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317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
91 lines
2.9 KiB
C++
91 lines
2.9 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 <memory>
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#include <utility>
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#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
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#include "modules/audio_processing/include/audio_processing.h"
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using ::testing::_;
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using ::testing::AtLeast;
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using ::testing::Exactly;
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using ::testing::Matcher;
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using ::testing::StrictMock;
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namespace {
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std::unique_ptr<webrtc::AudioProcessing> CreateAudioProcessing() {
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webrtc::Config config;
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std::unique_ptr<webrtc::AudioProcessing> apm(
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webrtc::AudioProcessingBuilder().Create(config));
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RTC_DCHECK(apm);
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return apm;
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}
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std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
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auto mock_aec_dump =
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std::make_unique<testing::StrictMock<webrtc::test::MockAecDump>>();
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EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
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EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_, _)).Times(AtLeast(1));
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return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
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}
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std::unique_ptr<webrtc::AudioFrame> CreateFakeFrame() {
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auto fake_frame = std::make_unique<webrtc::AudioFrame>();
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fake_frame->num_channels_ = 1;
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fake_frame->sample_rate_hz_ = 48000;
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fake_frame->samples_per_channel_ = 480;
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return fake_frame;
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}
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} // namespace
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TEST(AecDumpIntegration, ConfigurationAndInitShouldBeLogged) {
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auto apm = CreateAudioProcessing();
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apm->AttachAecDump(CreateMockAecDump());
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}
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TEST(AecDumpIntegration,
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RenderStreamShouldBeLoggedOnceEveryProcessReverseStream) {
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auto apm = CreateAudioProcessing();
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auto mock_aec_dump = CreateMockAecDump();
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auto fake_frame = CreateFakeFrame();
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EXPECT_CALL(*mock_aec_dump.get(),
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WriteRenderStreamMessage(Matcher<const webrtc::AudioFrame&>(_)))
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.Times(Exactly(1));
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apm->AttachAecDump(std::move(mock_aec_dump));
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apm->ProcessReverseStream(fake_frame.get());
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}
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TEST(AecDumpIntegration, CaptureStreamShouldBeLoggedOnceEveryProcessStream) {
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auto apm = CreateAudioProcessing();
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auto mock_aec_dump = CreateMockAecDump();
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auto fake_frame = CreateFakeFrame();
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EXPECT_CALL(*mock_aec_dump.get(),
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AddCaptureStreamInput(Matcher<const webrtc::AudioFrame&>(_)))
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.Times(AtLeast(1));
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EXPECT_CALL(*mock_aec_dump.get(),
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AddCaptureStreamOutput(Matcher<const webrtc::AudioFrame&>(_)))
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.Times(Exactly(1));
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EXPECT_CALL(*mock_aec_dump.get(), AddAudioProcessingState(_))
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.Times(Exactly(1));
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EXPECT_CALL(*mock_aec_dump.get(), WriteCaptureStreamMessage())
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.Times(Exactly(1));
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apm->AttachAecDump(std::move(mock_aec_dump));
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apm->ProcessStream(fake_frame.get());
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}
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