webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
Per Åhgren 38e2d95bda AEC3 delay estimator refactoring and introducing ability to customize
This CL refactors the delay estimator in AEC3.
Furthermore, it adds:
1. Allow for a customized delay estimator behavior to simplify
development.
2. Exposes that behavior to clear configuration settings.
3. Adds logging of the delay range supported by the delay
estimator.

Bug: webrtc:8519
Change-Id: I1764a090519a78b021b2e7de565c52a6c02c848e
Reviewed-on: https://webrtc-review.googlesource.com/21166
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20733}
2017-11-17 17:51:37 +00:00

125 lines
4.8 KiB
C++

/*
* 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 "modules/audio_processing/aec3/render_delay_buffer.h"
#include <memory>
#include <sstream>
#include <string>
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/random.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
std::string ProduceDebugText(int sample_rate_hz) {
std::ostringstream ss;
ss << "Sample rate: " << sample_rate_hz;
return ss.str();
}
constexpr size_t kDownSamplingFactor = 4;
constexpr size_t kNumMatchedFilters = 4;
} // namespace
// Verifies that the buffer overflow is correctly reported.
TEST(RenderDelayBuffer, BufferOverflow) {
for (auto rate : {8000, 16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
NumBandsForRate(rate), kDownSamplingFactor,
GetDownSampledBufferSize(kDownSamplingFactor, kNumMatchedFilters),
GetRenderDelayBufferSize(kDownSamplingFactor, kNumMatchedFilters)));
std::vector<std::vector<float>> block_to_insert(
NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
for (size_t k = 0; k < kMaxApiCallsJitterBlocks; ++k) {
EXPECT_TRUE(delay_buffer->Insert(block_to_insert));
}
EXPECT_FALSE(delay_buffer->Insert(block_to_insert));
}
}
// Verifies that the check for available block works.
TEST(RenderDelayBuffer, AvailableBlock) {
constexpr size_t kNumBands = 1;
std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
kNumBands, kDownSamplingFactor,
GetDownSampledBufferSize(kDownSamplingFactor, kNumMatchedFilters),
GetRenderDelayBufferSize(kDownSamplingFactor, kNumMatchedFilters)));
std::vector<std::vector<float>> input_block(
kNumBands, std::vector<float>(kBlockSize, 1.f));
EXPECT_TRUE(delay_buffer->Insert(input_block));
delay_buffer->UpdateBuffers();
}
// Verifies the SetDelay method.
TEST(RenderDelayBuffer, SetDelay) {
std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
1, kDownSamplingFactor,
GetDownSampledBufferSize(kDownSamplingFactor, kNumMatchedFilters),
GetRenderDelayBufferSize(kDownSamplingFactor, kNumMatchedFilters)));
EXPECT_EQ(0u, delay_buffer->Delay());
for (size_t delay = 0; delay < 20; ++delay) {
delay_buffer->SetDelay(delay);
EXPECT_EQ(delay, delay_buffer->Delay());
}
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for feasible delay.
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.
TEST(RenderDelayBuffer, DISABLED_WrongDelay) {
std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
3, kDownSamplingFactor,
GetDownSampledBufferSize(kDownSamplingFactor, kNumMatchedFilters),
GetRenderDelayBufferSize(kDownSamplingFactor, kNumMatchedFilters)));
EXPECT_DEATH(delay_buffer->SetDelay(21), "");
}
// Verifies the check for the number of bands in the inserted blocks.
TEST(RenderDelayBuffer, WrongNumberOfBands) {
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
NumBandsForRate(rate), kDownSamplingFactor,
GetDownSampledBufferSize(kDownSamplingFactor, kNumMatchedFilters),
GetRenderDelayBufferSize(kDownSamplingFactor, kNumMatchedFilters)));
std::vector<std::vector<float>> block_to_insert(
NumBandsForRate(rate < 48000 ? rate + 16000 : 16000),
std::vector<float>(kBlockSize, 0.f));
EXPECT_DEATH(delay_buffer->Insert(block_to_insert), "");
}
}
// Verifies the check of the length of the inserted blocks.
TEST(RenderDelayBuffer, WrongBlockLength) {
for (auto rate : {8000, 16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
3, kDownSamplingFactor,
GetDownSampledBufferSize(kDownSamplingFactor, kNumMatchedFilters),
GetRenderDelayBufferSize(kDownSamplingFactor, kNumMatchedFilters)));
std::vector<std::vector<float>> block_to_insert(
NumBandsForRate(rate), std::vector<float>(kBlockSize - 1, 0.f));
EXPECT_DEATH(delay_buffer->Insert(block_to_insert), "");
}
}
#endif
} // namespace webrtc