git-svn-id: http://webrtc.googlecode.com/svn/trunk@167 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
parent
5adc73aad3
commit
5740135c6d
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Name: WebRTC
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URL: http://www.webrtc.org
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Version: 90
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License: BSD
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License File: LICENSE
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Description:
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WebRTC provides real time voice and video processing
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functionality to enable the implementation of
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PeerConnection/MediaStream.
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Third party code used in this project is described
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in the file LICENSE_THIRD_PARTY.
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@ -1,21 +0,0 @@
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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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# Placeholder until all gyp files point to build/common.gypi instead.
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{
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'includes': [
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'build/common.gypi',
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],
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}
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# Local Variables:
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# tab-width:2
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# indent-tabs-mode:nil
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# End:
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# vim: set expandtab tabstop=2 shiftwidth=2:
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595
common_types.h
595
common_types.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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#ifndef WEBRTC_COMMON_TYPES_H
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#define WEBRTC_COMMON_TYPES_H
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#include "typedefs.h"
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#ifdef WEBRTC_EXPORT
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#define WEBRTC_DLLEXPORT _declspec(dllexport)
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#elif WEBRTC_DLL
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#define WEBRTC_DLLEXPORT _declspec(dllimport)
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#else
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#define WEBRTC_DLLEXPORT
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#endif
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#ifndef NULL
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#define NULL 0
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#endif
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namespace webrtc {
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class InStream
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{
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public:
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virtual int Read(void *buf,int len) = 0;
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virtual int Rewind() {return -1;}
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virtual ~InStream() {}
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protected:
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InStream() {}
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};
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class OutStream
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{
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public:
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virtual bool Write(const void *buf,int len) = 0;
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virtual int Rewind() {return -1;}
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virtual ~OutStream() {}
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protected:
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OutStream() {}
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};
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enum TraceModule
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{
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// not a module, triggered from the engine code
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kTraceVoice = 0x0001,
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// not a module, triggered from the engine code
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kTraceVideo = 0x0002,
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// not a module, triggered from the utility code
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kTraceUtility = 0x0003,
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kTraceRtpRtcp = 0x0004,
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kTraceTransport = 0x0005,
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kTraceSrtp = 0x0006,
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kTraceAudioCoding = 0x0007,
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kTraceAudioMixerServer = 0x0008,
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kTraceAudioMixerClient = 0x0009,
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kTraceFile = 0x000a,
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kTraceAudioProcessing = 0x000b,
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kTraceVideoCoding = 0x0010,
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kTraceVideoMixer = 0x0011,
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kTraceAudioDevice = 0x0012,
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kTraceVideoRenderer = 0x0014,
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kTraceVideoCapture = 0x0015,
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kTraceVideoPreocessing = 0x0016
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};
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enum TraceLevel
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{
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kTraceNone = 0x0000, // no trace
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kTraceStateInfo = 0x0001,
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kTraceWarning = 0x0002,
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kTraceError = 0x0004,
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kTraceCritical = 0x0008,
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kTraceApiCall = 0x0010,
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kTraceDefault = 0x00ff,
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kTraceModuleCall = 0x0020,
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kTraceMemory = 0x0100, // memory info
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kTraceTimer = 0x0200, // timing info
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kTraceStream = 0x0400, // "continuous" stream of data
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// used for debug purposes
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kTraceDebug = 0x0800, // debug
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kTraceInfo = 0x1000, // debug info
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kTraceAll = 0xffff
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};
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// External Trace API
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class TraceCallback
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{
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public:
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virtual void Print(const TraceLevel level,
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const char *traceString,
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const int length) = 0;
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protected:
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virtual ~TraceCallback() {}
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TraceCallback() {}
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};
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enum FileFormats
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{
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kFileFormatWavFile = 1,
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kFileFormatCompressedFile = 2,
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kFileFormatAviFile = 3,
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kFileFormatPreencodedFile = 4,
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kFileFormatPcm16kHzFile = 7,
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kFileFormatPcm8kHzFile = 8,
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kFileFormatPcm32kHzFile = 9
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};
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enum ProcessingTypes
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{
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kPlaybackPerChannel = 0,
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kPlaybackAllChannelsMixed,
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kRecordingPerChannel,
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kRecordingAllChannelsMixed
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};
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// Encryption enums
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enum CipherTypes
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{
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kCipherNull = 0,
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kCipherAes128CounterMode = 1
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};
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enum AuthenticationTypes
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{
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kAuthNull = 0,
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kAuthHmacSha1 = 3
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};
