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156 lines
5.7 KiB
C
156 lines
5.7 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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#ifndef WEBRTC_ENGINE_CONFIGURATIONS_H_
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#define WEBRTC_ENGINE_CONFIGURATIONS_H_
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#include "webrtc/typedefs.h"
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// ============================================================================
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// Voice and Video
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// ============================================================================
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// ----------------------------------------------------------------------------
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// [Voice] Codec settings
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// ----------------------------------------------------------------------------
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// iSAC is not included in the Mozilla build, but in all other builds.
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#ifndef WEBRTC_MOZILLA_BUILD
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#ifdef WEBRTC_ARCH_ARM
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#define WEBRTC_CODEC_ISACFX // Fix-point iSAC implementation.
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#else
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#define WEBRTC_CODEC_ISAC // Floating-point iSAC implementation (default).
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#endif // WEBRTC_ARCH_ARM
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#endif // !WEBRTC_MOZILLA_BUILD
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// AVT is included in all builds, along with G.711, NetEQ and CNG
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// (which are mandatory and don't have any defines).
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#define WEBRTC_CODEC_AVT
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// PCM16 is useful for testing and incurs only a small binary size cost.
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#define WEBRTC_CODEC_PCM16
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// iLBC, G.722, and Redundancy coding are excluded from Chromium and Mozilla
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// builds to reduce binary size.
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#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_MOZILLA_BUILD)
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#define WEBRTC_CODEC_ILBC
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#define WEBRTC_CODEC_G722
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#define WEBRTC_CODEC_RED
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#endif // !WEBRTC_CHROMIUM_BUILD && !WEBRTC_MOZILLA_BUILD
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// ----------------------------------------------------------------------------
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// [Video] Codec settings
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// ----------------------------------------------------------------------------
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#define VIDEOCODEC_I420
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#define VIDEOCODEC_VP8
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#define VIDEOCODEC_H264
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// ============================================================================
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// VoiceEngine
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// ============================================================================
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// ----------------------------------------------------------------------------
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// Settings for VoiceEngine
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// ----------------------------------------------------------------------------
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#define WEBRTC_VOICE_ENGINE_AGC // Near-end AGC
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#define WEBRTC_VOICE_ENGINE_ECHO // Near-end AEC
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#define WEBRTC_VOICE_ENGINE_NR // Near-end NS
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#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
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#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION // Typing detection
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#endif
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// ----------------------------------------------------------------------------
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// VoiceEngine sub-APIs
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// ----------------------------------------------------------------------------
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#define WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
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#define WEBRTC_VOICE_ENGINE_CODEC_API
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#define WEBRTC_VOICE_ENGINE_DTMF_API
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#define WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
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#define WEBRTC_VOICE_ENGINE_FILE_API
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#define WEBRTC_VOICE_ENGINE_HARDWARE_API
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#define WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
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#define WEBRTC_VOICE_ENGINE_RTP_RTCP_API
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#define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
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#define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
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// ============================================================================
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// VideoEngine
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// ============================================================================
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// ----------------------------------------------------------------------------
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// Settings for special VideoEngine configurations
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// ----------------------------------------------------------------------------
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// ----------------------------------------------------------------------------
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// VideoEngine sub-API:s
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// ----------------------------------------------------------------------------
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#define WEBRTC_VIDEO_ENGINE_CAPTURE_API
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#define WEBRTC_VIDEO_ENGINE_CODEC_API
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#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
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#define WEBRTC_VIDEO_ENGINE_RENDER_API
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#define WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
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#define WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
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// Now handled by gyp:
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// WEBRTC_VIDEO_ENGINE_FILE_API
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// ============================================================================
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// Platform specific configurations
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// ============================================================================
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// ----------------------------------------------------------------------------
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// VideoEngine Windows
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// ----------------------------------------------------------------------------
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#if defined(_WIN32)
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#define DIRECT3D9_RENDERING // Requires DirectX 9.
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#endif
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// ----------------------------------------------------------------------------
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// VideoEngine MAC
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// ----------------------------------------------------------------------------
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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// #define CARBON_RENDERING
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#define COCOA_RENDERING
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#endif
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// ----------------------------------------------------------------------------
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// VideoEngine Mobile iPhone
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// ----------------------------------------------------------------------------
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#if defined(WEBRTC_IOS)
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#define EAGL_RENDERING
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#endif
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// ----------------------------------------------------------------------------
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// Deprecated
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// ----------------------------------------------------------------------------
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// #define WEBRTC_CODEC_G729
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// #define WEBRTC_DTMF_DETECTION
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// For RedPhone
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#undef WEBRTC_CODEC_CELT
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#undef WEBRTC_CODEC_G722
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#undef WEBRTC_CODEC_ILBC
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#undef WEBRTC_CODEC_ISACFX
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#undef WEBRTC_CODEC_ISAC
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#undef WEBRTC_CODEC_OPUS
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#undef WEBRTC_CODEC_PCM16
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#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_
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