Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/audio_coding.gni

@@ -0,0 +1,36 @@
+# Copyright (c) 2016 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.
+
+import("../../webrtc.gni")
+
+audio_codec_defines = []
+if (rtc_include_ilbc) {
+  audio_codec_defines += [ "WEBRTC_CODEC_ILBC" ]
+}
+if (rtc_include_opus) {
+  audio_codec_defines += [ "WEBRTC_CODEC_OPUS" ]
+}
+if (rtc_opus_support_120ms_ptime) {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
+} else {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
+}
+if (!build_with_mozilla) {
+  if (current_cpu == "arm") {
+    audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
+  } else {
+    audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ]
+  }
+  audio_codec_defines += [ "WEBRTC_CODEC_G722" ]
+}
+if (!build_with_mozilla && !build_with_chromium) {
+  audio_codec_defines += [ "WEBRTC_CODEC_RED" ]
+}
+
+audio_coding_defines = audio_codec_defines
+neteq_defines = audio_codec_defines