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enum SecurityLevels
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{
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kNoProtection = 0,
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kEncryption = 1,
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kAuthentication = 2,
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kEncryptionAndAuthentication = 3
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};
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class Encryption
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{
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public:
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virtual void encrypt(
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int channel_no,
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unsigned char* in_data,
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unsigned char* out_data,
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int bytes_in,
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int* bytes_out) = 0;
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virtual void decrypt(
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int channel_no,
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unsigned char* in_data,
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unsigned char* out_data,
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int bytes_in,
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int* bytes_out) = 0;
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virtual void encrypt_rtcp(
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int channel_no,
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unsigned char* in_data,
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unsigned char* out_data,
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int bytes_in,
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int* bytes_out) = 0;
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virtual void decrypt_rtcp(
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int channel_no,
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unsigned char* in_data,
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unsigned char* out_data,
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int bytes_in,
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int* bytes_out) = 0;
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protected:
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virtual ~Encryption() {}
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Encryption() {}
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};
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// External transport callback interface
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class Transport
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{
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public:
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virtual int SendPacket(int channel, const void *data, int len) = 0;
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virtual int SendRTCPPacket(int channel, const void *data, int len) = 0;
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protected:
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virtual ~Transport() {}
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Transport() {}
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};
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// ==================================================================
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// Voice specific types
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// ==================================================================
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// Each codec supported can be described by this structure.
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struct CodecInst
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{
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int pltype;
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char plname[32];
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int plfreq;
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int pacsize;
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int channels;
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int rate;
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};
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enum FrameType
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{
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kFrameEmpty = 0,
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kAudioFrameSpeech = 1,
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kAudioFrameCN = 2,
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kVideoFrameKey = 3, // independent frame
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kVideoFrameDelta = 4, // depends on the previus frame
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kVideoFrameGolden = 5, // depends on a old known previus frame
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kVideoFrameAltRef = 6
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};
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// RTP
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enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
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enum RTPDirections
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{
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kRtpIncoming = 0,
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kRtpOutgoing
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};
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enum PayloadFrequencies
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{
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kFreq8000Hz = 8000,
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kFreq16000Hz = 16000,
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kFreq32000Hz = 32000
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};
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enum VadModes // degree of bandwidth reduction
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{
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kVadConventional = 0, // lowest reduction
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kVadAggressiveLow,
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kVadAggressiveMid,
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kVadAggressiveHigh // highest reduction
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};
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struct NetworkStatistics // NETEQ statistics
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{
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// current jitter buffer size in ms
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WebRtc_UWord16 currentBufferSize;
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// preferred (optimal) buffer size in ms
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WebRtc_UWord16 preferredBufferSize;
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// loss rate (network + late) in percent (in Q14)
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WebRtc_UWord16 currentPacketLossRate;
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// late loss rate in percent (in Q14)
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WebRtc_UWord16 currentDiscardRate;
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// fraction (of original stream) of synthesized speech inserted through
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// expansion (in Q14)
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WebRtc_UWord16 currentExpandRate;
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// fraction of synthesized speech inserted through pre-emptive expansion
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// (in Q14)
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WebRtc_UWord16 currentPreemptiveRate;
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// fraction of data removed through acceleration (in Q14)
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WebRtc_UWord16 currentAccelerateRate;
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};
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struct JitterStatistics
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{
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// smallest Jitter Buffer size during call in ms
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WebRtc_UWord32 jbMinSize;
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// largest Jitter Buffer size during call in ms
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WebRtc_UWord32 jbMaxSize;
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// the average JB size, measured over time - ms
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WebRtc_UWord32 jbAvgSize;
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// number of times the Jitter Buffer changed (using Accelerate or
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// Pre-emptive Expand)
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WebRtc_UWord32 jbChangeCount;
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// amount (in ms) of audio data received late
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WebRtc_UWord32 lateLossMs;
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// milliseconds removed to reduce jitter buffer size
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WebRtc_UWord32 accelerateMs;
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// milliseconds discarded through buffer flushing
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WebRtc_UWord32 flushedMs;
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// milliseconds of generated silence
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WebRtc_UWord32 generatedSilentMs;
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// milliseconds of synthetic audio data (non-background noise)
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WebRtc_UWord32 interpolatedVoiceMs;
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// milliseconds of synthetic audio data (background noise level)
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WebRtc_UWord32 interpolatedSilentMs;
