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GN: Add modules_unittests

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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}
This commit is contained in:
kjellander
2016-06-13 00:19:48 -07:00
committed by Commit bot
parent 142f8c5b3b
commit fb11424551
14 changed files with 874 additions and 39 deletions
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30
webrtc/modules/audio_coding/audio_coding.gni Normal file
View File

@ -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