GN: Add modules_unittests
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
* webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
* webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
diff --git a/webrtc/modules/audio_coding/audio_coding.gni b/webrtc/modules/audio_coding/audio_coding.gni
new file mode 100644
index 0000000..d346d47
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_coding.gni
@@ -0,0 +1,30 @@
+# 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("../../build/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 (!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