1# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2# 3# Use of this source code is governed by a BSD-style license 4# that can be found in the LICENSE file in the root of the source 5# tree. An additional intellectual property rights grant can be found 6# in the file PATENTS. All contributing project authors may 7# be found in the AUTHORS file in the root of the source tree. 8 9import("../webrtc.gni") 10if (is_android) { 11 import("//build/config/android/config.gni") 12 import("//build/config/android/rules.gni") 13} 14 15rtc_library("audio") { 16 sources = [ 17 "audio_level.cc", 18 "audio_level.h", 19 "audio_receive_stream.cc", 20 "audio_receive_stream.h", 21 "audio_send_stream.cc", 22 "audio_send_stream.h", 23 "audio_state.cc", 24 "audio_state.h", 25 "audio_transport_impl.cc", 26 "audio_transport_impl.h", 27 "channel_receive.cc", 28 "channel_receive.h", 29 "channel_receive_frame_transformer_delegate.cc", 30 "channel_receive_frame_transformer_delegate.h", 31 "channel_send.cc", 32 "channel_send.h", 33 "channel_send_frame_transformer_delegate.cc", 34 "channel_send_frame_transformer_delegate.h", 35 "conversion.h", 36 "null_audio_poller.cc", 37 "null_audio_poller.h", 38 "remix_resample.cc", 39 "remix_resample.h", 40 ] 41 42 deps = [ 43 "../api:array_view", 44 "../api:call_api", 45 "../api:frame_transformer_interface", 46 "../api:function_view", 47 "../api:rtp_headers", 48 "../api:rtp_parameters", 49 "../api:scoped_refptr", 50 "../api:transport_api", 51 "../api/audio:aec3_factory", 52 "../api/audio:audio_frame_api", 53 "../api/audio:audio_mixer_api", 54 "../api/audio_codecs:audio_codecs_api", 55 "../api/crypto:frame_decryptor_interface", 56 "../api/crypto:frame_encryptor_interface", 57 "../api/crypto:options", 58 "../api/neteq:neteq_api", 59 "../api/rtc_event_log", 60 "../api/task_queue", 61 "../api/transport/rtp:rtp_source", 62 "../call:audio_sender_interface", 63 "../call:bitrate_allocator", 64 "../call:call_interfaces", 65 "../call:rtp_interfaces", 66 "../common_audio", 67 "../common_audio:common_audio_c", 68 "../logging:rtc_event_audio", 69 "../logging:rtc_stream_config", 70 "../modules/audio_coding", 71 "../modules/audio_coding:audio_coding_module_typedefs", 72 "../modules/audio_coding:audio_encoder_cng", 73 "../modules/audio_coding:audio_network_adaptor_config", 74 "../modules/audio_coding:red", 75 "../modules/audio_device", 76 "../modules/audio_processing", 77 "../modules/audio_processing:api", 78 "../modules/audio_processing:audio_frame_proxies", 79 "../modules/audio_processing:rms_level", 80 "../modules/pacing", 81 "../modules/remote_bitrate_estimator", 82 "../modules/rtp_rtcp", 83 "../modules/rtp_rtcp:rtp_rtcp_format", 84 "../modules/utility", 85 "../rtc_base", 86 "../rtc_base:audio_format_to_string", 87 "../rtc_base:checks", 88 "../rtc_base:rate_limiter", 89 "../rtc_base:rtc_base_approved", 90 "../rtc_base:rtc_task_queue", 91 "../rtc_base:safe_minmax", 92 "../rtc_base/experiments:field_trial_parser", 93 "../rtc_base/synchronization:mutex", 94 "../rtc_base/synchronization:sequence_checker", 95 "../rtc_base/task_utils:to_queued_task", 96 "../system_wrappers", 97 "../system_wrappers:field_trial", 98 "../system_wrappers:metrics", 99 "utility:audio_frame_operations", 100 ] 101 absl_deps = [ 102 "//third_party/abseil-cpp/absl/memory", 103 "//third_party/abseil-cpp/absl/types:optional", 104 ] 105} 106if (rtc_include_tests) { 107 rtc_library("audio_end_to_end_test") { 108 testonly = true 109 110 sources = [ 111 "test/audio_end_to_end_test.cc", 112 "test/audio_end_to_end_test.h", 113 ] 114 deps = [ 115 ":audio", 116 "../api:simulated_network_api", 117 "../api/task_queue", 118 "../call:fake_network", 119 "../call:simulated_network", 120 "../system_wrappers", 121 "../test:test_common", 122 "../test:test_support", 123 ] 124 } 125 126 rtc_library("audio_tests") { 127 testonly = true 128 129 sources = [ 130 "audio_receive_stream_unittest.cc", 131 "audio_send_stream_tests.cc", 132 "audio_send_stream_unittest.cc", 133 "audio_state_unittest.cc", 134 "channel_receive_frame_transformer_delegate_unittest.cc", 135 "channel_send_frame_transformer_delegate_unittest.cc", 136 "mock_voe_channel_proxy.h", 137 "remix_resample_unittest.cc", 138 "test/audio_stats_test.cc", 139 ] 140 deps = [ 141 ":audio", 142 ":audio_end_to_end_test", 143 "../api:libjingle_peerconnection_api", 144 "../api:mock_audio_mixer", 145 "../api:mock_frame_decryptor", 146 "../api:mock_frame_encryptor", 147 "../api/audio:audio_frame_api", 148 "../api/audio_codecs:audio_codecs_api", 149 "../api/audio_codecs/opus:audio_decoder_opus", 150 "../api/audio_codecs/opus:audio_encoder_opus", 