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Remove <(webrtc_root) from source file entries.

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This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.

Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).

I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.

BUG=4185
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37859004

Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kjellander@webrtc.org
2015-01-29 10:23:45 +00:00
parent 73ca1945ec
commit 2d2a1f9f05
18 changed files with 129 additions and 115 deletions
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66
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi
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@ -1,66 +0,0 @@
# Copyright (c) 2014 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.
{
'targets': [
{
'target_name': 'audio_codec_speed_tests',
'type': '<(gtest_target_type)',
'dependencies': [
'audio_processing',
'iSACFix',
'webrtc_opus',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'sources': [
'audio_codec_speed_test.h',
'audio_codec_speed_test.cc',
'<(webrtc_root)/modules/audio_coding/codecs/opus/opus_speed_test.cc',
'<(webrtc_root)/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
}],
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'audio_codec_speed_tests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):audio_codec_speed_tests_apk',
],
},
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'audio_codec_speed_tests_run',
'type': 'none',
'dependencies': [
'audio_codec_speed_tests',
],
'includes': [
'../../../../build/isolate.gypi',
],
'sources': [
'audio_codec_speed_tests.isolate',
],
},
],
}],
],
}

33
webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate
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@ -1,33 +0,0 @@
# Copyright (c) 2014 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.
{
'conditions': [
['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', {
'variables': {
'files': [
'<(DEPTH)/resources/audio_coding/music_stereo_48kHz.pcm',
'<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
'<(DEPTH)/resources/audio_coding/speech_mono_32_48kHz.pcm',
],
},
}],
['OS=="linux" or OS=="mac" or OS=="win"', {
'variables': {
'command': [
'<(DEPTH)/testing/test_env.py',
'<(PRODUCT_DIR)/audio_codec_speed_tests<(EXECUTABLE_SUFFIX)',
],
'files': [
'<(DEPTH)/DEPS',
'<(DEPTH)/testing/test_env.py',
'<(PRODUCT_DIR)/audio_codec_speed_tests<(EXECUTABLE_SUFFIX)',
],
},
}],
],
}