8fa03a15ab
PCM16 can be useful for unit tests in Chromium. In particular Mikhal would like to use it for ChromeCast. This currently (r222592) has no impact on Chrome binary size, presumably because PCM16 is unused and the linker strips the symbols. To measure the potential impact, I looked at the size (bytes) of out/Release/vie_auto_test on Linux with various codecs removed: r4724 : 4567384 No PCM16 : 4565936 No ILBC : 4500424 No G722 : 4555800 No RED : 4565880 Giving the following size increases of adding each codec: PCM16 : 1.4 kB (0.03%) ILBC : 70.0 kB (1.49%) G722 : 11.6 kB (0.25%) RED : 1.5 kB (0.03%) R=mikhal@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2195005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
152 lines
5.7 KiB
C
152 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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// ============================================================================
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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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#define WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
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#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) && \
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!defined(WEBRTC_CHROMIUM_BUILD)
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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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#ifndef WEBRTC_CHROMIUM_BUILD
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#define WEBRTC_VOICE_ENGINE_CALL_REPORT_API
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#define WEBRTC_VOICE_ENGINE_ENCRYPTION_API
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#endif
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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_ENCRYPTION_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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#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_
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