83317146ba
This new helper class writes RTP packets to file in rtpdump format. A unit test is also included. The new test class is used while re-writing the test tool RTPcat. BUG=2692 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7768 4adac7df-926f-26a2-2b94-8c16560cd09d
167 lines
4.7 KiB
Python
167 lines
4.7 KiB
Python
# Copyright (c) 2013 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': [
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'../build/common.gypi',
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],
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'targets': [
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{
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'target_name': 'webrtc_test_common',
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'type': 'static_library',
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'sources': [
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'call_test.cc',
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'call_test.h',
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'configurable_frame_size_encoder.cc',
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'configurable_frame_size_encoder.h',
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'direct_transport.cc',
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'direct_transport.h',
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'encoder_settings.cc',
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'encoder_settings.h',
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'fake_audio_device.cc',
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'fake_audio_device.h',
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'fake_decoder.cc',
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'fake_decoder.h',
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'fake_encoder.cc',
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'fake_encoder.h',
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'fake_network_pipe.cc',
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'fake_network_pipe.h',
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'frame_generator_capturer.cc',
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'frame_generator_capturer.h',
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'mock_transport.h',
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'null_transport.cc',
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'null_transport.h',
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'rtp_rtcp_observer.h',
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'run_loop.cc',
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'run_loop.h',
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'statistics.cc',
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'statistics.h',
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'vcm_capturer.cc',
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'vcm_capturer.h',
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'video_capturer.cc',
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'video_capturer.h',
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'win/run_loop_win.cc',
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],
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'conditions': [
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['OS=="win"', {
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'sources!': [
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'run_loop.cc',
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],
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}],
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],
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'dependencies': [
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'<(webrtc_root)/base/base.gyp:rtc_base',
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'<(webrtc_root)/modules/modules.gyp:media_file',
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'<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
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'<(webrtc_root)/test/test.gyp:frame_generator',
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'<(webrtc_root)/test/test.gyp:test_support',
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'<(webrtc_root)/test/test.gyp:rtp_test_utils',
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'<(webrtc_root)/webrtc.gyp:webrtc',
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],
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},
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{
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'target_name': 'webrtc_test_renderer',
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'type': 'static_library',
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'sources': [
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'gl/gl_renderer.cc',
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'gl/gl_renderer.h',
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'linux/glx_renderer.cc',
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'linux/glx_renderer.h',
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'linux/video_renderer_linux.cc',
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'mac/video_renderer_mac.h',
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'mac/video_renderer_mac.mm',
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'null_platform_renderer.cc',
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'video_renderer.cc',
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'video_renderer.h',
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'win/d3d_renderer.cc',
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'win/d3d_renderer.h',
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],
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'conditions': [
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['OS=="linux"', {
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'sources!': [
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'null_platform_renderer.cc',
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],
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}],
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['OS=="mac"', {
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'sources!': [
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'null_platform_renderer.cc',
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],
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}],
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['OS!="linux" and OS!="mac"', {
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'sources!' : [
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'gl/gl_renderer.cc',
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'gl/gl_renderer.h',
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],
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}],
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['OS=="win"', {
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'sources!': [
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'null_platform_renderer.cc',
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],
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'include_dirs': [
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'<(directx_sdk_path)/Include',
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],
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}],
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],
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'dependencies': [
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/modules/modules.gyp:media_file',
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'<(webrtc_root)/test/test.gyp:frame_generator',
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'<(webrtc_root)/test/test.gyp:test_support',
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],
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'direct_dependent_settings': {
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'conditions': [
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['OS=="linux"', {
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'libraries': [
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'-lXext',
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'-lX11',
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'-lGL',
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],
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}],
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['OS=="android"', {
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'libraries' : [
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'-lGLESv2', '-llog',
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],
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}],
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['OS=="mac"', {
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'xcode_settings' : {
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'OTHER_LDFLAGS' : [
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'-framework Cocoa',
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'-framework OpenGL',
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'-framework CoreVideo',
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],
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},
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}],
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],
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},
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},
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],
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'conditions': [
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['include_tests==1', {
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'targets': [
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{
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'target_name': 'webrtc_test_common_unittests',
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'type': '<(gtest_target_type)',
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'dependencies': [
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'webrtc_test_common',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/testing/gmock.gyp:gmock',
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'<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
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'<(webrtc_root)/test/test.gyp:test_support_main',
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],
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'sources': [
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'fake_network_pipe_unittest.cc',
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'rtp_file_reader_unittest.cc',
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'rtp_file_writer_unittest.cc',
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],
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},
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], #targets
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}], # include_tests
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], # conditions
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}
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