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