webrtc_m130/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
peah a9cc40b7d2 Allow an external audio processing module to be used in WebRTC
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]

Allow an external audio processing module to be used in WebRTC

This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.

As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.

BUG=webrtc:7775

Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
2017-06-29 15:32:09 +00:00

81 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;
using ::testing::Return;
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