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webrtc_m130/webrtc/voice_engine/voice_engine.gyp
Stefan Holmer b86d4e4a8d Prepare the AudioSendStream to be hooked up to send-side BWE. 
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.

BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1479023002 .

Cr-Commit-Position: refs/heads/master@{#10909}
2015-12-07 09:26:32 +00:00

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# Copyright (c) 2011 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': 'voice_engine',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
'<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
'<(webrtc_root)/modules/modules.gyp:audio_device',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/modules/modules.gyp:bitrate_controller',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:paced_sender',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
],
'export_dependent_settings': [
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
],
'sources': [
'include/voe_audio_processing.h',
'include/voe_base.h',
'include/voe_codec.h',
'include/voe_dtmf.h',
'include/voe_errors.h',
'include/voe_external_media.h',
'include/voe_file.h',
'include/voe_hardware.h',
'include/voe_neteq_stats.h',
'include/voe_network.h',
'include/voe_rtp_rtcp.h',
'include/voe_video_sync.h',
'include/voe_volume_control.h',
'channel.cc',
'channel.h',
'channel_manager.cc',
'channel_manager.h',
'channel_proxy.cc',
'channel_proxy.h',
'dtmf_inband.cc',
'dtmf_inband.h',
'dtmf_inband_queue.cc',
'dtmf_inband_queue.h',
'level_indicator.cc',
'level_indicator.h',
'monitor_module.cc',
'monitor_module.h',
'network_predictor.cc',
'network_predictor.h',
'output_mixer.cc',
'output_mixer.h',
'shared_data.cc',
'shared_data.h',
'statistics.cc',
'statistics.h',
'transmit_mixer.cc',
'transmit_mixer.h',
'utility.cc',
'utility.h',
'voe_audio_processing_impl.cc',
'voe_audio_processing_impl.h',
'voe_base_impl.cc',
'voe_base_impl.h',
'voe_codec_impl.cc',
'voe_codec_impl.h',
'voe_dtmf_impl.cc',
'voe_dtmf_impl.h',
'voe_external_media_impl.cc',
'voe_external_media_impl.h',
'voe_file_impl.cc',
'voe_file_impl.h',
'voe_hardware_impl.cc',
'voe_hardware_impl.h',
'voe_neteq_stats_impl.cc',
'voe_neteq_stats_impl.h',
'voe_network_impl.cc',
'voe_network_impl.h',
'voe_rtp_rtcp_impl.cc',
'voe_rtp_rtcp_impl.h',
'voe_video_sync_impl.cc',
'voe_video_sync_impl.h',
'voe_volume_control_impl.cc',
'voe_volume_control_impl.h',
'voice_engine_defines.h',
'voice_engine_impl.cc',
'voice_engine_impl.h',
],
},
],
'conditions': [
['OS=="win"', {
'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',],
}],
['include_tests==1', {
'targets': [
{
'target_name': 'voice_engine_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
'voice_engine',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
# The rest are to satisfy the unittests' include chain.
# This would be unnecessary if we used qualified includes.
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/modules/modules.gyp:audio_device',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
'<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'sources': [
'channel_unittest.cc',
'network_predictor_unittest.cc',
'transmit_mixer_unittest.cc',
'utility_unittest.cc',
'voe_audio_processing_unittest.cc',
'voe_base_unittest.cc',
'voe_codec_unittest.cc',
'voe_network_unittest.cc',
'voice_engine_fixture.cc',
'voice_engine_fixture.h',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
},
{
# command line test that should work on linux/mac/win
'target_name': 'voe_cmd_test',
'type': 'executable',
'dependencies': [
'voice_engine',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
],
'sources': [
'test/cmd_test/voe_cmd_test.cc',
],
},
], # targets
'conditions': [
['OS!="ios"', {
'targets': [
{
'target_name': 'voe_auto_test',
'type': 'executable',
'dependencies': [
'voice_engine',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
],
'sources': [
'test/auto_test/automated_mode.cc',
'test/auto_test/extended/agc_config_test.cc',
'test/auto_test/extended/ec_metrics_test.cc',
'test/auto_test/fakes/conference_transport.cc',
'test/auto_test/fakes/conference_transport.h',
'test/auto_test/fakes/loudest_filter.cc',
'test/auto_test/fakes/loudest_filter.h',
'test/auto_test/fixtures/after_initialization_fixture.cc',
'test/auto_test/fixtures/after_initialization_fixture.h',
'test/auto_test/fixtures/after_streaming_fixture.cc',
'test/auto_test/fixtures/after_streaming_fixture.h',
'test/auto_test/fixtures/before_initialization_fixture.cc',
'test/auto_test/fixtures/before_initialization_fixture.h',
'test/auto_test/fixtures/before_streaming_fixture.cc',
'test/auto_test/fixtures/before_streaming_fixture.h',
'test/auto_test/standard/audio_processing_test.cc',
'test/auto_test/standard/codec_before_streaming_test.cc',
'test/auto_test/standard/codec_test.cc',
'test/auto_test/standard/dtmf_test.cc',
'test/auto_test/standard/external_media_test.cc',
'test/auto_test/standard/file_before_streaming_test.cc',
'test/auto_test/standard/file_test.cc',
'test/auto_test/standard/hardware_before_initializing_test.cc',
'test/auto_test/standard/hardware_before_streaming_test.cc',
'test/auto_test/standard/hardware_test.cc',
'test/auto_test/standard/mixing_test.cc',
'test/auto_test/standard/neteq_stats_test.cc',
'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc',
'test/auto_test/standard/rtp_rtcp_extensions.cc',
'test/auto_test/standard/rtp_rtcp_test.cc',
'test/auto_test/standard/voe_base_misc_test.cc',
'test/auto_test/standard/video_sync_test.cc',
'test/auto_test/standard/volume_test.cc',
'test/auto_test/resource_manager.cc',
'test/auto_test/voe_conference_test.cc',
'test/auto_test/voe_cpu_test.cc',
'test/auto_test/voe_cpu_test.h',
'test/auto_test/voe_output_test.cc',
'test/auto_test/voe_standard_test.cc',
'test/auto_test/voe_standard_test.h',
'test/auto_test/voe_stress_test.cc',
'test/auto_test/voe_stress_test.h',
'test/auto_test/voe_test_defines.h',
'test/auto_test/voe_test_interface.h',
],
'conditions': [
['OS=="android"', {
# some tests are not supported on android yet, exclude these tests.
'sources!': [
'test/auto_test/standard/hardware_before_streaming_test.cc',
],
}],
['enable_protobuf==1', {
'defines': [
'ENABLE_RTC_EVENT_LOG',
],
}],
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
],
}],
['OS=="android"', {
'targets': [
{
'target_name': 'voice_engine_unittests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):voice_engine_unittests_apk',
],
},
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'voice_engine_unittests_run',
'type': 'none',
'dependencies': [
'voice_engine_unittests',
],
'includes': [
'../build/isolate.gypi',
],
'sources': [
'voice_engine_unittests.isolate',
],
},
{
'target_name': 'voe_auto_test_run',
'type': 'none',
'dependencies': [
'voe_auto_test',
],
'includes': [
'../build/isolate.gypi',
],
'sources': [
'voe_auto_test.isolate',
],
},
],
}],
], # conditions
}], # include_tests==1
], # conditions
}
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