webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
Karl Wiberg 918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00

91 lines
2.9 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 <utility>
#include "absl/memory/memory.h"
#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
#include "modules/audio_processing/include/audio_processing.h"
using testing::_;
using testing::AtLeast;
using testing::Exactly;
using testing::Matcher;
using testing::StrictMock;
namespace {
std::unique_ptr<webrtc::AudioProcessing> CreateAudioProcessing() {
webrtc::Config config;
std::unique_ptr<webrtc::AudioProcessing> apm(
webrtc::AudioProcessingBuilder().Create(config));
RTC_DCHECK(apm);
return apm;
}
std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
auto mock_aec_dump =
absl::make_unique<testing::StrictMock<webrtc::test::MockAecDump>>();
EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_)).Times(AtLeast(1));
return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
}
std::unique_ptr<webrtc::AudioFrame> CreateFakeFrame() {
auto fake_frame = absl::make_unique<webrtc::AudioFrame>();
fake_frame->num_channels_ = 1;
fake_frame->sample_rate_hz_ = 48000;
fake_frame->samples_per_channel_ = 480;
return fake_frame;
}
} // namespace
TEST(AecDumpIntegration, ConfigurationAndInitShouldBeLogged) {
auto apm = CreateAudioProcessing();
apm->AttachAecDump(CreateMockAecDump());
}
TEST(AecDumpIntegration,
RenderStreamShouldBeLoggedOnceEveryProcessReverseStream) {
auto apm = CreateAudioProcessing();
auto mock_aec_dump = CreateMockAecDump();
auto fake_frame = CreateFakeFrame();
EXPECT_CALL(*mock_aec_dump.get(),
WriteRenderStreamMessage(Matcher<const webrtc::AudioFrame&>(_)))
.Times(Exactly(1));
apm->AttachAecDump(std::move(mock_aec_dump));
apm->ProcessReverseStream(fake_frame.get());
}
TEST(AecDumpIntegration, CaptureStreamShouldBeLoggedOnceEveryProcessStream) {
auto apm = CreateAudioProcessing();
auto mock_aec_dump = CreateMockAecDump();
auto fake_frame = CreateFakeFrame();
EXPECT_CALL(*mock_aec_dump.get(),
AddCaptureStreamInput(Matcher<const webrtc::AudioFrame&>(_)))
.Times(AtLeast(1));
EXPECT_CALL(*mock_aec_dump.get(),
AddCaptureStreamOutput(Matcher<const webrtc::AudioFrame&>(_)))
.Times(Exactly(1));
EXPECT_CALL(*mock_aec_dump.get(), AddAudioProcessingState(_))
.Times(Exactly(1));
EXPECT_CALL(*mock_aec_dump.get(), WriteCaptureStreamMessage())
.Times(Exactly(1));
apm->AttachAecDump(std::move(mock_aec_dump));
apm->ProcessStream(fake_frame.get());
}