6c35e0b0f7
This CL will lower the number of test targets in WebRTC by: Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006): * resampler_unittests * signal_processing_unittests * vad_unittests Merge into modules_unittests: * bitrate_controller_unittests * desktop_capture_unittests * media_file_unittests * remote_bitrate_estimator_unittests * rtp_rtcp_unittests * paced_sender_unittests Merge into test_support_unittests: * channel_transport_unittests channel_transport.gyp was also removed in favor for test.gyp. I had to remove a main method from rtcp_format_remb_unittest.cc since it caused the fileutils.h code to not be able to find the right project root path in ordrer to provide correct paths to test files. Buildbot configuration update will be synced with the commit of this CL. TEST=trybots BUG=1843 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
27 lines
899 B
Python
27 lines
899 B
Python
# 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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'includes': ['../../webrtc/build/common.gypi'],
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'targets': [
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{
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'target_name': 'audio_e2e_harness',
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'type': 'executable',
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'dependencies': [
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'<(webrtc_root)/test/test.gyp:channel_transport',
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'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
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],
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'sources': [
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'audio/audio_e2e_harness.cc',
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],
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},
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],
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}
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