Add Android.mk for apm unit test and make it compile on android
Review URL: http://webrtc-codereview.appspot.com/54001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@132 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -42,7 +42,7 @@ LOCAL_SHARED_LIBRARIES := \
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libstlport \
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libwebrtc_audio_preprocessing
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LOCAL_MODULE:= webrtc-apmtest
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LOCAL_MODULE:= webrtc_apm_process_test
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include external/stlport/libstlport.mk
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include $(BUILD_EXECUTABLE)
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49
modules/audio_processing/main/test/unit_test/Android.mk
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49
modules/audio_processing/main/test/unit_test/Android.mk
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@ -0,0 +1,49 @@
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# Copyright (c) 2011 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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LOCAL_PATH:= $(call my-dir)
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# apm test app
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include $(CLEAR_VARS)
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LOCAL_MODULE_TAGS := tests
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LOCAL_CPP_EXTENSION := .cc
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LOCAL_SRC_FILES:= \
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unit_test.cc
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# Flags passed to both C and C++ files.
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LOCAL_CFLAGS := \
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'-DWEBRTC_TARGET_PC' \
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'-DWEBRTC_LINUX' \
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'-DWEBRTC_THREAD_RR' \
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'-DWEBRTC_ANDROID' \
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'-DANDROID'
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LOCAL_CPPFLAGS :=
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LOCAL_LDFLAGS :=
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LOCAL_C_INCLUDES := \
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external/gtest/include \
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$(LOCAL_PATH)/../../../../../system_wrappers/interface \
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$(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface \
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$(LOCAL_PATH)/../../interface \
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$(LOCAL_PATH)/../../../../interface \
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$(LOCAL_PATH)/../../../../..
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LOCAL_STATIC_LIBRARIES := \
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libgtest
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LOCAL_SHARED_LIBRARIES := \
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libutils \
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libstlport \
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libwebrtc_audio_preprocessing
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LOCAL_MODULE:= webrtc_apm_unit_test
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include external/stlport/libstlport.mk
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include $(BUILD_EXECUTABLE)
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@ -10,15 +10,29 @@
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#include "unit_test.h"
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#include "tick_util.h"
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#include "cpu_features_wrapper.h"
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#include "event_wrapper.h"
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#include "module_common_types.h"
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#include "thread_wrapper.h"
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#include "trace.h"
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#include "signal_processing_library.h"
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#include "audio_processing.h"
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namespace webrtc {
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using webrtc::AudioProcessing;
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using webrtc::AudioFrame;
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using webrtc::GainControl;
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using webrtc::NoiseSuppression;
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using webrtc::EchoCancellation;
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using webrtc::TickInterval;
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using webrtc::TickTime;
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using webrtc::EventWrapper;
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using webrtc::Trace;
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using webrtc::LevelEstimator;
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using webrtc::EchoCancellation;
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using webrtc::EchoControlMobile;
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using webrtc::VoiceDetection;
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namespace {
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// If false, this will write out a new statistics file.
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@ -465,7 +479,7 @@ TEST_F(ApmTest, Process) {
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sizeof(WebRtc_Word16),
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render_audio._payloadDataLengthInSamples * 2,
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far_file_);
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if (read_count !=
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if (read_count !=
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static_cast<size_t>
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(render_audio._payloadDataLengthInSamples * 2)) {
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break; // This is expected.
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@ -1011,4 +1025,3 @@ int main(int argc, char** argv) {
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return RUN_ALL_TESTS();
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}
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}
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@ -14,9 +14,9 @@
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#include <gtest/gtest.h>
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namespace webrtc {
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class AudioProcessing;
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class AudioFrame;
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}
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class ApmTest : public ::testing::Test {
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protected:
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@ -28,9 +28,8 @@ class ApmTest : public ::testing::Test {
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FILE* far_file_;
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FILE* near_file_;
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FILE* stat_file_;
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AudioFrame* frame_;
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AudioFrame* reverse_frame_;
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webrtc::AudioFrame* frame_;
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webrtc::AudioFrame* reverse_frame_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_TEST_UNIT_TEST_UNIT_TEST_H_
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