| /* |
| * 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 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 "engine_configurations.h" |
| |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| |
| // 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 { EcAec = 0 }; // AEC mode |
| enum { EcAecm = 1 }; // AECM mode |
| |
| enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 }; // assumes Ethernet |
| |
| enum { kVoiceEngineMaxModuleVersionSize = 960 }; |
| |
| // Base |
| enum { kVoiceEngineVersionMaxMessageSize = 1024 }; |
| |
| // Encryption |
| // SRTP uses 30 bytes key length |
| enum { kVoiceEngineMaxSrtpKeyLength = 30 }; |
| // SRTP minimum key/tag length for encryption level |
| enum { kVoiceEngineMinSrtpEncryptLength = 16 }; |
| // SRTP maximum key/tag length for encryption level |
| enum { kVoiceEngineMaxSrtpEncryptLength = 256 }; |
| // SRTP maximum key/tag length for authentication level, |
| // HMAC SHA1 authentication type |
| enum { kVoiceEngineMaxSrtpAuthSha1Length = 20 }; |
| // SRTP maximum tag length for authentication level, |
| // null authentication type |
| enum { kVoiceEngineMaxSrtpTagAuthNullLength = 12 }; |
| // SRTP maximum key length for authentication level, |
| // null authentication type |
| enum { kVoiceEngineMaxSrtpKeyAuthNullLength = 256 }; |
| |
| // Audio processing |
| enum { kVoiceEngineAudioProcessingDeviceSampleRateHz = 48000 }; |
| |
| // 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 = 1000 }; |
| |
| // 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 |
| |
| // TODO(andrew): we shouldn't be using the precompiler for this. |
| // Use enums or bools as appropriate. |
| #define WEBRTC_AUDIO_PROCESSING_OFF false |
| |
| #define WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE true |
| // AudioProcessing HP is ON |
| #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| // AudioProcessing NS off |
| #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE true |
| // AudioProcessing AGC on |
| #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| // AudioProcessing EC off |
| #define WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| // AudioProcessing off |
| #define WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| // AudioProcessing RX AGC off |
| #define WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| // AudioProcessing RX NS off |
| #define WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| // AudioProcessing RX High Pass Filter off |
| |
| #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE NoiseSuppression::kModerate |
| // AudioProcessing NS moderate suppression |
| #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE GainControl::kAdaptiveAnalog |
| // AudioProcessing AGC analog digital combined |
| #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAec |
| // AudioProcessing EC AEC |
| #define WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE GainControl::kAdaptiveDigital |
| // AudioProcessing AGC mode |
| #define WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE NoiseSuppression::kModerate |
| // AudioProcessing RX NS mode |
| |
| // Macros |
| // Comparison of two strings without regard to case |
| #define STR_CASE_CMP(x,y) ::_stricmp(x,y) |
| // Compares characters of two strings without regard to case |
| #define STR_NCASE_CMP(x,y,n) ::_strnicmp(x,y,n) |
| |
| // ---------------------------------------------------------------------------- |
| // Build information macros |
| // ---------------------------------------------------------------------------- |
| |
| #if defined(_DEBUG) |
| #define BUILDMODE "d" |
| #elif defined(DEBUG) |
| #define BUILDMODE "d" |
| #elif defined(NDEBUG) |
| #define BUILDMODE "r" |
| #else |
| #define BUILDMODE "?" |
| #endif |
| |
| #define BUILDTIME __TIME__ |
| #define BUILDDATE __DATE__ |
| |
| // Example: "Oct 10 2002 12:05:30 r" |
| #define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE |
| |
| // ---------------------------------------------------------------------------- |
| // Macros |
| // ---------------------------------------------------------------------------- |
| |
| #if (defined(_DEBUG) && 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(_DEBUG) && defined(_WIN32) |
| |
| #define CHECK_CHANNEL(channel) if (CheckChannel(channel) == -1) return -1; |
| |
| // ---------------------------------------------------------------------------- |
| // Default Trace filter |
| // ---------------------------------------------------------------------------- |