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// count of tiny expansions in output audio
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WebRtc_UWord32 countExpandMoreThan120ms;
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// count of small expansions in output audio
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WebRtc_UWord32 countExpandMoreThan250ms;
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// count of medium expansions in output audio
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WebRtc_UWord32 countExpandMoreThan500ms;
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// count of long expansions in output audio
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WebRtc_UWord32 countExpandMoreThan2000ms;
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// duration of longest audio drop-out
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WebRtc_UWord32 longestExpandDurationMs;
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// count of times we got small network outage (inter-arrival time in
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// [500, 1000) ms)
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WebRtc_UWord32 countIAT500ms;
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// count of times we got medium network outage (inter-arrival time in
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// [1000, 2000) ms)
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WebRtc_UWord32 countIAT1000ms;
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// count of times we got large network outage (inter-arrival time >=
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// 2000 ms)
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WebRtc_UWord32 countIAT2000ms;
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// longest packet inter-arrival time in ms
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WebRtc_UWord32 longestIATms;
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// min time incoming Packet "waited" to be played
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WebRtc_UWord32 minPacketDelayMs;
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// max time incoming Packet "waited" to be played
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WebRtc_UWord32 maxPacketDelayMs;
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// avg time incoming Packet "waited" to be played
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WebRtc_UWord32 avgPacketDelayMs;
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};
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typedef struct
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{
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int min; // minumum
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int max; // maximum
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int average; // average
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} StatVal;
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typedef struct // All levels are reported in dBm0
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{
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StatVal speech_rx; // long-term speech levels on receiving side
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StatVal speech_tx; // long-term speech levels on transmitting side
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StatVal noise_rx; // long-term noise/silence levels on receiving side
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StatVal noise_tx; // long-term noise/silence levels on transmitting side
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} LevelStatistics;
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typedef struct // All levels are reported in dB
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{
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StatVal erl; // Echo Return Loss
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StatVal erle; // Echo Return Loss Enhancement
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StatVal rerl; // RERL = ERL + ERLE
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// Echo suppression inside EC at the point just before its NLP
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StatVal a_nlp;
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} EchoStatistics;
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enum TelephoneEventDetectionMethods
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{
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kInBand = 0,
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kOutOfBand = 1,
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kInAndOutOfBand = 2
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};
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enum NsModes // type of Noise Suppression
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{
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kNsUnchanged = 0, // previously set mode
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kNsDefault, // platform default
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kNsConference, // conferencing default
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kNsLowSuppression, // lowest suppression
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kNsModerateSuppression,
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kNsHighSuppression,
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kNsVeryHighSuppression, // highest suppression
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};
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enum AgcModes // type of Automatic Gain Control
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{
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kAgcUnchanged = 0, // previously set mode
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kAgcDefault, // platform default
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// adaptive mode for use when analog volume control exists (e.g. for
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// PC softphone)
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kAgcAdaptiveAnalog,
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// scaling takes place in the digital domain (e.g. for conference servers
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// and embedded devices)
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kAgcAdaptiveDigital,
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// can be used on embedded devices where the the capture signal is level
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// is predictable
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kAgcFixedDigital
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};
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// EC modes
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enum EcModes // type of Echo Control
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{
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kEcUnchanged = 0, // previously set mode
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kEcDefault, // platform default
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kEcConference, // conferencing default (aggressive AEC)
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kEcAec, // Acoustic Echo Cancellation
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kEcAecm, // AEC mobile
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};
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// AECM modes
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enum AecmModes // mode of AECM
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{
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kAecmQuietEarpieceOrHeadset = 0,
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// Quiet earpiece or headset use
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kAecmEarpiece, // most earpiece use
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kAecmLoudEarpiece, // Loud earpiece or quiet speakerphone use
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kAecmSpeakerphone, // most speakerphone use (default)
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kAecmLoudSpeakerphone // Loud speakerphone
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};
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// AGC configuration
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typedef struct
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{
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unsigned short targetLeveldBOv;
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unsigned short digitalCompressionGaindB;
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bool limiterEnable;
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} AgcConfig; // AGC configuration parameters
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||||
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enum StereoChannel
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{
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kStereoLeft = 0,
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kStereoRight,
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kStereoBoth
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};
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||||
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// Audio device layers
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enum AudioLayers
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{
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kAudioPlatformDefault = 0,
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kAudioWindowsWave = 1,
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kAudioWindowsCore = 2,
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kAudioLinuxAlsa = 3,
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kAudioLinuxPulse = 4
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};
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||||
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enum NetEqModes // NetEQ playout configurations
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||||
{
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||||
// Optimized trade-off between low delay and jitter robustness for two-way
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// communication.
