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Eric Laurent
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Added webrtc audio processing library
Only the modules necessary for audio processing have been imported: src/common_audio/ src/modules/audio_processing/ src/modules/interface/ src/system_wrappers/ src/typedefs.h src/common_types.h Android.mk android-webrtc.mk Android.mk and android-webrtc.mk have been modified to build only the audio processing modules. Files for Windows compatibility have been removed from system_wrappers. fft_ARM9E directory has been removed from src/common_audio/signal_processing_library/main/source/ Fixed x86 build. SVN checkout at working revision 180. Change-Id: If650f61d96557be8247b17eb4f4d32b7a6ba025d
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‎Android.mk

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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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MY_WEBRTC_ROOT_PATH := $(call my-dir)
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include $(MY_WEBRTC_ROOT_PATH)/src/system_wrappers/source/Android.mk
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# audio processing
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include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing_library/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/main/source/Android.mk
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
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# build .so
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include $(MY_WEBRTC_ROOT_PATH)/android-webrtc.mk
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# build test apps
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include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/process_test/Android.mk

‎MODULE_LICENSE_BSD

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‎NOTICE

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===============================================================================
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/*
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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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*/
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===============================================================================
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/*
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* Copyright (C) 2010 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*
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*/
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===============================================================================
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/*
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* http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
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* Copyright Takuya OOURA, 1996-2001
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*
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* You may use, copy, modify and distribute this code for any purpose (include
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* commercial use) and without fee. Please refer to this package when you modify
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* this code.
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*
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* Changes by the WebRTC authors:
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* - Trivial type modifications.
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* - Minimal code subset to do rdft of length 128.
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* - Optimizations because of known length.
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*
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* All changes are covered by the WebRTC license and IP grant:
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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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===============================================================================
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/*
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* http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
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* Copyright Takuya OOURA, 1996-2001
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*
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* You may use, copy, modify and distribute this code for any purpose (include
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* commercial use) and without fee. Please refer to this package when you modify
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* this code.
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*
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* Changes:
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* Trivial type modifications by the WebRTC authors.
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*/
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===============================================================================
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/*Boost Software License - Version 1.0 - August 17th, 2003
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Permission is hereby granted, free of charge, to any person or organization
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obtaining a copy of the software and accompanying documentation covered by
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this license (the "Software") to use, reproduce, display, distribute,
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execute, and transmit the Software, and to prepare derivative works of the
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Software, and to permit third-parties to whom the Software is furnished to
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do so, all subject to the following:
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The copyright notices in the Software and this entire statement, including
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the above license grant, this restriction and the following disclaimer,
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must be included in all copies of the Software, in whole or in part, and
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all derivative works of the Software, unless such copies or derivative
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works are solely in the form of machine-executable object code generated by
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a source language processor.
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THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
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SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
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FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
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ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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DEALINGS IN THE SOFTWARE.*/
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===============================================================================
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// Copyright Steven J. Ross 2001 - 2009.
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// Distributed under the Boost Software License, Version 1.0.
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// (See accompanying file LICENSE_1_0.txt or copy at
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// http://www.boost.org/LICENSE_1_0.txt)
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// See http://www.boost.org/ for updates, documentation, and revision history.
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/*
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Some improvements suggested by:
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Phil Endecott and Frank Gennari
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Cygwin fix provided by:
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Scott McMurray
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*/
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‎android-webrtc.mk

