f66a925142
We used to link with all audio codecs unconditionally (except Opus); this patch makes gyp and gn only link to the ones that are used. (This unfortunately fails to have a measurable impact on Chromium binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC fix were already being excluded from Chromium by some other means (likely just the linker omitting compilation units with no incoming references).) BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1349393003 Cr-Commit-Position: refs/heads/master@{#10046}
90 lines
3.3 KiB
C
90 lines
3.3 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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// [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_VP9
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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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// 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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#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_
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