98ab3a46d6
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. (This was previously landed as revisions 10046 and 10060, and got reverted because it broke several of the Chromium FYI bots.) BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1368843003 Cr-Commit-Position: refs/heads/master@{#10127}
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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