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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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MY_APM_WHOLE_STATIC_LIBRARIES := \
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libwebrtc_spl \
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libwebrtc_resampler \
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libwebrtc_apm \
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libwebrtc_apm_utility \
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libwebrtc_vad \
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libwebrtc_ns \
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libwebrtc_agc \
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libwebrtc_aec \
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libwebrtc_aecm
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LOCAL_PATH := $(call my-dir)
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include $(CLEAR_VARS)
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LOCAL_ARM_MODE := arm
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LOCAL_MODULE := libwebrtc_audio_preprocessing
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LOCAL_MODULE_TAGS := optional
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LOCAL_LDFLAGS :=
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LOCAL_WHOLE_STATIC_LIBRARIES := \
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$(MY_APM_WHOLE_STATIC_LIBRARIES) \
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libwebrtc_system_wrappers \
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LOCAL_SHARED_LIBRARIES := \
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libcutils \
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libdl \
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libstlport
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LOCAL_ADDITIONAL_DEPENDENCIES :=
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include external/stlport/libstlport.mk
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include $(BUILD_SHARED_LIBRARY)
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###
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#LOCAL_PATH := $(call my-dir)
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#
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#include $(CLEAR_VARS)
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#
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#LOCAL_ARM_MODE := arm
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#LOCAL_MODULE := libwebrtc
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#LOCAL_MODULE_TAGS := optional
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#LOCAL_LDFLAGS :=
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#
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#LOCAL_WHOLE_STATIC_LIBRARIES := \
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# libwebrtc_system_wrappers \
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# libwebrtc_audio_device \
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# libwebrtc_pcm16b \
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# libwebrtc_cng \
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# libwebrtc_audio_coding \
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# libwebrtc_rtp_rtcp \
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# libwebrtc_media_file \
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# libwebrtc_udp_transport \
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# libwebrtc_utility \
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# libwebrtc_neteq \
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# libwebrtc_audio_conference_mixer \
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# libwebrtc_isac \
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# libwebrtc_ilbc \
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# libwebrtc_isacfix \
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# libwebrtc_g722 \
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# libwebrtc_g711 \
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# libwebrtc_voe_core \
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# libwebrtc_video_render \
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# libwebrtc_video_capture \
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# libwebrtc_i420 \
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# libwebrtc_video_coding \
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# libwebrtc_video_processing \
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# libwebrtc_vp8 \
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# libwebrtc_vie_core \
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# libwebrtc_vplib \
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# libwebrtc_jpeg \
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# libwebrtc_vpx
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#
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#LOCAL_STATIC_LIBRARIES :=
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#LOCAL_SHARED_LIBRARIES := \
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# libcutils \
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# libdl \
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# libstlport \
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# libjpeg \
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# libGLESv2 \
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# libOpenSLES \
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# libwebrtc_audio_preprocessing
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#
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#LOCAL_ADDITIONAL_DEPENDENCIES :=
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#
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#include external/stlport/libstlport.mk
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#include $(BUILD_SHARED_LIBRARY)

‎src/common_audio/OWNERS

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[email protected]

‎src/common_audio/resampler/OWNERS

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[email protected]
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[email protected]
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[email protected]

‎src/common_audio/resampler/main/interface/resampler.h

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/*
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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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*/
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/*
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* A wrapper for resampling a numerous amount of sampling combinations.
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*/
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#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
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#define WEBRTC_RESAMPLER_RESAMPLER_H_
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#include "typedefs.h"
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namespace webrtc
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{
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enum ResamplerType
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{
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// 4 MSB = Number of channels
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// 4 LSB = Synchronous or asynchronous
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kResamplerSynchronous = 0x10,
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kResamplerAsynchronous = 0x11,
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kResamplerSynchronousStereo = 0x20,
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kResamplerAsynchronousStereo = 0x21,
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kResamplerInvalid = 0xff
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};
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enum ResamplerMode
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{
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kResamplerMode1To1,
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kResamplerMode1To2,
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kResamplerMode1To3,
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kResamplerMode1To4,
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kResamplerMode1To6,
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kResamplerMode2To3,
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kResamplerMode2To11,
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kResamplerMode4To11,
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kResamplerMode8To11,
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kResamplerMode11To16,
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kResamplerMode11To32,
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kResamplerMode2To1,
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kResamplerMode3To1,
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kResamplerMode4To1,
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kResamplerMode6To1,
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kResamplerMode3To2,
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kResamplerMode11To2,
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kResamplerMode11To4,
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kResamplerMode11To8
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};
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class Resampler
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{
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public:
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Resampler();
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Resampler(int inFreq, int outFreq, ResamplerType type);
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~Resampler();
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// Reset all states
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int Reset(int inFreq, int outFreq, ResamplerType type);
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// Reset all states if any parameter has changed
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int ResetIfNeeded(int inFreq, int outFreq, ResamplerType type);
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// Synchronous resampling, all output samples are written to samplesOut
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int Push(const WebRtc_Word16* samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
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int maxLen, int &outLen);
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// Asynchronous resampling, input
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int Insert(WebRtc_Word16* samplesIn, int lengthIn);
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// Asynchronous resampling output, remaining samples are buffered
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int Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen);
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private:
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// Generic pointers since we don't know what states we'll need
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void* state1_;
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void* state2_;
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void* state3_;
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// Storage if needed
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WebRtc_Word16* in_buffer_;
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WebRtc_Word16* out_buffer_;
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int in_buffer_size_;
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int out_buffer_size_;
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int in_buffer_size_max_;
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int out_buffer_size_max_;
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// State
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int my_in_frequency_khz_;
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int my_out_frequency_khz_;
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ResamplerMode my_mode_;
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ResamplerType my_type_;
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// Extra instance for stereo
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Resampler* slave_left_;
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Resampler* slave_right_;
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};
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} // namespace webrtc
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#endif // WEBRTC_RESAMPLER_RESAMPLER_H_

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