diff --git a/README.chromium b/README.chromium deleted file mode 100644 index 246c13dc09..0000000000 --- a/README.chromium +++ /dev/null @@ -1,13 +0,0 @@ -Name: WebRTC -URL: http://www.webrtc.org -Version: 90 -License: BSD -License File: LICENSE - -Description: -WebRTC provides real time voice and video processing -functionality to enable the implementation of -PeerConnection/MediaStream. - -Third party code used in this project is described -in the file LICENSE_THIRD_PARTY. diff --git a/common_settings.gypi b/common_settings.gypi deleted file mode 100644 index 8d6428877a..0000000000 --- a/common_settings.gypi +++ /dev/null @@ -1,21 +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. - -# Placeholder until all gyp files point to build/common.gypi instead. - -{ - 'includes': [ - 'build/common.gypi', - ], -} - -# Local Variables: -# tab-width:2 -# indent-tabs-mode:nil -# End: -# vim: set expandtab tabstop=2 shiftwidth=2: diff --git a/common_types.h b/common_types.h deleted file mode 100644 index 8b0b8a59c1..0000000000 --- a/common_types.h +++ /dev/null @@ -1,595 +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_COMMON_TYPES_H -#define WEBRTC_COMMON_TYPES_H - -#include "typedefs.h" - -#ifdef WEBRTC_EXPORT - #define WEBRTC_DLLEXPORT _declspec(dllexport) -#elif WEBRTC_DLL - #define WEBRTC_DLLEXPORT _declspec(dllimport) -#else - #define WEBRTC_DLLEXPORT -#endif - -#ifndef NULL - #define NULL 0 -#endif - -namespace webrtc { - -class InStream -{ -public: - virtual int Read(void *buf,int len) = 0; - virtual int Rewind() {return -1;} - virtual ~InStream() {} -protected: - InStream() {} -}; - -class OutStream -{ -public: - virtual bool Write(const void *buf,int len) = 0; - virtual int Rewind() {return -1;} - virtual ~OutStream() {} -protected: - OutStream() {} -}; - -enum TraceModule -{ - // not a module, triggered from the engine code - kTraceVoice = 0x0001, - // not a module, triggered from the engine code - kTraceVideo = 0x0002, - // not a module, triggered from the utility code - kTraceUtility = 0x0003, - kTraceRtpRtcp = 0x0004, - kTraceTransport = 0x0005, - kTraceSrtp = 0x0006, - kTraceAudioCoding = 0x0007, - kTraceAudioMixerServer = 0x0008, - kTraceAudioMixerClient = 0x0009, - kTraceFile = 0x000a, - kTraceAudioProcessing = 0x000b, - kTraceVideoCoding = 0x0010, - kTraceVideoMixer = 0x0011, - kTraceAudioDevice = 0x0012, - kTraceVideoRenderer = 0x0014, - kTraceVideoCapture = 0x0015, - kTraceVideoPreocessing = 0x0016 -}; - -enum TraceLevel -{ - kTraceNone = 0x0000, // no trace - kTraceStateInfo = 0x0001, - kTraceWarning = 0x0002, - kTraceError = 0x0004, - kTraceCritical = 0x0008, - kTraceApiCall = 0x0010, - kTraceDefault = 0x00ff, - - kTraceModuleCall = 0x0020, - kTraceMemory = 0x0100, // memory info - kTraceTimer = 0x0200, // timing info - kTraceStream = 0x0400, // "continuous" stream of data - - // used for debug purposes - kTraceDebug = 0x0800, // debug - kTraceInfo = 0x1000, // debug info - - kTraceAll = 0xffff -}; - -// External Trace API -class TraceCallback -{ -public: - virtual void Print(const TraceLevel level, - const char *traceString, - const int length) = 0; -protected: - virtual ~TraceCallback() {} - TraceCallback() {} -}; - - -enum FileFormats -{ - kFileFormatWavFile = 1, - kFileFormatCompressedFile = 2, - kFileFormatAviFile = 3, - kFileFormatPreencodedFile = 4, - kFileFormatPcm16kHzFile = 7, - kFileFormatPcm8kHzFile = 8, - kFileFormatPcm32kHzFile = 9 -}; - - -enum ProcessingTypes -{ - kPlaybackPerChannel = 0, - kPlaybackAllChannelsMixed, - kRecordingPerChannel, - kRecordingAllChannelsMixed -}; - -// Encryption enums -enum CipherTypes -{ - kCipherNull = 0, - kCipherAes128CounterMode = 1 -}; - -enum AuthenticationTypes -{ - kAuthNull = 0, - kAuthHmacSha1 = 3 -}; - -enum SecurityLevels -{ - kNoProtection = 0, - kEncryption = 1, - kAuthentication = 2, - kEncryptionAndAuthentication = 3 -}; - -class Encryption -{ -public: - virtual void encrypt( - int channel_no, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - - virtual void decrypt( - int channel_no, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - - virtual void encrypt_rtcp( - int channel_no, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - - virtual void decrypt_rtcp( - int channel_no, - unsigned char* in_data, - unsigned char* out_data, - int bytes_in, - int* bytes_out) = 0; - -protected: - virtual ~Encryption() {} - Encryption() {} -}; - -// External transport callback interface -class Transport -{ -public: - virtual int SendPacket(int channel, const void *data, int len) = 0; - virtual int SendRTCPPacket(int channel, const void *data, int len) = 0; - -protected: - virtual ~Transport() {} - Transport() {} -}; - -// ================================================================== -// Voice specific types -// ================================================================== - -// Each codec supported can be described by this structure. -struct CodecInst -{ - int pltype; - char plname[32]; - int plfreq; - int pacsize; - int channels; - int rate; -}; - -enum FrameType -{ - kFrameEmpty = 0, - kAudioFrameSpeech = 1, - kAudioFrameCN = 2, - kVideoFrameKey = 3, // independent frame - kVideoFrameDelta = 4, // depends on the previus frame - kVideoFrameGolden = 5, // depends on a old known previus frame - kVideoFrameAltRef = 6 -}; - -// RTP -enum {kRtpCsrcSize = 15}; // RFC 3550 page 13 - -enum RTPDirections -{ - kRtpIncoming = 0, - kRtpOutgoing -}; - -enum PayloadFrequencies -{ - kFreq8000Hz = 8000, - kFreq16000Hz = 16000, - kFreq32000Hz = 32000 -}; - -enum VadModes // degree of bandwidth reduction -{ - kVadConventional = 0, // lowest reduction - kVadAggressiveLow, - kVadAggressiveMid, - kVadAggressiveHigh // highest reduction -}; - -struct NetworkStatistics // NETEQ statistics -{ - // current jitter buffer size in ms - WebRtc_UWord16 currentBufferSize; - // preferred (optimal) buffer size in ms - WebRtc_UWord16 preferredBufferSize; - // loss rate (network + late) in percent (in Q14) - WebRtc_UWord16 currentPacketLossRate; - // late loss rate in percent (in Q14) - WebRtc_UWord16 currentDiscardRate; - // fraction (of original stream) of synthesized speech inserted through - // expansion (in Q14) - WebRtc_UWord16 currentExpandRate; - // fraction of synthesized speech inserted through pre-emptive expansion - // (in Q14) - WebRtc_UWord16 currentPreemptiveRate; - // fraction of data removed through acceleration (in Q14) - WebRtc_UWord16 currentAccelerateRate; -}; - -struct JitterStatistics -{ - // smallest Jitter Buffer size during call in ms - WebRtc_UWord32 jbMinSize; - // largest Jitter Buffer size during call in ms - WebRtc_UWord32 jbMaxSize; - // the average JB size, measured over time - ms - WebRtc_UWord32 jbAvgSize; - // number of times the Jitter Buffer changed (using Accelerate or - // Pre-emptive Expand) - WebRtc_UWord32 jbChangeCount; - // amount (in ms) of audio data received late - WebRtc_UWord32 lateLossMs; - // milliseconds