for the render signal in an
|
||||
// APM bitexactness test.
|
||||
std::string GetApmRenderTestVectorFileName(int sample_rate_hz);
|
||||
|
||||
// Returns test vector to use for the capture signal in an
|
||||
// APM bitexactness test.
|
||||
std::string GetApmCaptureTestVectorFileName(int sample_rate_hz);
|
||||
|
||||
// Extract float samples from a pcm file.
|
||||
void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
|
||||
size_t num_channels,
|
||||
InputAudioFile* stereo_pcm_file,
|
||||
rtc::ArrayView<float> data);
|
||||
|
||||
// Verifies a frame against a reference and returns the results as an
|
||||
// AssertionResult.
|
||||
::testing::AssertionResult BitExactFrame(size_t samples_per_channel,
|
||||
size_t num_channels,
|
||||
rtc::ArrayView<const float> reference,
|
||||
rtc::ArrayView<const float> output,
|
||||
float tolerance);
|
||||
|
||||
// Verifies a vector against a reference and returns the results as an
|
||||
// AssertionResult.
|
||||
::testing::AssertionResult BitExactVector(rtc::ArrayView<const float> reference,
|
||||
rtc::ArrayView<const float> output,
|
||||
float tolerance);
|
||||
|
||||
} // namespace test
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
|
||||
@ -441,9 +441,12 @@
|
||||
'audio_processing/audio_processing_impl_locking_unittest.cc',
|
||||
'audio_processing/audio_processing_impl_unittest.cc',
|
||||
'audio_processing/high_pass_filter_bitexactness_unittest.cc',
|
||||
'audio_processing/noise_suppression_bitexactness_unittest.cc',
|
||||
'audio_processing/test/audio_buffer_tools.cc',
|
||||
'audio_processing/test/audio_buffer_tools.h',
|
||||
'audio_processing/test/audio_processing_unittest.cc',
|
||||
'audio_processing/test/bitexactness_tools.cc',
|
||||
'audio_processing/test/bitexactness_tools.h',
|
||||
'audio_processing/test/debug_dump_replayer.cc',
|
||||
'audio_processing/test/debug_dump_replayer.h',
|
||||
'audio_processing/test/debug_dump_test.cc',
|
||||
|
||||
Loading…
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Reference in New Issue
Block a user