f577ae9eac
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings struct. The EncoderSettings struct uses an external encoder for all codecs. This means that external users, such as libjingle, will provide the encoders themselves, removing the previous distinction of internal and external codecs. For now VideoSendStream translates to VideoCodec internally. In the interrim (before the corresponding change is implemented in VideoReceiveStream) tests convert EncoderSettings to VideoCodecs. Removes Call::GetVideoCodecs(). Disables RampUpTest.WithPacingAndRtx as its further exposed with changes to bitrates used in tests. BUG=2854,2992 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
115 lines
3.4 KiB
Python
115 lines
3.4 KiB
Python
# Copyright (c) 2013 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.
|
|
{
|
|
'includes': [
|
|
'../build/common.gypi',
|
|
],
|
|
'targets': [
|
|
{
|
|
'target_name': 'webrtc_test_common',
|
|
'type': 'static_library',
|
|
'sources': [
|
|
'configurable_frame_size_encoder.cc',
|
|
'configurable_frame_size_encoder.h',
|
|
'direct_transport.cc',
|
|
'direct_transport.h',
|
|
'encoder_settings.cc',
|
|
'encoder_settings.h',
|
|
'fake_audio_device.cc',
|
|
'fake_audio_device.h',
|
|
'fake_decoder.cc',
|
|
'fake_decoder.h',
|
|
'fake_encoder.cc',
|
|
'fake_encoder.h',
|
|
'fake_network_pipe.cc',
|
|
'fake_network_pipe.h',
|
|
'flags.cc',
|
|
'flags.h',
|
|
'frame_generator_capturer.cc',
|
|
'frame_generator_capturer.h',
|
|
'mock_transport.h',
|
|
'null_platform_renderer.cc',
|
|
'null_transport.cc',
|
|
'null_transport.h',
|
|
'rtp_rtcp_observer.h',
|
|
'run_tests.cc',
|
|
'run_tests.h',
|
|
'run_loop.cc',
|
|
'run_loop.h',
|
|
'statistics.cc',
|
|
'statistics.h',
|
|
'vcm_capturer.cc',
|
|
'vcm_capturer.h',
|
|
'video_capturer.cc',
|
|
'video_capturer.h',
|
|
'video_renderer.cc',
|
|
'video_renderer.h',
|
|
],
|
|
# TODO(pbos): As far as I can tell these are dependencies from
|
|
# video_render and they should really not be here. This target provides
|
|
# no platform-specific rendering.
|
|
'direct_dependent_settings': {
|
|
'conditions': [
|
|
['OS=="linux"', {
|
|
'libraries': [
|
|
'-lXext',
|
|
'-lX11',
|
|
'-lGL',
|
|
],
|
|
}],
|
|
['OS=="android"', {
|
|
'libraries' : [
|
|
'-lGLESv2', '-llog',
|
|
],
|
|
}],
|
|
['OS=="mac"', {
|
|
'xcode_settings' : {
|
|
'OTHER_LDFLAGS' : [
|
|
'-framework Foundation',
|
|
'-framework AppKit',
|
|
'-framework Cocoa',
|
|
'-framework OpenGL',
|
|
'-framework CoreVideo',
|
|
'-framework CoreAudio',
|
|
'-framework AudioToolbox',
|
|
],
|
|
},
|
|
}],
|
|
],
|
|
},
|
|
'dependencies': [
|
|
'<(DEPTH)/testing/gtest.gyp:gtest',
|
|
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
|
|
'<(webrtc_root)/modules/modules.gyp:video_capture_module',
|
|
'<(webrtc_root)/modules/modules.gyp:media_file',
|
|
'<(webrtc_root)/test/test.gyp:frame_generator',
|
|
'<(webrtc_root)/test/test.gyp:test_support',
|
|
],
|
|
},
|
|
],
|
|
'conditions': [
|
|
['include_tests==1', {
|
|
'targets': [
|
|
{
|
|
'target_name': 'webrtc_test_common_unittests',
|
|
'type': '<(gtest_target_type)',
|
|
'dependencies': [
|
|
'webrtc_test_common',
|
|
'<(DEPTH)/testing/gtest.gyp:gtest',
|
|
'<(DEPTH)/testing/gmock.gyp:gmock',
|
|
'<(webrtc_root)/test/test.gyp:test_support_main',
|
|
],
|
|
'sources': [
|
|
'fake_network_pipe_unittest.cc',
|
|
],
|
|
},
|
|
], #targets
|
|
}], # include_tests
|
|
], # conditions
|
|
}
|