890988c9cb
This CL contains automatically applied fixes suggested by the ClangTidy analyzer (http://clang.llvm.org/extra/clang-tidy/). The following kinds of fixes is present: * renaming variables when the names in the method signature don't match the names in the method definition (ClangTidy:readability-inconsistent-declaration-parameter-name) * ClangTidy:readability-container-size-empty, ClangTidy:misc-unused-using-decls, ClangTidy:performance-unnecessary-value-param, ClangTidy:readability-redundant-control-flow This is a 'pilot' CL to check if automatic code analyzers can feasibly be integrated into the WebRTC infrastructuve. The renamings have been manually expected for consistency with surrounding code. In echo_cancellation.cc, I changed several names in the function implementation to match the function declaration. The tool suggested changing everything to match the function definitions instead. Bug: None Change-Id: Id3b7ba18c51f15b025f26090c7bdcc642e48d8fd Reviewed-on: https://chromium-review.googlesource.com/635766 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19630}
80 lines
2.5 KiB
C++
80 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2014 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 "webrtc/modules/audio_processing/audio_processing_impl.h"
|
|
|
|
#include "webrtc/config.h"
|
|
#include "webrtc/modules/audio_processing/test/test_utils.h"
|
|
#include "webrtc/modules/include/module_common_types.h"
|
|
#include "webrtc/test/gmock.h"
|
|
#include "webrtc/test/gtest.h"
|
|
|
|
using ::testing::Invoke;
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
class MockInitialize : public AudioProcessingImpl {
|
|
public:
|
|
explicit MockInitialize(const webrtc::Config& config)
|
|
: AudioProcessingImpl(config) {}
|
|
|
|
MOCK_METHOD0(InitializeLocked, int());
|
|
int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS {
|
|
return AudioProcessingImpl::InitializeLocked();
|
|
}
|
|
|
|
MOCK_CONST_METHOD0(AddRef, int());
|
|
MOCK_CONST_METHOD0(Release, int());
|
|
};
|
|
|
|
} // namespace
|
|
|
|
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
|
|
webrtc::Config config;
|
|
MockInitialize mock(config);
|
|
ON_CALL(mock, InitializeLocked())
|
|
.WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
|
|
|
|
EXPECT_CALL(mock, InitializeLocked()).Times(1);
|
|
mock.Initialize();
|
|
|
|
AudioFrame frame;
|
|
// Call with the default parameters; there should be an init.
|
|
frame.num_channels_ = 1;
|
|
SetFrameSampleRate(&frame, 16000);
|
|
EXPECT_CALL(mock, InitializeLocked()).Times(0);
|
|
EXPECT_NOERR(mock.ProcessStream(&frame));
|
|
EXPECT_NOERR(mock.ProcessReverseStream(&frame));
|
|
|
|
// New sample rate. (Only impacts ProcessStream).
|
|
SetFrameSampleRate(&frame, 32000);
|
|
EXPECT_CALL(mock, InitializeLocked())
|
|
.Times(1);
|
|
EXPECT_NOERR(mock.ProcessStream(&frame));
|
|
|
|
// New number of channels.
|
|
// TODO(peah): Investigate why this causes 2 inits.
|
|
frame.num_channels_ = 2;
|
|
EXPECT_CALL(mock, InitializeLocked())
|
|
.Times(2);
|
|
EXPECT_NOERR(mock.ProcessStream(&frame));
|
|
// ProcessStream sets num_channels_ == num_output_channels.
|
|
frame.num_channels_ = 2;
|
|
EXPECT_NOERR(mock.ProcessReverseStream(&frame));
|
|
|
|
// A new sample rate passed to ProcessReverseStream should cause an init.
|
|
SetFrameSampleRate(&frame, 16000);
|
|
EXPECT_CALL(mock, InitializeLocked()).Times(1);
|
|
EXPECT_NOERR(mock.ProcessReverseStream(&frame));
|
|
}
|
|
|
|
} // namespace webrtc
|