This reverts commit eeae96299784515f573379a64655eb07a5973a3a. Reason for revert: breaks WebRTC Chromium FYI ios-device https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/14896/overview Original change's description: > Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" > > This reverts commit 69c8d3c843326aff9dee32cc639741c1cd7f8ae9. > > Reason for revert: Reland with a fix > > Original change's description: > > Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" > > > > This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b. > > > > Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814 > > > > Original change's description: > > > Migrate TestAudioDeviceModule on AudioDeviceModuleImpl > > > > > > Bug: b/272350185 > > > Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363 > > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > > Cr-Commit-Position: refs/heads/main@{#39877} > > > > Bug: b/272350185 > > Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701 > > Commit-Queue: Jeremy Leconte <jleconte@google.com> > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > > Reviewed-by: Jeremy Leconte <jleconte@google.com> > > Auto-Submit: Christoffer Jansson <jansson@google.com> > > Owners-Override: Christoffer Jansson <jansson@google.com> > > Cr-Commit-Position: refs/heads/main@{#39881} > > Bug: b/272350185 > Change-Id: I809466306b2e1fd54c44b90311059c98a53ef8ee > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301704 > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39936} Bug: b/272350185 Change-Id: If0a10717bf14a0a618e52728fc3a61b9c55f3bd2 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303460 Commit-Queue: Jeremy Leconte <jleconte@google.com> Owners-Override: Jeremy Leconte <jleconte@google.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#39947}
193 lines
7.5 KiB
C++
193 lines
7.5 KiB
C++
/*
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* Copyright (c) 2017 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_device/include/test_audio_device.h"
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#include <algorithm>
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#include <array>
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#include "api/array_view.h"
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#include "common_audio/wav_file.h"
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#include "common_audio/wav_header.h"
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#include "rtc_base/logging.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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namespace {
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void RunTest(const std::vector<int16_t>& input_samples,
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const std::vector<int16_t>& expected_samples,
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size_t samples_per_frame) {
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const ::testing::TestInfo* const test_info =
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::testing::UnitTest::GetInstance()->current_test_info();
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const std::string output_filename =
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test::OutputPath() + "BoundedWavFileWriterTest_" + test_info->name() +
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"_" + std::to_string(std::rand()) + ".wav";
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static const size_t kSamplesPerFrame = 8;
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static const int kSampleRate = kSamplesPerFrame * 100;
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EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
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kSamplesPerFrame);
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// Test through file name API.
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{
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std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
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TestAudioDeviceModule::CreateBoundedWavFileWriter(output_filename, 800);
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for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
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EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
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&input_samples[i],
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std::min(kSamplesPerFrame, input_samples.size() - i))));
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}
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}
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{
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WavReader reader(output_filename);
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std::vector<int16_t> read_samples(expected_samples.size());
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EXPECT_EQ(expected_samples.size(),
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reader.ReadSamples(read_samples.size(), read_samples.data()));
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EXPECT_EQ(expected_samples, read_samples);
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EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
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}
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remove(output_filename.c_str());
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}
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} // namespace
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TEST(BoundedWavFileWriterTest, NoSilence) {
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static const std::vector<int16_t> kInputSamples = {
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75, 1234, 243, -1231, -22222, 0, 3, 88,
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1222, -1213, -13222, -7, -3525, 5787, -25247, 8};
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static const std::vector<int16_t> kExpectedSamples = kInputSamples;
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(BoundedWavFileWriterTest, SomeStartSilence) {
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static const std::vector<int16_t> kInputSamples = {
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0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
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static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
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kInputSamples.end());
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(BoundedWavFileWriterTest, NegativeStartSilence) {
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static const std::vector<int16_t> kInputSamples = {
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0, -4, -6, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
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static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
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kInputSamples.end());
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(BoundedWavFileWriterTest, SomeEndSilence) {
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static const std::vector<int16_t> kInputSamples = {
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75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0};
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static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
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kInputSamples.end() - 9);
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(BoundedWavFileWriterTest, DoubleEndSilence) {
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static const std::vector<int16_t> kInputSamples = {
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75, 1234, 243, -1231, -22222, 0, 0, 0,
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0, -1213, -13222, -7, -3525, 5787, 0, 0};
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static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
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kInputSamples.end() - 2);
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(BoundedWavFileWriterTest, DoubleSilence) {
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static const std::vector<int16_t> kInputSamples = {0, -1213, -13222, -7,
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-3525, 5787, 0, 0};
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static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
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kInputSamples.end() - 2);
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(BoundedWavFileWriterTest, EndSilenceCutoff) {
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static const std::vector<int16_t> kInputSamples = {
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75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0};
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static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
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kInputSamples.end() - 4);
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RunTest(kInputSamples, kExpectedSamples, 8);
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}
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TEST(WavFileReaderTest, RepeatedTrueWithSingleFrameFileReadTwice) {
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static const std::vector<int16_t> kInputSamples = {75, 1234, 243, -1231,
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-22222, 0, 3, 88};
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static const rtc::BufferT<int16_t> kExpectedSamples(kInputSamples.data(),
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kInputSamples.size());
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const std::string output_filename = test::OutputPath() +
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"WavFileReaderTest_RepeatedTrue_" +
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std::to_string(std::rand()) + ".wav";
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static const size_t kSamplesPerFrame = 8;
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static const int kSampleRate = kSamplesPerFrame * 100;
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EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
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kSamplesPerFrame);
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// Create wav file to read.
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{
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std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
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TestAudioDeviceModule::CreateWavFileWriter(output_filename, 800);
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for (size_t i = 0; i < kInputSamples.size(); i += kSamplesPerFrame) {
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EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
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&kInputSamples[i],
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std::min(kSamplesPerFrame, kInputSamples.size() - i))));
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}
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}
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{
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std::unique_ptr<TestAudioDeviceModule::Capturer> reader =
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TestAudioDeviceModule::CreateWavFileReader(output_filename, true);
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rtc::BufferT<int16_t> buffer(kExpectedSamples.size());
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EXPECT_TRUE(reader->Capture(&buffer));
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EXPECT_EQ(kExpectedSamples, buffer);
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EXPECT_TRUE(reader->Capture(&buffer));
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EXPECT_EQ(kExpectedSamples, buffer);
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}
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remove(output_filename.c_str());
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}
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TEST(PulsedNoiseCapturerTest, SetMaxAmplitude) {
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const int16_t kAmplitude = 50;
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std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
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TestAudioDeviceModule::CreatePulsedNoiseCapturer(
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kAmplitude, /*sampling_frequency_in_hz=*/8000);
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rtc::BufferT<int16_t> recording_buffer;
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// Verify that the capturer doesn't create entries louder than than
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// kAmplitude. Since the pulse generator alternates between writing
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// zeroes and actual entries, we need to do the capturing twice.
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capturer->Capture(&recording_buffer);
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capturer->Capture(&recording_buffer);
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int16_t max_sample =
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*std::max_element(recording_buffer.begin(), recording_buffer.end());
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EXPECT_LE(max_sample, kAmplitude);
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// Increase the amplitude and verify that the samples can now be louder
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// than the previous max.
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capturer->SetMaxAmplitude(kAmplitude * 2);
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capturer->Capture(&recording_buffer);
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capturer->Capture(&recording_buffer);
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max_sample =
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*std::max_element(recording_buffer.begin(), recording_buffer.end());
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EXPECT_GT(max_sample, kAmplitude);
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}
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} // namespace webrtc
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