| |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_TRACE_FILTER \ |
| kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | \ |
| kTraceApiCall |
| |
| // ---------------------------------------------------------------------------- |
| // Inline functions |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| |
| inline int VoEId(const int veId, const int chId) |
| { |
| if (chId == -1) |
| { |
| const int dummyChannel(99); |
| return (int) ((veId << 16) + dummyChannel); |
| } |
| return (int) ((veId << 16) + chId); |
| } |
| |
| inline int VoEModuleId(const int veId, const int chId) |
| { |
| return (int) ((veId << 16) + chId); |
| } |
| |
| // Convert module ID to internal VoE channel ID |
| inline int VoEChannelId(const int moduleId) |
| { |
| return (int) (moduleId & 0xffff); |
| } |
| |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Platform settings |
| // ---------------------------------------------------------------------------- |
| |
| // *** WINDOWS *** |
| |
| #if defined(_WIN32) |
| |
| #pragma comment( lib, "winmm.lib" ) |
| |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| #pragma comment( lib, "ws2_32.lib" ) |
| #endif |
| |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| // Max number of supported channels |
| enum { kVoiceEngineMaxNumOfChannels = 32 }; |
| // Max number of channels which can be played out simultaneously |
| enum { kVoiceEngineMaxNumOfActiveChannels = 16 }; |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Defines |
| // ---------------------------------------------------------------------------- |
| |
| #include <windows.h> |
| #include <mmsystem.h> // timeGetTime |
| |
| #define GET_TIME_IN_MS() ::timeGetTime() |
| #define SLEEP(x) ::Sleep(x) |
| // Comparison of two strings without regard to case |
| #define STR_CASE_CMP(x,y) ::_stricmp(x,y) |
| // Compares characters of two strings without regard to case |
| #define STR_NCASE_CMP(x,y,n) ::_strnicmp(x,y,n) |
| |
| // Default device for Windows PC |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \ |
| AudioDeviceModule::kDefaultCommunicationDevice |
| |
| #endif // #if (defined(_WIN32) |
| |
| // *** LINUX *** |
| |
| #ifdef WEBRTC_LINUX |
| |
| #include <pthread.h> |
| #include <sys/types.h> |
| #include <sys/socket.h> |
| #include <netinet/in.h> |
| #include <arpa/inet.h> |
| #ifndef QNX |
| #include <linux/net.h> |
| #ifndef ANDROID |
| #include <sys/soundcard.h> |
| #endif // ANDROID |
| #endif // QNX |
| #include <stdio.h> |
| #include <string.h> |
| #include <stdlib.h> |
| #include <errno.h> |
| #include <sys/stat.h> |
| #include <sys/ioctl.h> |
| #include <unistd.h> |
| #include <fcntl.h> |
| #include <sched.h> |
| #include <time.h> |
| #include <sys/time.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 * |
| |
| namespace |
| { |
| void Sleep(unsigned long x) |
| { |
| timespec t; |
| t.tv_sec = x/1000; |
| t.tv_nsec = (x-(x/1000)*1000)*1000000; |
| nanosleep(&t,NULL); |
| } |
| |
| DWORD timeGetTime() |
| { |
| struct timeval tv; |
| struct timezone tz; |
| unsigned long val; |
| |
| gettimeofday(&tv, &tz); |
| val= tv.tv_sec*1000+ tv.tv_usec/1000; |
| return(val); |
| } |
| } |
| |
| #define SLEEP(x) ::Sleep(x) |
| #define GET_TIME_IN_MS timeGetTime |
| |
| // Default device for Linux and Android |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0 |
| |
| #ifdef ANDROID |
| |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| // Max number of supported channels |
| enum { kVoiceEngineMaxNumOfChannels = 2 }; |
| // Max number of channels which can be played out simultaneously |
| enum { kVoiceEngineMaxNumOfActiveChannels = 2 }; |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Defines |
| // ---------------------------------------------------------------------------- |
| |
| // Always excluded for Android builds |
| #undef WEBRTC_CODEC_ISAC |
| #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT |
| #undef WEBRTC_CONFERENCING |
| #undef WEBRTC_TYPING_DETECTION |
| |
| // Default audio processing states |
| #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE |
| #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE |
| #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE |
| #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| |
| // Default audio processing modes |
| #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE |
| #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE |
| #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE |
| #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE \ |
