Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 158 insertions, 0 deletions

diff --git a/webrtc/engine_configurations.h b/webrtc/engine_configurations.h
new file mode 100644
index 0000000000..553aed4def
--- /dev/null
+++ b/webrtc/engine_configurations.h
@@ -0,0 +1,158 @@
+/*
+ *  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_
+
+// ============================================================================
+//                              Voice and Video
+// ============================================================================
+
+// Don't link in socket support in Chrome
+#ifdef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_EXTERNAL_TRANSPORT
+#endif
+
+// Optional to enable stand-alone
+// #define WEBRTC_EXTERNAL_TRANSPORT
+
+// ----------------------------------------------------------------------------
+//  [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
+
+// iLBC, G.722, PCM16B and Redundancy coding are excluded from Chromium and
+// Mozilla builds.
+#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_MOZILLA_BUILD)
+#define WEBRTC_CODEC_ILBC
+#define WEBRTC_CODEC_G722
+#define WEBRTC_CODEC_PCM16
+#define WEBRTC_CODEC_RED
+#endif  // !WEBRTC_CHROMIUM_BUILD && !WEBRTC_MOZILLA_BUILD
+
+// ----------------------------------------------------------------------------
+//  [Video] Codec settings
+// ----------------------------------------------------------------------------
+
+#define VIDEOCODEC_I420
+#define VIDEOCODEC_VP8
+
+// ============================================================================
+//                                 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
+#define WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
+
+#ifndef WEBRTC_CHROMIUM_BUILD
+#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_NETWORK_API
+#define WEBRTC_VOICE_ENGINE_RTP_RTCP_API
+#define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
+#define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
+
+#ifndef WEBRTC_CHROMIUM_BUILD
+#define WEBRTC_VOICE_ENGINE_CALL_REPORT_API
+#define WEBRTC_VOICE_ENGINE_ENCRYPTION_API
+#endif
+
+// ============================================================================
+//                                 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_ENCRYPTION_API
+#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
+#define WEBRTC_VIDEO_ENGINE_NETWORK_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
+// #define WEBRTC_SRTP
+// #define WEBRTC_SRTP_ALLOW_ROC_ITERATION
+
+#endif  // WEBRTC_ENGINE_CONFIGURATIONS_H_