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||||
kNetEqDefault = 0,
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||||
// Improved jitter robustness at the cost of increased delay. Can be
|
||||
// used in one-way communication.
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kNetEqStreaming = 1,
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// Optimzed for decodability of fax signals rather than for perceived audio
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// quality.
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||||
kNetEqFax = 2,
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||||
};
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||||
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||||
enum NetEqBgnModes // NetEQ Background Noise (BGN) configurations
|
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{
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// BGN is always on and will be generated when the incoming RTP stream
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// stops (default).
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kBgnOn = 0,
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// The BGN is faded to zero (complete silence) after a few seconds.
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kBgnFade = 1,
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// BGN is not used at all. Silence is produced after speech extrapolation
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// has faded.
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kBgnOff = 2,
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};
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enum OnHoldModes // On Hold direction
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||||
{
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kHoldSendAndPlay = 0, // Put both sending and playing in on-hold state.
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kHoldSendOnly, // Put only sending in on-hold state.
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kHoldPlayOnly // Put only playing in on-hold state.
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};
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||||
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||||
enum AmrMode
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{
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kRfc3267BwEfficient = 0,
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kRfc3267OctetAligned = 1,
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||||
kRfc3267FileStorage = 2,
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};
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||||
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// ==================================================================
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||||
// Video specific types
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||||
// ==================================================================
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||||
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||||
// Raw video types
|
||||
enum RawVideoType
|
||||
{
|
||||
kVideoI420 = 0,
|
||||
kVideoYV12 = 1,
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||||
kVideoYUY2 = 2,
|
||||
kVideoUYVY = 3,
|
||||
kVideoIYUV = 4,
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||||
kVideoARGB = 5,
|
||||
kVideoRGB24 = 6,
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kVideoRGB565 = 7,
|
||||
kVideoARGB4444 = 8,
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||||
kVideoARGB1555 = 9,
|
||||
kVideoMJPEG = 10,
|
||||
kVideoNV12 = 11,
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||||
kVideoNV21 = 12,
|
||||
kVideoUnknown = 99
|
||||
};
|
||||
|
||||
// Video codec
|
||||
enum { kConfigParameterSize = 128};
|
||||
enum { kPayloadNameSize = 32};
|
||||
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||||
// H.263 specific
|
||||
struct VideoCodecH263
|
||||
{
|
||||
char quality;
|
||||
};
|
||||
|
||||
// H.264 specific
|
||||
enum H264Packetization
|
||||
{
|
||||
kH264SingleMode = 0,
|
||||
kH264NonInterleavedMode = 1
|
||||
};
|
||||
|
||||
enum VideoCodecComplexity
|
||||
{
|
||||
kComplexityNormal = 0,
|
||||
kComplexityHigh = 1,
|
||||
kComplexityHigher = 2,
|
||||
kComplexityMax = 3
|
||||
};
|
||||
|
||||
enum VideoCodecProfile
|
||||
{
|
||||
kProfileBase = 0x00,
|
||||
kProfileMain = 0x01
|
||||
};
|
||||
|
||||
struct VideoCodecH264
|
||||
{
|
||||
H264Packetization packetization;
|
||||
VideoCodecComplexity complexity;
|
||||
VideoCodecProfile profile;
|
||||
char level;
|
||||
char quality;
|
||||
|
||||
bool useFMO;
|
||||
|
||||
unsigned char configParameters[kConfigParameterSize];
|
||||
unsigned char configParametersSize;
|
||||
};
|
||||
|
||||
// VP8 specific
|
||||
struct VideoCodecVP8
|
||||
{
|
||||
bool pictureLossIndicationOn;
|
||||
bool feedbackModeOn;
|
||||
VideoCodecComplexity complexity;
|
||||
};
|
||||
|
||||
// MPEG-4 specific
|
||||
struct VideoCodecMPEG4
|
||||
{
|
||||
unsigned char configParameters[kConfigParameterSize];
|