151 "../api/rtc_event_log", 152 "../api/task_queue:default_task_queue_factory", 153 "../api/units:time_delta", 154 "../call:mock_bitrate_allocator", 155 "../call:mock_call_interfaces", 156 "../call:mock_rtp_interfaces", 157 "../call:rtp_interfaces", 158 "../call:rtp_receiver", 159 "../call:rtp_sender", 160 "../common_audio", 161 "../logging:mocks", 162 "../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule 163 "../modules/audio_device:mock_audio_device", 164 "../modules/audio_mixer:audio_mixer_impl", 165 "../modules/audio_mixer:audio_mixer_test_utils", 166 "../modules/audio_processing:audio_processing_statistics", 167 "../modules/audio_processing:mocks", 168 "../modules/pacing", 169 "../modules/rtp_rtcp:mock_rtp_rtcp", 170 "../modules/rtp_rtcp:rtp_rtcp_format", 171 "../modules/utility", 172 "../rtc_base:checks", 173 "../rtc_base:rtc_base_approved", 174 "../rtc_base:rtc_base_tests_utils", 175 "../rtc_base:safe_compare", 176 "../rtc_base:task_queue_for_test", 177 "../rtc_base:timeutils", 178 "../system_wrappers", 179 "../test:audio_codec_mocks", 180 "../test:field_trial", 181 "../test:mock_frame_transformer", 182 "../test:mock_transformable_frame", 183 "../test:mock_transport", 184 "../test:rtp_test_utils", 185 "../test:test_common", 186 "../test:test_support", 187 "utility:utility_tests", 188 "//testing/gtest", 189 ] 190 } 191 192 if (rtc_enable_protobuf) { 193 rtc_test("low_bandwidth_audio_test") { 194 testonly = true 195 196 sources = [ 197 "test/low_bandwidth_audio_test.cc", 198 "test/low_bandwidth_audio_test_flags.cc", 199 "test/pc_low_bandwidth_audio_test.cc", 200 ] 201 202 deps = [ 203 ":audio_end_to_end_test", 204 "../api:create_network_emulation_manager", 205 "../api:create_peerconnection_quality_test_fixture", 206 "../api:network_emulation_manager_api", 207 "../api:peer_connection_quality_test_fixture_api", 208 "../api:simulated_network_api", 209 "../api:time_controller", 210 "../call:simulated_network", 211 "../common_audio", 212 "../system_wrappers", 213 "../test:fileutils", 214 "../test:perf_test", 215 "../test:test_common", 216 "../test:test_main", 217 "../test:test_support", 218 "../test/pc/e2e:network_quality_metrics_reporter", 219 "//testing/gtest", 220 "//third_party/abseil-cpp/absl/flags:flag", 221 ] 222 if (is_android) { 223 deps += [ "//testing/android/native_test:native_test_native_code" ] 224 } 225 data = [ 226 "../resources/voice_engine/audio_tiny16.wav", 227 "../resources/voice_engine/audio_tiny48.wav", 228 ] 229 } 230 231 group("low_bandwidth_audio_perf_test") { 232 testonly = true 233 234 deps = [ 235 ":low_bandwidth_audio_test", 236 "//third_party/catapult/tracing/tracing/proto:histogram_proto", 237 "//third_party/protobuf:py_proto_runtime", 238 ] 239 240 data = [ 241 "test/low_bandwidth_audio_test.py", 242 "../resources/voice_engine/audio_tiny16.wav", 243 "../resources/voice_engine/audio_tiny48.wav", 244 "${root_out_dir}/pyproto/tracing/tracing/proto/histogram_pb2.py", 245 ] 246 247 # TODO(http://crbug.com/1029452): Create a cleaner target with just the 248 # tracing python code. We don't need Polymer for instance. 249 data_deps = [ "//third_party/catapult/tracing:convert_chart_json" ] 250 251 if (is_win) { 252 data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ] 253 } else { 254 data += [ "${root_out_dir}/low_bandwidth_audio_test" ] 255 } 256 257 if (is_linux || is_android) { 258 data += [ 259 "../tools_webrtc/audio_quality/linux/PolqaOem64", 260 "../tools_webrtc/audio_quality/linux/pesq", 261 ] 262 } 263 if (is_win) { 264 data += [ 265 "../tools_webrtc/audio_quality/win/PolqaOem64.dll", 266 "../tools_webrtc/audio_quality/win/PolqaOem64.exe", 267 "../tools_webrtc/audio_quality/win/pesq.exe", 268 "../tools_webrtc/audio_quality/win/vcomp120.dll", 269 ] 270 } 271 if (is_mac) { 272 data += [ "../tools_webrtc/audio_quality/mac/pesq" ] 273 } 274 275 write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps" 276 } 277 } 278 279 rtc_library("audio_perf_tests") { 280 testonly = true 281 282 sources = [ 283 "test/audio_bwe_integration_test.cc", 284 "test/audio_bwe_integration_test.h", 285 ] 286 deps = [ 287 "../api:simulated_network_api", 288 "../api/task_queue", 289 "../call:fake_network", 290 "../call:simulated_network", 291 "../common_audio", 292 "../rtc_base:rtc_base_approved", 293 "../rtc_base:task_queue_for_test", 294 "../system_wrappers", 295 "../test:field_trial", 296 "../test:fileutils", 297 "../test:test_common", 298 "../test:test_main", 299 "../test:test_support", 300 "//testing/gtest", 301 ] 302 303 data = [ "//resources/voice_engine/audio_dtx16.wav" ] 304 } 305} 306