| /* |
| * Copyright (c) 2012 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 common constants for VoiceEngine, as well as |
| * platform specific settings and include files. |
| */ |
| |
| #ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H |
| #define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H |
| |
| #include "webrtc/common_types.h" |
| #include "webrtc/engine_configurations.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc { |
| |
| // Internal buffer size required for mono audio, based on the highest sample |
| // rate voice engine supports (10 ms of audio at 192 kHz). |
| static const size_t kMaxMonoDataSizeSamples = 1920; |
| |
| // VolumeControl |
| enum { kMinVolumeLevel = 0 }; |
| enum { kMaxVolumeLevel = 255 }; |
| // Min scale factor for per-channel volume scaling |
| const float kMinOutputVolumeScaling = 0.0f; |
| // Max scale factor for per-channel volume scaling |
| const float kMaxOutputVolumeScaling = 10.0f; |
| // Min scale factor for output volume panning |
| const float kMinOutputVolumePanning = 0.0f; |
| // Max scale factor for output volume panning |
| const float kMaxOutputVolumePanning = 1.0f; |
| |
| // DTMF |
| enum { kMinDtmfEventCode = 0 }; // DTMF digit "0" |
| enum { kMaxDtmfEventCode = 15 }; // DTMF digit "D" |
| enum { kMinTelephoneEventCode = 0 }; // RFC4733 (Section 2.3.1) |
| enum { kMaxTelephoneEventCode = 255 }; // RFC4733 (Section 2.3.1) |
| enum { kMinTelephoneEventDuration = 100 }; |
| enum { kMaxTelephoneEventDuration = 60000 }; // Actual limit is 2^16 |
| enum { kMinTelephoneEventAttenuation = 0 }; // 0 dBm0 |
| enum { kMaxTelephoneEventAttenuation = 36 }; // -36 dBm0 |
| enum { kMinTelephoneEventSeparationMs = 100 }; // Min delta time between two |
| // telephone events |
| enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 }; // assumes Ethernet |
| |
| enum { kVoiceEngineMaxModuleVersionSize = 960 }; |
| |
| // Audio processing |
| const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate; |
| const GainControl::Mode kDefaultAgcMode = |
| #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) |
| GainControl::kAdaptiveDigital; |
| #else |
| GainControl::kAdaptiveAnalog; |
| #endif |
| const bool kDefaultAgcState = |
| #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) |
| false; |
| #else |
| true; |
| #endif |
| const GainControl::Mode kDefaultRxAgcMode = GainControl::kAdaptiveDigital; |
| |
| // Codec |
| // Min init target rate for iSAC-wb |
| enum { kVoiceEngineMinIsacInitTargetRateBpsWb = 10000 }; |
| // Max init target rate for iSAC-wb |
| enum { kVoiceEngineMaxIsacInitTargetRateBpsWb = 32000 }; |
| // Min init target rate for iSAC-swb |
| enum { kVoiceEngineMinIsacInitTargetRateBpsSwb = 10000 }; |
| // Max init target rate for iSAC-swb |
| enum { kVoiceEngineMaxIsacInitTargetRateBpsSwb = 56000 }; |
| // Lowest max rate for iSAC-wb |
| enum { kVoiceEngineMinIsacMaxRateBpsWb = 32000 }; |
| // Highest max rate for iSAC-wb |
| enum { kVoiceEngineMaxIsacMaxRateBpsWb = 53400 }; |
| // Lowest max rate for iSAC-swb |
| enum { kVoiceEngineMinIsacMaxRateBpsSwb = 32000 }; |
| // Highest max rate for iSAC-swb |
| enum { kVoiceEngineMaxIsacMaxRateBpsSwb = 107000 }; |
| // Lowest max payload size for iSAC-wb |
| enum { kVoiceEngineMinIsacMaxPayloadSizeBytesWb = 120 }; |
| // Highest max payload size for iSAC-wb |
| enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesWb = 400 }; |
| // Lowest max payload size for iSAC-swb |
| enum { kVoiceEngineMinIsacMaxPayloadSizeBytesSwb = 120 }; |
| // Highest max payload size for iSAC-swb |
| enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb = 600 }; |
| |
| // VideoSync |
| // Lowest minimum playout delay |
| enum { kVoiceEngineMinMinPlayoutDelayMs = 0 }; |
| // Highest minimum playout delay |
| enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 }; |
| |
| // Network |
| // Min packet-timeout time for received RTP packets |
| enum { kVoiceEngineMinPacketTimeoutSec = 1 }; |
| // Max packet-timeout time for received RTP packets |
| enum { kVoiceEngineMaxPacketTimeoutSec = 150 }; |
| // Min sample time for dead-or-alive detection |
| enum { kVoiceEngineMinSampleTimeSec = 1 }; |
| // Max sample time for dead-or-alive detection |
| enum { kVoiceEngineMaxSampleTimeSec = 150 }; |
| |
| // RTP/RTCP |
| // Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285) |
| enum { kVoiceEngineMinRtpExtensionId = 1 }; |
| // Max 4-bit ID for RTP extension |
| enum { kVoiceEngineMaxRtpExtensionId = 14 }; |
| |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Macros |
| // ---------------------------------------------------------------------------- |
| |
| #define NOT_SUPPORTED(stat) \ |
| LOG_F(LS_ERROR) << "not supported"; \ |
| stat.SetLastError(VE_FUNC_NOT_SUPPORTED); \ |
| return -1; |