removed to reduce jitter buffer size - WebRtc_UWord32 accelerateMs; - // milliseconds discarded through buffer flushing - WebRtc_UWord32 flushedMs; - // milliseconds of generated silence - WebRtc_UWord32 generatedSilentMs; - // milliseconds of synthetic audio data (non-background noise) - WebRtc_UWord32 interpolatedVoiceMs; - // milliseconds of synthetic audio data (background noise level) - WebRtc_UWord32 interpolatedSilentMs; - // count of tiny expansions in output audio - WebRtc_UWord32 countExpandMoreThan120ms; - // count of small expansions in output audio - WebRtc_UWord32 countExpandMoreThan250ms; - // count of medium expansions in output audio - WebRtc_UWord32 countExpandMoreThan500ms; - // count of long expansions in output audio - WebRtc_UWord32 countExpandMoreThan2000ms; - // duration of longest audio drop-out - WebRtc_UWord32 longestExpandDurationMs; - // count of times we got small network outage (inter-arrival time in - // [500, 1000) ms) - WebRtc_UWord32 countIAT500ms; - // count of times we got medium network outage (inter-arrival time in - // [1000, 2000) ms) - WebRtc_UWord32 countIAT1000ms; - // count of times we got large network outage (inter-arrival time >= - // 2000 ms) - WebRtc_UWord32 countIAT2000ms; - // longest packet inter-arrival time in ms - WebRtc_UWord32 longestIATms; - // min time incoming Packet "waited" to be played - WebRtc_UWord32 minPacketDelayMs; - // max time incoming Packet "waited" to be played - WebRtc_UWord32 maxPacketDelayMs; - // avg time incoming Packet "waited" to be played - WebRtc_UWord32 avgPacketDelayMs; -}; - -typedef struct -{ - int min; // minumum - int max; // maximum - int average; // average -} StatVal; - -typedef struct // All levels are reported in dBm0 -{ - StatVal speech_rx; // long-term speech levels on receiving side - StatVal speech_tx; // long-term speech levels on transmitting side - StatVal noise_rx; // long-term noise/silence levels on receiving side - StatVal noise_tx; // long-term noise/silence levels on transmitting side -} LevelStatistics; - -typedef struct // All levels are reported in dB -{ - StatVal erl; // Echo Return Loss - StatVal erle; // Echo Return Loss Enhancement - StatVal rerl; // RERL = ERL + ERLE - // Echo suppression inside EC at the point just before its NLP - StatVal a_nlp; -} EchoStatistics; - -enum TelephoneEventDetectionMethods -{ - kInBand = 0, - kOutOfBand = 1, - kInAndOutOfBand = 2 -}; - -enum NsModes // type of Noise Suppression -{ - kNsUnchanged = 0, // previously set mode - kNsDefault, // platform default - kNsConference, // conferencing default - kNsLowSuppression, // lowest suppression - kNsModerateSuppression, - kNsHighSuppression, - kNsVeryHighSuppression, // highest suppression -}; - -enum AgcModes // type of Automatic Gain Control -{ - kAgcUnchanged = 0, // previously set mode - kAgcDefault, // platform default - // adaptive mode for use when analog volume control exists (e.g. for - // PC softphone) - kAgcAdaptiveAnalog, - // scaling takes place in the digital domain (e.g. for conference servers - // and embedded devices) - kAgcAdaptiveDigital, - // can be used on embedded devices where the the capture signal is level - // is predictable - kAgcFixedDigital -}; - -// EC modes -enum EcModes // type of Echo Control -{ - kEcUnchanged = 0, // previously set mode - kEcDefault, // platform default - kEcConference, // conferencing default (aggressive AEC) - kEcAec, // Acoustic Echo Cancellation - kEcAecm, // AEC mobile -}; - -// AECM modes -enum AecmModes // mode of AECM -{ - kAecmQuietEarpieceOrHeadset = 0, - // Quiet earpiece or headset use - kAecmEarpiece, // most earpiece use - kAecmLoudEarpiece, // Loud earpiece or quiet speakerphone use - kAecmSpeakerphone, // most speakerphone use (default) - kAecmLoudSpeakerphone // Loud speakerphone -}; - -// AGC configuration -typedef struct -{ - unsigned short targetLeveldBOv; - unsigned short digitalCompressionGaindB; - bool limiterEnable; -} AgcConfig; // AGC configuration parameters - -enum StereoChannel -{ - kStereoLeft = 0, - kStereoRight, - kStereoBoth -}; - -// Audio device layers -enum AudioLayers -{ - kAudioPlatformDefault = 0, - kAudioWindowsWave = 1, - kAudioWindowsCore = 2, - kAudioLinuxAlsa = 3, - kAudioLinuxPulse = 4 -}; - -enum NetEqModes // NetEQ playout configurations -{ - // Optimized trade-off between low delay and jitter robustness for two-way - // communication. - kNetEqDefault = 0, - // Improved jitter robustness at the cost of increased delay. Can be - // used in one-way communication. - kNetEqStreaming = 1, - // Optimzed for decodability of fax signals rather than for perceived audio - // quality. - kNetEqFax = 2, -}; - -enum NetEqBgnModes // NetEQ Background Noise (BGN) configurations -{ - // BGN is always on and will be generated when the incoming RTP stream - // stops (default). - kBgnOn = 0, - // The BGN is faded to zero (complete silence) after a few seconds. - kBgnFade = 1, - // BGN is not used at all. Silence is produced after speech extrapolation - // has faded. - kBgnOff = 2, -}; - -enum OnHoldModes // On Hold direction -{ - kHoldSendAndPlay = 0, // Put both sending and playing in on-hold state. - kHoldSendOnly, // Put only sending in on-hold state. - kHoldPlayOnly // Put only playing in on-hold state. -}; - -enum AmrMode -{ - kRfc3267BwEfficient = 0, - kRfc3267OctetAligned = 1, - kRfc3267FileStorage = 2, -}; - -// ================================================================== -// Video specific types -// ================================================================== - -// Raw video types -enum RawVideoType -{ - kVideoI420 = 0, - kVideoYV12 = 1, - kVideoYUY2 = 2, - kVideoUYVY = 3, - kVideoIYUV = 4, - kVideoARGB = 5, - kVideoRGB24 = 6, - kVideoRGB565 = 7, - kVideoARGB4444 = 8, - kVideoARGB1555 = 9, - kVideoMJPEG = 10, - kVideoNV12 = 11, - kVideoNV21 = 12, - kVideoUnknown = 99 -}; - -// Video codec -enum { kConfigParameterSize = 128}; -enum { kPayloadNameSize = 32}; - -// 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 diff --git a/engine_configurations.h b/engine_configurations.h deleted file mode 100644 index c24e3d2ced..0000000000 --- a/engine_configurations.h +++ /dev/null @@ -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_ diff --git a/typedefs.h b/typedefs.h deleted file mode 100644 index ae71690f18..0000000000 --- a/typedefs.h +++ /dev/null @@ -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 -#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 - - 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 diff --git a/video_engine.gyp b/video_engine.gyp deleted file mode 100644 index 28131f02a2..0000000000 --- a/video_engine.gyp +++ /dev/null @@ -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: diff --git a/voice_engine.gyp b/voice_engine.gyp deleted file mode 100644 index 96a0527eea..0000000000 --- a/voice_engine.gyp +++ /dev/null @@ -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: