f353dd59b5
Changes: 1. Documented return values of VoENetwork methods. 2. In VoENetworkImpl: replaced calls to SetLastError() with LOG_F(). SetLastError() is not used anymore because no one is calling LastError() to check for last error. Also, its usage is being removed in Video Engine and we want to be consistent. 3. In VoENetworkImpl: removed WEBRTC_TRACE() usage. 4. In VoENetworkImpl: replaced some defensive code with assert(). We are now assuming that the caller has called VoEBase::Init() where needed. We are also assuming that it is invalid to pass nullptr where data is expected. 5. Updated unit tests accordingly. R=henrika@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/53369004 Cr-Commit-Position: refs/heads/master@{#9145}
321 lines
9.2 KiB
C++
321 lines
9.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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/*
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* This file contains common constants for VoiceEngine, as well as
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* platform specific settings and include files.
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*/
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#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
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#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
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#include "webrtc/common_types.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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// ----------------------------------------------------------------------------
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// Enumerators
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// ----------------------------------------------------------------------------
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namespace webrtc {
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// Internal buffer size required for mono audio, based on the highest sample
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// rate voice engine supports (10 ms of audio at 192 kHz).
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static const int kMaxMonoDataSizeSamples = 1920;
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// VolumeControl
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enum { kMinVolumeLevel = 0 };
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enum { kMaxVolumeLevel = 255 };
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// Min scale factor for per-channel volume scaling
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const float kMinOutputVolumeScaling = 0.0f;
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// Max scale factor for per-channel volume scaling
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const float kMaxOutputVolumeScaling = 10.0f;
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// Min scale factor for output volume panning
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const float kMinOutputVolumePanning = 0.0f;
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// Max scale factor for output volume panning
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const float kMaxOutputVolumePanning = 1.0f;
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// DTMF
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enum { kMinDtmfEventCode = 0 }; // DTMF digit "0"
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enum { kMaxDtmfEventCode = 15 }; // DTMF digit "D"
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enum { kMinTelephoneEventCode = 0 }; // RFC4733 (Section 2.3.1)
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enum { kMaxTelephoneEventCode = 255 }; // RFC4733 (Section 2.3.1)
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enum { kMinTelephoneEventDuration = 100 };
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enum { kMaxTelephoneEventDuration = 60000 }; // Actual limit is 2^16
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enum { kMinTelephoneEventAttenuation = 0 }; // 0 dBm0
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enum { kMaxTelephoneEventAttenuation = 36 }; // -36 dBm0
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enum { kMinTelephoneEventSeparationMs = 100 }; // Min delta time between two
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// telephone events
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enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 }; // assumes Ethernet
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enum { kVoiceEngineMaxModuleVersionSize = 960 };
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// Base
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enum { kVoiceEngineVersionMaxMessageSize = 1024 };
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// Audio processing
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const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
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const GainControl::Mode kDefaultAgcMode =
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
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GainControl::kAdaptiveDigital;
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#else
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GainControl::kAdaptiveAnalog;
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#endif
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const bool kDefaultAgcState =
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
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false;
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#else
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true;
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#endif
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const GainControl::Mode kDefaultRxAgcMode = GainControl::kAdaptiveDigital;
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// Codec
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// Min init target rate for iSAC-wb
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enum { kVoiceEngineMinIsacInitTargetRateBpsWb = 10000 };
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// Max init target rate for iSAC-wb
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enum { kVoiceEngineMaxIsacInitTargetRateBpsWb = 32000 };
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// Min init target rate for iSAC-swb
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enum { kVoiceEngineMinIsacInitTargetRateBpsSwb = 10000 };
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// Max init target rate for iSAC-swb
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enum { kVoiceEngineMaxIsacInitTargetRateBpsSwb = 56000 };
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// Lowest max rate for iSAC-wb
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enum { kVoiceEngineMinIsacMaxRateBpsWb = 32000 };
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// Highest max rate for iSAC-wb
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enum { kVoiceEngineMaxIsacMaxRateBpsWb = 53400 };
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// Lowest max rate for iSAC-swb
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enum { kVoiceEngineMinIsacMaxRateBpsSwb = 32000 };
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// Highest max rate for iSAC-swb
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enum { kVoiceEngineMaxIsacMaxRateBpsSwb = 107000 };
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// Lowest max payload size for iSAC-wb
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enum { kVoiceEngineMinIsacMaxPayloadSizeBytesWb = 120 };
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// Highest max payload size for iSAC-wb
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enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesWb = 400 };
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// Lowest max payload size for iSAC-swb
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enum { kVoiceEngineMinIsacMaxPayloadSizeBytesSwb = 120 };
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// Highest max payload size for iSAC-swb
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enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb = 600 };
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// VideoSync
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// Lowest minimum playout delay
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enum { kVoiceEngineMinMinPlayoutDelayMs = 0 };
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// Highest minimum playout delay
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enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 };
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// Network
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// Min packet-timeout time for received RTP packets
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enum { kVoiceEngineMinPacketTimeoutSec = 1 };
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// Max packet-timeout time for received RTP packets
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enum { kVoiceEngineMaxPacketTimeoutSec = 150 };
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// Min sample time for dead-or-alive detection
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enum { kVoiceEngineMinSampleTimeSec = 1 };
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// Max sample time for dead-or-alive detection
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enum { kVoiceEngineMaxSampleTimeSec = 150 };
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// RTP/RTCP
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// Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285)
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enum { kVoiceEngineMinRtpExtensionId = 1 };
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// Max 4-bit ID for RTP extension
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enum { kVoiceEngineMaxRtpExtensionId = 14 };
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} // namespace webrtc
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// ----------------------------------------------------------------------------
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// Macros
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// ----------------------------------------------------------------------------
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#define NOT_SUPPORTED(stat) \
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LOG_F(LS_ERROR) << "not supported"; \
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stat.SetLastError(VE_FUNC_NOT_SUPPORTED); \
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return -1;
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#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
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#include <windows.h>
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#include <stdio.h>
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#define DEBUG_PRINT(...) \
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{ \
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char msg[256]; \
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sprintf(msg, __VA_ARGS__); \
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OutputDebugStringA(msg); \
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}
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#else
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// special fix for visual 2003
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#define DEBUG_PRINT(exp) ((void)0)
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#endif // defined(_DEBUG) && defined(_WIN32)
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#define CHECK_CHANNEL(channel) \
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if (CheckChannel(channel) == -1) \
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return -1;
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// ----------------------------------------------------------------------------
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// Inline functions
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// ----------------------------------------------------------------------------
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namespace webrtc {
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inline int VoEId(int veId, int chId) {
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if (chId == -1) {
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const int dummyChannel(99);
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return (int)((veId << 16) + dummyChannel);
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}
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return (int)((veId << 16) + chId);
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}
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inline int VoEModuleId(int veId, int chId) {
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return (int)((veId << 16) + chId);
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}
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// Convert module ID to internal VoE channel ID
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inline int VoEChannelId(int moduleId) {
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return (int)(moduleId & 0xffff);
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}
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} // namespace webrtc
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// ----------------------------------------------------------------------------
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// Platform settings
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// ----------------------------------------------------------------------------
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// *** WINDOWS ***
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#if defined(_WIN32)
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#include <windows.h>
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#pragma comment(lib, "winmm.lib")
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#ifndef WEBRTC_EXTERNAL_TRANSPORT
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#pragma comment(lib, "ws2_32.lib")
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#endif
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// ----------------------------------------------------------------------------
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// Defines
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// ----------------------------------------------------------------------------
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// Default device for Windows PC
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#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
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AudioDeviceModule::kDefaultCommunicationDevice
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#endif // #if (defined(_WIN32)
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// *** LINUX ***
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#ifdef WEBRTC_LINUX
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#include <arpa/inet.h>
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#include <netinet/in.h>
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#include <pthread.h>
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#include <sys/socket.h>
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#include <sys/types.h>
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#ifndef QNX
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#include <linux/net.h>
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#ifndef ANDROID
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#include <sys/soundcard.h>
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#endif // ANDROID
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#endif // QNX
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#include <errno.h>
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#include <fcntl.h>
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#include <sched.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/ioctl.h>
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#include <sys/stat.h>
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#include <sys/time.h>
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#include <time.h>
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#include <unistd.h>
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#define DWORD unsigned long int
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#define WINAPI
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#define LPVOID void *
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#define FALSE 0
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#define TRUE 1
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#define UINT unsigned int
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#define UCHAR unsigned char
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#define TCHAR char
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#ifdef QNX
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#define _stricmp stricmp
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#else
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#define _stricmp strcasecmp
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#endif
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#define GetLastError() errno
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#define WSAGetLastError() errno
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#define LPCTSTR const char *
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#define LPCSTR const char *
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#define wsprintf sprintf
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#define TEXT(a) a
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#define _ftprintf fprintf
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#define _tcslen strlen
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#define FAR
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#define __cdecl
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#define LPSOCKADDR struct sockaddr *
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// Default device for Linux and Android
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#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
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#endif // #ifdef WEBRTC_LINUX
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// *** WEBRTC_MAC ***
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// including iPhone
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#ifdef WEBRTC_MAC
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#include <AudioUnit/AudioUnit.h>
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#include <arpa/inet.h>
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#include <errno.h>
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#include <fcntl.h>
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#include <netinet/in.h>
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#include <pthread.h>
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#include <sched.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/socket.h>
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#include <sys/stat.h>
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#include <sys/time.h>
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#include <sys/types.h>
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#include <time.h>
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#include <unistd.h>
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#if !defined(WEBRTC_IOS)
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#include <CoreServices/CoreServices.h>
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#include <CoreAudio/CoreAudio.h>
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#include <AudioToolbox/DefaultAudioOutput.h>
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#include <AudioToolbox/AudioConverter.h>
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#include <CoreAudio/HostTime.h>
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#endif
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#define DWORD unsigned long int
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#define WINAPI
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#define LPVOID void *
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#define FALSE 0
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#define TRUE 1
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#define SOCKADDR_IN struct sockaddr_in
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#define UINT unsigned int
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#define UCHAR unsigned char
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#define TCHAR char
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#define _stricmp strcasecmp
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#define GetLastError() errno
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#define WSAGetLastError() errno
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#define LPCTSTR const char *
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#define wsprintf sprintf
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#define TEXT(a) a
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#define _ftprintf fprintf
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#define _tcslen strlen
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#define FAR
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#define __cdecl
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#define LPSOCKADDR struct sockaddr *
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#define LPCSTR const char *
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#define ULONG unsigned long
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// Default device for Mac and iPhone
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#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
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#endif // #ifdef WEBRTC_MAC
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#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
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