| NoiseSuppression::kModerate |
| #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE \ |
| GainControl::kAdaptiveDigital |
| #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAecm |
| |
| #define ANDROID_NOT_SUPPORTED(stat) \ |
| stat.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError, \ |
| "API call not supported"); \ |
| return -1; |
| |
| #else // LINUX PC |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| // Max number of supported channels |
| enum { kVoiceEngineMaxNumOfChannels = 32 }; |
| // Max number of channels which can be played out simultaneously |
| enum { kVoiceEngineMaxNumOfActiveChannels = 16 }; |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Defines |
| // ---------------------------------------------------------------------------- |
| |
| #define ANDROID_NOT_SUPPORTED(stat) |
| |
| #endif // ANDROID - LINUX PC |
| |
| #else |
| #define ANDROID_NOT_SUPPORTED(stat) |
| #endif // #ifdef WEBRTC_LINUX |
| |
| // *** WEBRTC_MAC *** |
| // including iPhone |
| |
| #ifdef WEBRTC_MAC |
| |
| #include <pthread.h> |
| #include <sys/types.h> |
| #include <sys/socket.h> |
| #include <netinet/in.h> |
| #include <arpa/inet.h> |
| #include <stdio.h> |
| #include <string.h> |
| #include <stdlib.h> |
| #include <errno.h> |
| #include <sys/stat.h> |
| #include <unistd.h> |
| #include <fcntl.h> |
| #include <sched.h> |
| #include <sys/time.h> |
| #include <time.h> |
| #include <AudioUnit/AudioUnit.h> |
| #if !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM) |
| #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 |
| |
| namespace |
| { |
| void Sleep(unsigned long x) |
| { |
| timespec t; |
| t.tv_sec = x/1000; |
| t.tv_nsec = (x-(x/1000)*1000)*1000000; |
| nanosleep(&t,NULL); |
| } |
| |
| DWORD WebRtcTimeGetTime() |
| { |
| struct timeval tv; |
| struct timezone tz; |
| unsigned long val; |
| |
| gettimeofday(&tv, &tz); |
| val= tv.tv_sec*1000+ tv.tv_usec/1000; |
| return(val); |
| } |
| } |
| |
| #define SLEEP(x) ::Sleep(x) |
| #define GET_TIME_IN_MS WebRtcTimeGetTime |
| |
| // Default device for Mac and iPhone |
| #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0 |
| |
| // iPhone specific |
| #if defined(MAC_IPHONE) || defined(MAC_IPHONE_SIM) |
| |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| // Max number of supported channels |
| enum { kVoiceEngineMaxNumOfChannels = 2 }; |
| // Max number of channels which can be played out simultaneously |
| enum { kVoiceEngineMaxNumOfActiveChannels = 2 }; |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Defines |
| // ---------------------------------------------------------------------------- |
| |
| // Always excluded for iPhone builds |
| #undef WEBRTC_CODEC_ISAC |
| #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT |
| |
| #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE |
| #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE |
| #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE |
| #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF |
| |
| #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE |
| #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE |
| #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE |
| #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE \ |
| NoiseSuppression::kModerate |
| #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE \ |
| GainControl::kAdaptiveDigital |
| #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAecm |
| |
| #define IPHONE_NOT_SUPPORTED() \ |
| _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError, \ |
| "API call not supported"); \ |
| return -1; |
| |
| #else // Non-iPhone |
| |
| // ---------------------------------------------------------------------------- |
| // Enumerators |
| // ---------------------------------------------------------------------------- |
| |
| namespace webrtc |
| { |
| // Max number of supported channels |
| enum { kVoiceEngineMaxNumOfChannels = 32 }; |
| // Max number of channels which can be played out simultaneously |
| enum { kVoiceEngineMaxNumOfActiveChannels = 16 }; |
| } // namespace webrtc |
| |
| // ---------------------------------------------------------------------------- |
| // Defines |
| // ---------------------------------------------------------------------------- |
| |
| #define IPHONE_NOT_SUPPORTED() |
| #endif |
| |
| #else |
| #define IPHONE_NOT_SUPPORTED() |
| #endif // #ifdef WEBRTC_MAC |
| |
| |
| |
| #endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H |