| |
| #if (!defined(NDEBUG) && defined(_WIN32) && (_MSC_VER >= 1400)) |
| #include <windows.h> |
| #include <stdio.h> |
| #define DEBUG_PRINT(...) \ |
| { \ |
| char msg[256]; \ |
| sprintf(msg, __VA_ARGS__); \ |
| OutputDebugStringA(msg); \ |
| } |
| #else |
| // special fix for visual 2003 |
| #define DEBUG_PRINT(exp) ((void)0) |
| #endif // !defined(NDEBUG) && defined(_WIN32) |
| |
| #define CHECK_CHANNEL(channel) \ |
| if (CheckChannel(channel) == -1) \ |
| return -1; |
| |
| // ---------------------------------------------------------------------------- |
| // Inline functions |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc { |
| |
| inline int VoEId(int veId, int chId) { |
| if (chId == -1) { |
| const int dummyChannel(99); |
| return (int)((veId << 16) + dummyChannel); |
| } |
| return (int)((veId << 16) + chId); |
| } |
| |
| inline int VoEModuleId(int veId, int chId) { |
| return (int)((veId << 16) + chId); |
| } |
| |
| // Convert module ID to internal VoE channel ID |
| inline int VoEChannelId(int moduleId) { |
| return (int)(moduleId & 0xffff); |
| } |
| |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Platform settings |
| // ---------------------------------------------------------------------------- |
| |
| // *** WINDOWS *** |
| |
| #if defined(_WIN32) |
| |
| #include <windows.h> |
| |
| #pragma comment(lib, "winmm.lib") |
| |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| #pragma comment(lib, "ws2_32.lib") |
| #endif |
| |
| // ---------------------------------------------------------------------------- |
| // Defines |
| // ---------------------------------------------------------------------------- |
| |
| // Default device for Windows PC |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \ |
| AudioDeviceModule::kDefaultCommunicationDevice |
| |
| #endif // #if (defined(_WIN32) |
| |
| // *** LINUX *** |
| |
| #ifdef WEBRTC_LINUX |
| |
| #include <arpa/inet.h> |
| #include <netinet/in.h> |
| #include <pthread.h> |
| #include <sys/socket.h> |
| #include <sys/types.h> |
| #ifndef QNX |
| #include <linux/net.h> |
| #ifndef ANDROID |
| #include <sys/soundcard.h> |
| #endif // ANDROID |
| #endif // QNX |
| #include <errno.h> |
| #include <fcntl.h> |
| #include <sched.h> |
| #include <stdio.h> |
| #include <stdlib.h> |
| #include <string.h> |
| #include <sys/ioctl.h> |
| #include <sys/stat.h> |
| #include <sys/time.h> |
| #include <time.h> |
| #include <unistd.h> |
| |
| #define DWORD unsigned long int |
| #define WINAPI |
| #define LPVOID void * |
| #define FALSE 0 |
| #define TRUE 1 |
| #define UINT unsigned int |
| #define UCHAR unsigned char |
| #define TCHAR char |
| #ifdef QNX |
| #define _stricmp stricmp |
| #else |
| #define _stricmp strcasecmp |
| #endif |
| #define GetLastError() errno |
| #define WSAGetLastError() errno |
| #define LPCTSTR const char * |
| #define LPCSTR const char * |
| #define wsprintf sprintf |
| #define TEXT(a) a |
| #define _ftprintf fprintf |
| #define _tcslen strlen |
| #define FAR |
| #define __cdecl |
| #define LPSOCKADDR struct sockaddr * |
| |
| // Default device for Linux and Android |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0 |
| |
| #endif // #ifdef WEBRTC_LINUX |
| |
| // *** WEBRTC_MAC *** |
| // including iPhone |
| |
| #ifdef WEBRTC_MAC |
| |
| #include <AudioUnit/AudioUnit.h> |
| #include <arpa/inet.h> |
| #include <errno.h> |
| #include <fcntl.h> |
| #include <netinet/in.h> |
| #include <pthread.h> |
| #include <sched.h> |
| #include <stdio.h> |
| #include <stdlib.h> |
| #include <string.h> |
| #include <sys/socket.h> |
| #include <sys/stat.h> |
| #include <sys/time.h> |
| #include <sys/types.h> |
| #include <time.h> |
| #include <unistd.h> |
| #if !defined(WEBRTC_IOS) |
| #include <CoreServices/CoreServices.h> |
| #include <CoreAudio/CoreAudio.h> |
| #include <AudioToolbox/DefaultAudioOutput.h> |
| #include <AudioToolbox/AudioConverter.h> |
| #include <CoreAudio/HostTime.h> |
| #endif |
| |
| #define DWORD unsigned long int |
| #define WINAPI |
| #define LPVOID void * |
| #define FALSE 0 |
| #define TRUE 1 |
| #define SOCKADDR_IN struct sockaddr_in |
| #define UINT unsigned int |
| #define UCHAR unsigned char |
| #define TCHAR char |
| #define _stricmp strcasecmp |
| #define GetLastError() errno |
| #define WSAGetLastError() errno |
| #define LPCTSTR const char * |
| #define wsprintf sprintf |
| #define TEXT(a) a |
| #define _ftprintf fprintf |
| #define _tcslen strlen |
| #define FAR |
| #define __cdecl |
| #define LPSOCKADDR struct sockaddr * |
| #define LPCSTR const char * |
| #define ULONG unsigned long |
| |
| // Default device for Mac and iPhone |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0 |
| #endif // #ifdef WEBRTC_MAC |
| |
| #endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H |