||||
unsigned char configParametersSize;
|
||||
char level;
|
||||
};
|
||||
|
||||
// Unknown specific
|
||||
struct VideoCodecGeneric
|
||||
{
|
||||
};
|
||||
|
||||
// Video codec types
|
||||
enum VideoCodecType
|
||||
{
|
||||
kVideoCodecH263,
|
||||
kVideoCodecH264,
|
||||
kVideoCodecVP8,
|
||||
kVideoCodecMPEG4,
|
||||
kVideoCodecI420,
|
||||
kVideoCodecRED,
|
||||
kVideoCodecULPFEC,
|
||||
kVideoCodecUnknown
|
||||
};
|
||||
|
||||
union VideoCodecUnion
|
||||
{
|
||||
VideoCodecH263 H263;
|
||||
VideoCodecH264 H264;
|
||||
VideoCodecVP8 VP8;
|
||||
VideoCodecMPEG4 MPEG4;
|
||||
VideoCodecGeneric Generic;
|
||||
};
|
||||
|
||||
// Common video codec properties
|
||||
struct VideoCodec
|
||||
{
|
||||
VideoCodecType codecType;
|
||||
char plName[kPayloadNameSize];
|
||||
unsigned char plType;
|
||||
|
||||
unsigned short width;
|
||||
unsigned short height;
|
||||
|
||||
unsigned int startBitrate;
|
||||
unsigned int maxBitrate;
|
||||
unsigned int minBitrate;
|
||||
unsigned char maxFramerate;
|
||||
|
||||
VideoCodecUnion codecSpecific;
|
||||
|
||||
unsigned int qpMax;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_COMMON_TYPES_H
|
||||
@ -1,131 +0,0 @@
|
||||
/*
|
||||
* Copyright (c) 2011 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_ENGINE_CONFIGURATIONS_H_
|
||||
#define WEBRTC_ENGINE_CONFIGURATIONS_H_
|
||||
|
||||
// ============================================================================
|
||||
// Voice and Video
|
||||
// ============================================================================
|
||||
|
||||
// #define WEBRTC_EXTERNAL_TRANSPORT
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// [Voice] Codec settings
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#define WEBRTC_CODEC_ILBC
|
||||
#define WEBRTC_CODEC_ISAC // floating-point iSAC implementation (default)
|
||||
// #define WEBRTC_CODEC_ISACFX // fix-point iSAC implementation
|
||||
#define WEBRTC_CODEC_G722
|
||||
#define WEBRTC_CODEC_PCM16
|
||||
#define WEBRTC_CODEC_RED
|
||||
#define WEBRTC_CODEC_AVT
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// [Video] Codec settings
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#define VIDEOCODEC_I420
|
||||
#define VIDEOCODEC_VP8
|
||||
|
||||
// ============================================================================
|
||||
// VoiceEngine
|
||||
// ============================================================================
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// Settings for VoiceEngine
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#define WEBRTC_VOICE_ENGINE_AGC // Near-end AGC
|
||||
#define WEBRTC_VOICE_ENGINE_ECHO // Near-end AEC
|
||||
#define WEBRTC_VOICE_ENGINE_NR // Near-end NS
|
||||
#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION
|
||||
#define WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// VoiceEngine sub-APIs
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#define WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
|
||||
#define WEBRTC_VOICE_ENGINE_CALL_REPORT_API
|
||||
#define WEBRTC_VOICE_ENGINE_CODEC_API
|
||||
#define WEBRTC_VOICE_ENGINE_DTMF_API
|
||||
#define WEBRTC_VOICE_ENGINE_ENCRYPTION_API
|
||||
#define WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
|
||||
#define WEBRTC_VOICE_ENGINE_FILE_API
|
||||
#define WEBRTC_VOICE_ENGINE_HARDWARE_API
|
||||
#define WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
|
||||
#define WEBRTC_VOICE_ENGINE_NETWORK_API
|
||||
#define WEBRTC_VOICE_ENGINE_RTP_RTCP_API
|
||||
#define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
|
||||
#define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
|
||||
|
||||
// ============================================================================
|
||||
// VideoEngine
|
||||
// ============================================================================
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// Settings for special VideoEngine configurations
|
||||
// ----------------------------------------------------------------------------
|
||||
// ----------------------------------------------------------------------------
|
||||
// VideoEngine sub-API:s
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#define WEBRTC_VIDEO_ENGINE_CAPTURE_API
|
||||
#define WEBRTC_VIDEO_ENGINE_CODEC_API
|
||||
#define WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
|
||||
#define WEBRTC_VIDEO_ENGINE_FILE_API
|
||||
#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
|
||||
#define WEBRTC_VIDEO_ENGINE_NETWORK_API
|
||||
#define WEBRTC_VIDEO_ENGINE_RENDER_API
|
||||
#define WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
|
||||
// #define WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
|
||||
|
||||
// ============================================================================
|
||||
// Platform specific configurations
|
||||
// ============================================================================
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// VideoEngine Windows
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#if defined(_WIN32)
|
||||
// #define DIRECTDRAW_RENDERING
|
||||
#define DIRECT3D9_RENDERING // Requires DirectX 9.
|
||||
#endif
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// VideoEngine MAC
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#if defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
|
||||
// #define CARBON_RENDERING
|
||||
#define COCOA_RENDERING
|
||||
#endif
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// VideoEngine Mobile iPhone
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
#if defined(MAC_IPHONE)
|
||||
#define EAGL_RENDERING
|
||||
#endif
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// Deprecated
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
// #define WEBRTC_CODEC_G729
|
||||
// #define WEBRTC_DTMF_DETECTION
|
||||
// #define WEBRTC_SRTP
|
||||
// #define WEBRTC_SRTP_ALLOW_ROC_ITERATION
|
||||
|
||||
#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_
|
||||
107
typedefs.h
107
typedefs.h
@ -1,107 +0,0 @@
|
||||
/*
|
||||
* Copyright (c) 2011 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.
|
||||
*/
|
||||
|
||||
/*
|
||||
*
|
||||
* This file contains type definitions used in all WebRtc APIs.
|
||||
*
|
||||
*/
|
||||
|
||||
/* Reserved words definitions */
|
||||
#define WEBRTC_EXTERN extern
|
||||
#define G_CONST const
|
||||
#define WEBRTC_INLINE extern __inline
|
||||
|
||||
#ifndef WEBRTC_TYPEDEFS_H
|
||||
#define WEBRTC_TYPEDEFS_H
|
||||
|
||||
/* Define WebRtc preprocessor identifiers based on the current build platform */
|
||||
#if defined(WIN32)
|
||||
// Windows & Windows Mobile
|
||||
#if !defined(WEBRTC_TARGET_PC)
|
||||
#define WEBRTC_TARGET_PC
|
||||
#endif
|
||||
#elif defined(__APPLE__)
|
||||
// Mac OS X
|
||||
#if defined(__LITTLE_ENDIAN__ ) //TODO: is this used?
|
||||
#if !defined(WEBRTC_TARGET_MAC_INTEL)
|
||||
#define WEBRTC_TARGET_MAC_INTEL
|
||||
#endif
|
||||
#else
|
||||
#if !defined(WEBRTC_TARGET_MAC)
|
||||
#define WEBRTC_TARGET_MAC
|
||||
#endif
|
||||
#endif
|
||||
#else
|
||||
// Linux etc.
|
||||
#if !defined(WEBRTC_TARGET_PC)
|
||||
#define WEBRTC_TARGET_PC
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#if defined(WEBRTC_TARGET_PC)
|
||||
|
||||
#if !defined(_MSC_VER)
|
||||
#include <stdint.h>
|
||||
#else
|
||||
// Define C99 equivalent types.
|
||||
// Since MSVC doesn't include these headers, we have to write our own
|
||||
// version to provide a compatibility layer between MSVC and the WebRTC
|
||||
// headers.
