mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00

The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder. Bug: webrtc:8668 Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67 Reviewed-on: https://webrtc-review.googlesource.com/36082 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21534}
91 lines
2.9 KiB
C++
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 "modules/audio_processing/aec_dump/mock_aec_dump.h"
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
#include "rtc_base/ptr_util.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 =
|
|
rtc::MakeUnique<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 = rtc::MakeUnique<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());
|
|
}
|