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This CL adds a build flag to allow building the non-test parts of WebRTC without the audio processing module. The CL also ensures that the WebRTC code correctly handles the case when no APM is available. Bug: webrtc:5298 Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31133}
96 lines
3.4 KiB
C++
96 lines
3.4 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 <array>
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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/audio_processing_impl.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/test/audio_processing_builder_for_testing.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::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::AudioProcessingBuilderForTesting().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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} // 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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constexpr int kNumChannels = 1;
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constexpr int kNumSampleRateHz = 16000;
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constexpr int kNumSamplesPerChannel = kNumSampleRateHz / 100;
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std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
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frame.fill(0.f);
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webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels,
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/*has_keyboard=*/false);
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EXPECT_CALL(*mock_aec_dump.get(), WriteRenderStreamMessage(_, _, _))
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.Times(Exactly(1));
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apm->AttachAecDump(std::move(mock_aec_dump));
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apm->ProcessReverseStream(frame.data(), stream_config, stream_config,
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frame.data());
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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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constexpr int kNumChannels = 1;
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constexpr int kNumSampleRateHz = 16000;
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constexpr int kNumSamplesPerChannel = kNumSampleRateHz / 100;
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std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
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frame.fill(0.f);
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webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels,
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/*has_keyboard=*/false);
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EXPECT_CALL(*mock_aec_dump.get(), AddCaptureStreamInput(_, _, _))
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.Times(AtLeast(1));
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EXPECT_CALL(*mock_aec_dump.get(), AddCaptureStreamOutput(_, _, _))
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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(frame.data(), stream_config, stream_config, frame.data());
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}
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