|
||||
typedef signed char int8_t;
|
||||
typedef signed short int16_t;
|
||||
typedef signed int int32_t;
|
||||
typedef signed long long int64_t;
|
||||
typedef unsigned char uint8_t;
|
||||
typedef unsigned short uint16_t;
|
||||
typedef unsigned int uint32_t;
|
||||
typedef unsigned long long uint64_t;
|
||||
#endif
|
||||
|
||||
#if defined(WIN32)
|
||||
typedef __int64 WebRtc_Word64;
|
||||
typedef unsigned __int64 WebRtc_UWord64;
|
||||
#else
|
||||
typedef int64_t WebRtc_Word64;
|
||||
typedef uint64_t WebRtc_UWord64;
|
||||
#endif
|
||||
typedef int32_t WebRtc_Word32;
|
||||
typedef uint32_t WebRtc_UWord32;
|
||||
typedef int16_t WebRtc_Word16;
|
||||
typedef uint16_t WebRtc_UWord16;
|
||||
typedef char WebRtc_Word8;
|
||||
typedef uint8_t WebRtc_UWord8;
|
||||
|
||||
/* Define endian for the platform */
|
||||
#define WEBRTC_LITTLE_ENDIAN
|
||||
|
||||
#elif defined(WEBRTC_TARGET_MAC_INTEL)
|
||||
#include <stdint.h>
|
||||
|
||||
typedef int64_t WebRtc_Word64;
|
||||
typedef uint64_t WebRtc_UWord64;
|
||||
typedef int32_t WebRtc_Word32;
|
||||
typedef uint32_t WebRtc_UWord32;
|
||||
typedef int16_t WebRtc_Word16;
|
||||
typedef char WebRtc_Word8;
|
||||
typedef uint16_t WebRtc_UWord16;
|
||||
typedef uint8_t WebRtc_UWord8;
|
||||
|
||||
/* Define endian for the platform */
|
||||
#define WEBRTC_LITTLE_ENDIAN
|
||||
|
||||
#else
|
||||
|
||||
#error "No platform defined for WebRtc type definitions (webrtc_typedefs.h)"
|
||||
|
||||
#endif
|
||||
|
||||
|
||||
#endif // WEBRTC_TYPEDEFS_H
|
||||
@ -1,17 +0,0 @@
|
||||
# Copyright (c) 2009 The Chromium Authors. All rights reserved.
|
||||
# Use of this source code is governed by a BSD-style license that can be
|
||||
# found in the LICENSE file.
|
||||
|
||||
{
|
||||
'includes': [
|
||||
'common_settings.gypi', # Common settings
|
||||
# Defines target vie_auto_test
|
||||
'video_engine/main/test/AutoTest/vie_auto_test.gypi',
|
||||
],
|
||||
}
|
||||
|
||||
# Local Variables:
|
||||
# tab-width:2
|
||||
# indent-tabs-mode:nil
|
||||
# End:
|
||||
# vim: set expandtab tabstop=2 shiftwidth=2:
|
||||
163
voice_engine.gyp
163
voice_engine.gyp
@ -1,163 +0,0 @@
|
||||
# Copyright (c) 2011 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.
|
||||
|
||||
{
|
||||
'includes': [
|
||||
'common_settings.gypi',
|
||||
],
|
||||
'targets': [
|
||||
# Auto test - command line test for all platforms
|
||||
{
|
||||
'target_name': 'voe_auto_test',
|
||||
'type': 'executable',
|
||||
'dependencies': [
|
||||
'voice_engine/main/source/voice_engine_core.gyp:voice_engine_core',
|
||||
'system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
||||
],
|
||||
'include_dirs': [
|
||||
'voice_engine/main/test/auto_test',
|
||||
],
|
||||
'sources': [
|
||||
'voice_engine/main/test/auto_test/voe_cpu_test.cc',
|
||||
'voice_engine/main/test/auto_test/voe_cpu_test.h',
|
||||
'voice_engine/main/test/auto_test/voe_extended_test.cc',
|
||||
'voice_engine/main/test/auto_test/voe_extended_test.h',
|
||||
'voice_engine/main/test/auto_test/voe_standard_test.cc',
|
||||
'voice_engine/main/test/auto_test/voe_standard_test.h',
|
||||
'voice_engine/main/test/auto_test/voe_stress_test.cc',
|
||||
'voice_engine/main/test/auto_test/voe_stress_test.h',
|
||||
'voice_engine/main/test/auto_test/voe_test_defines.h',
|
||||
'voice_engine/main/test/auto_test/voe_test_interface.h',
|
||||
'voice_engine/main/test/auto_test/voe_unit_test.cc',
