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Revert "Separate test/fake_audio_device on API and implementation."

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This reverts commit 8ea5f9ae5b757aa3a0e6abe46f5c9ef3aaf4b337.

Reason for revert: breaks downstream project

Original change's description:
> Separate test/fake_audio_device on API and implementation.
> 
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
> 
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> 
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}

TBR=kwiberg@webrtc.org,titovartem@webrtc.org

Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
This commit is contained in:
Artem Titov
2018-03-05 15:36:09 +00:00
committed by Commit Bot
parent f9f71b91ae
commit 6723cdc8a4
20 changed files with 723 additions and 797 deletions
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146
modules/audio_device/include/test_audio_device_unittest.cc
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@ -1,146 +0,0 @@
/*
* Copyright (c) 2017 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 <algorithm>
#include <array>
#include "common_audio/wav_file.h"
#include "common_audio/wav_header.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
namespace {
void RunTest(const std::vector<int16_t>& input_samples,
const std::vector<int16_t>& expected_samples,
size_t samples_per_frame) {
const ::testing::TestInfo* const test_info =
::testing::UnitTest::GetInstance()->current_test_info();
const std::string output_filename = test::OutputPath() +
"BoundedWavFileWriterTest_" +
test_info->name() + ".wav";
static const size_t kSamplesPerFrame = 8;
static const int kSampleRate = kSamplesPerFrame * 100;
EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
kSamplesPerFrame);
{
std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
TestAudioDeviceModule::CreateBoundedWavFileWriter(output_filename, 800);
for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
&input_samples[i],
std::min(kSamplesPerFrame, input_samples.size() - i))));
}
}
{
WavReader reader(output_filename);
std::vector<int16_t> read_samples(expected_samples.size());
EXPECT_EQ(expected_samples.size(),
reader.ReadSamples(read_samples.size(), read_samples.data()));
EXPECT_EQ(expected_samples, read_samples);
EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
}
remove(output_filename.c_str());
}
} // namespace
TEST(BoundedWavFileWriterTest, NoSilence) {
static const std::vector<int16_t> kInputSamples = {
75, 1234, 243, -1231, -22222, 0, 3, 88,
1222, -1213, -13222, -7, -3525, 5787, -25247, 8};
static const std::vector<int16_t> kExpectedSamples = kInputSamples;
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(BoundedWavFileWriterTest, SomeStartSilence) {
static const std::vector<int16_t> kInputSamples = {
0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
kInputSamples.end());
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(BoundedWavFileWriterTest, NegativeStartSilence) {
static const std::vector<int16_t> kInputSamples = {
0, -4, -6, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
kInputSamples.end());
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(BoundedWavFileWriterTest, SomeEndSilence) {
static const std::vector<int16_t> kInputSamples = {
75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0};
static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
kInputSamples.end() - 9);
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(BoundedWavFileWriterTest, DoubleEndSilence) {
static const std::vector<int16_t> kInputSamples = {
75, 1234, 243, -1231, -22222, 0, 0, 0,
0, -1213, -13222, -7, -3525, 5787, 0, 0};
static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
kInputSamples.end() - 2);
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(BoundedWavFileWriterTest, DoubleSilence) {
static const std::vector<int16_t> kInputSamples = {0, -1213, -13222, -7,
-3525, 5787, 0, 0};
static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
kInputSamples.end() - 2);
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(BoundedWavFileWriterTest, EndSilenceCutoff) {
static const std::vector<int16_t> kInputSamples = {
75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0};
static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
kInputSamples.end() - 4);
RunTest(kInputSamples, kExpectedSamples, 8);
}
TEST(PulsedNoiseCapturerTest, SetMaxAmplitude) {
const int16_t kAmplitude = 50;
std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
TestAudioDeviceModule::CreatePulsedNoiseCapturer(
kAmplitude, /*sampling_frequency_in_hz=*/8000);
rtc::BufferT<int16_t> recording_buffer;
// Verify that the capturer doesn't create entries louder than than
// kAmplitude. Since the pulse generator alternates between writing
// zeroes and actual entries, we need to do the capturing twice.
capturer->Capture(&recording_buffer);
capturer->Capture(&recording_buffer);
int16_t max_sample =
*std::max_element(recording_buffer.begin(), recording_buffer.end());
EXPECT_LE(max_sample, kAmplitude);
// Increase the amplitude and verify that the samples can now be louder
// than the previous max.
capturer->SetMaxAmplitude(kAmplitude * 2);
capturer->Capture(&recording_buffer);
capturer->Capture(&recording_buffer);
max_sample =
*std::max_element(recording_buffer.begin(), recording_buffer.end());
EXPECT_GT(max_sample, kAmplitude);
}
} // namespace webrtc