|
||||
'voice_engine/main/test/auto_test/voe_unit_test.h',
|
||||
],
|
||||
'conditions': [
|
||||
['OS=="linux" or OS=="mac"', {
|
||||
'actions': [
|
||||
{
|
||||
'action_name': 'copy audio file',
|
||||
'inputs': [
|
||||
'test/data/voice_engine/audio_long16.pcm',
|
||||
],
|
||||
'outputs': [
|
||||
'/tmp/audio_long16.pcm',
|
||||
],
|
||||
'action': [
|
||||
'/bin/sh', '-c',
|
||||
'cp -f test/data/voice_engine/audio_* /tmp/;'\
|
||||
],
|
||||
},
|
||||
],
|
||||
}],
|
||||
['OS=="win"', {
|
||||
'dependencies': [
|
||||
'voice_engine.gyp:voe_ui_win_test',
|
||||
],
|
||||
}],
|
||||
['OS=="win"', {
|
||||
'actions': [
|
||||
{
|
||||
'action_name': 'copy audio file',
|
||||
'inputs': [
|
||||
'test/data/voice_engine/audio_long16.pcm',
|
||||
],
|
||||
'outputs': [
|
||||
'/tmp/audio_long16.pcm',
|
||||
],
|
||||
'action': [
|
||||
'cmd', '/c',
|
||||
'xcopy /Y /R .\\test\\data\\voice_engine\\audio_* \\tmp',
|
||||
],
|
||||
},
|
||||
],
|
||||
}],
|
||||
],
|
||||
},
|
||||
{
|
||||
# command line test that should work on linux/mac/win
|
||||
'target_name': 'voe_cmd_test',
|
||||
'type': 'executable',
|
||||
'dependencies': [
|
||||
'voice_engine/main/source/voice_engine_core.gyp:voice_engine_core',
|
||||
'system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
||||
],
|
||||
'sources': [
|
||||
'voice_engine/main/test/cmd_test/voe_cmd_test.cc',
|
||||
],
|
||||
},
|
||||
],
|
||||
'conditions': [
|
||||
['OS=="win"', {
|
||||
'targets': [
|
||||
# WinTest - GUI test for Windows
|
||||
{
|
||||
'target_name': 'voe_ui_win_test',
|
||||
'type': 'executable',
|
||||
'dependencies': [
|
||||
'voice_engine/main/source/voice_engine_core.gyp:voice_engine_core',
|
||||
'system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
||||
],
|
||||
'include_dirs': [
|
||||
'voice_engine/main/test/win_test',
|
||||
],
|
||||
'sources': [
|
||||
'voice_engine/main/test/win_test/Resource.h',
|
||||
'voice_engine/main/test/win_test/WinTest.cpp',
|
||||
'voice_engine/main/test/win_test/WinTest.h',
|
||||
'voice_engine/main/test/win_test/WinTest.rc',
|
||||
'voice_engine/main/test/win_test/WinTestDlg.cpp',
|
||||
'voice_engine/main/test/win_test/WinTestDlg.h',
|
||||
'voice_engine/main/test/win_test/res/WinTest.ico',
|
||||
'voice_engine/main/test/win_test/res/WinTest.rc2',
|
||||
'voice_engine/main/test/win_test/stdafx.cpp',
|
||||
'voice_engine/main/test/win_test/stdafx.h',
|
||||
],
|
||||
'actions': [
|
||||
{
|
||||
'action_name': 'copy audio file',
|
||||
'inputs': [
|
||||
'test/data/voice_engine/audio_tiny11.wav',
|
||||
],
|
||||
'outputs': [
|
||||
'/tmp/audio_tiny11.wav',
|
||||
],
|
||||
'action': [
|
||||
'cmd', '/c',
|
||||
'xcopy /Y /R .\\test\\data\\voice_engine\\audio_* \\tmp',
|
||||
],
|
||||
},
|
||||
],
|
||||
'configurations': {
|
||||
'Common_Base': {
|
||||
'msvs_configuration_attributes': {
|
||||
'conditions': [
|
||||
['component=="shared_library"', {
|
||||
'UseOfMFC': '2', # Shared DLL
|
||||
},{
|
||||
'UseOfMFC': '1', # Static
|
||||
}],
|
||||
],
|
||||
},
|
||||
},
|
||||
},
|
||||
'msvs_settings': {
|
||||
'VCLinkerTool': {
|
||||
'SubSystem': '2', # Windows
|
||||
},
|
||||
},
|
||||
},
|
||||
],
|
||||
}],
|
||||
],
|
||||
}
|
||||
|
||||
# Local Variables:
|
||||
# tab-width:2
|
||||
# indent-tabs-mode:nil
|
||||
# End:
|
||||
# vim: set expandtab tabstop=2 shiftwidth=2:
|
||||
Loading…
x
Reference in New Issue
Block a user