| # Copyright (c) 2015 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. |
| |
| import("../webrtc.gni") |
| if (is_android) { |
| import("//build/config/android/config.gni") |
| import("//build/config/android/rules.gni") |
| } |
| |
| rtc_library("audio") { |
| sources = [ |
| "audio_level.cc", |
| "audio_level.h", |
| "audio_receive_stream.cc", |
| "audio_receive_stream.h", |
| "audio_send_stream.cc", |
| "audio_send_stream.h", |
| "audio_state.cc", |
| "audio_state.h", |
| "audio_transport_impl.cc", |
| "audio_transport_impl.h", |
| "channel_receive.cc", |
| "channel_receive.h", |
| "channel_receive_frame_transformer_delegate.cc", |
| "channel_receive_frame_transformer_delegate.h", |
| "channel_send.cc", |
| "channel_send.h", |
| "channel_send_frame_transformer_delegate.cc", |
| "channel_send_frame_transformer_delegate.h", |
| "conversion.h", |
| "remix_resample.cc", |
| "remix_resample.h", |
| ] |
| |
| deps = [ |
| "../api:array_view", |
| "../api:bitrate_allocation", |
| "../api:call_api", |
| "../api:field_trials_view", |
| "../api:frame_transformer_interface", |
| "../api:function_view", |
| "../api:make_ref_counted", |
| "../api:rtc_error", |
| "../api:rtp_headers", |
| "../api:rtp_packet_info", |
| "../api:rtp_parameters", |
| "../api:rtp_sender_interface", |
| "../api:scoped_refptr", |
| "../api:sequence_checker", |
| "../api:transport_api", |
| "../api/audio:aec3_factory", |
| "../api/audio:audio_device", |
| "../api/audio:audio_frame_api", |
| "../api/audio:audio_frame_processor", |
| "../api/audio:audio_mixer_api", |
| "../api/audio:audio_processing", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/crypto:frame_decryptor_interface", |
| "../api/crypto:frame_encryptor_interface", |
| "../api/crypto:options", |
| "../api/environment", |
| "../api/neteq:default_neteq_factory", |
| "../api/neteq:neteq_api", |
| "../api/rtc_event_log", |
| "../api/task_queue", |
| "../api/task_queue:pending_task_safety_flag", |
| "../api/transport/rtp:rtp_source", |
| "../api/units:data_rate", |
| "../api/units:data_size", |
| "../api/units:time_delta", |
| "../api/units:timestamp", |
| "../call:audio_sender_interface", |
| "../call:bitrate_allocator", |
| "../call:call_interfaces", |
| "../call:rtp_interfaces", |
| "../common_audio", |
| "../common_audio:common_audio_c", |
| "../logging:rtc_event_audio", |
| "../logging:rtc_stream_config", |
| "../media:media_channel", |
| "../media:media_channel_impl", |
| "../media:media_constants", |
| "../modules/async_audio_processing", |
| "../modules/audio_coding", |
| "../modules/audio_coding:audio_coding_module_typedefs", |
| "../modules/audio_coding:audio_encoder_cng", |
| "../modules/audio_coding:audio_network_adaptor_config", |
| "../modules/audio_coding:red", |
| "../modules/audio_device", |
| "../modules/audio_processing", |
| "../modules/audio_processing:audio_frame_proxies", |
| "../modules/audio_processing:rms_level", |
| "../modules/pacing", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:buffer", |
| "../rtc_base:checks", |
| "../rtc_base:event_tracer", |
| "../rtc_base:logging", |
| "../rtc_base:macromagic", |
| "../rtc_base:race_checker", |
| "../rtc_base:rate_limiter", |
| "../rtc_base:refcount", |
| "../rtc_base:rtc_event", |
| "../rtc_base:rtc_numerics", |
| "../rtc_base:safe_conversions", |
| "../rtc_base:safe_minmax", |
| "../rtc_base:stringutils", |
| "../rtc_base:threading", |
| "../rtc_base:timeutils", |
| "../rtc_base/containers:flat_set", |
| "../rtc_base/experiments:field_trial_parser", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/system:no_unique_address", |
| "../rtc_base/task_utils:repeating_task", |
| "../system_wrappers", |
| "../system_wrappers:metrics", |
| "utility:audio_frame_operations", |
| "//third_party/abseil-cpp/absl/functional:any_invocable", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/strings:string_view", |
| ] |
| } |
| if (rtc_include_tests) { |
| rtc_library("audio_end_to_end_test") { |
| testonly = true |
| |
| sources = [ |
| "test/audio_end_to_end_test.cc", |
| "test/audio_end_to_end_test.h", |
| ] |
| deps = [ |
| ":audio", |
| "../api:simulated_network_api", |
| "../api/audio:audio_device", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/task_queue", |
| "../call:call_interfaces", |
| "../call:fake_network", |
| "../modules/audio_device:test_audio_device_module", |
| "../system_wrappers", |
| "../test:test_common", |
| "../test:test_support", |
| "../test:video_test_constants", |
| ] |
| } |
| |
| rtc_library("audio_tests") { |
| testonly = true |
| |
| sources = [ |
| "audio_receive_stream_unittest.cc", |
| "audio_send_stream_tests.cc", |
| "audio_send_stream_unittest.cc", |
| "audio_state_unittest.cc", |
| "channel_receive_frame_transformer_delegate_unittest.cc", |
| "channel_send_frame_transformer_delegate_unittest.cc", |
| "channel_send_unittest.cc", |
| "mock_voe_channel_proxy.h", |
| "remix_resample_unittest.cc", |
| "test/audio_stats_test.cc", |
| "test/nack_test.cc", |
| "test/non_sender_rtt_test.cc", |
| ] |
| deps = [ |
| ":audio", |
| ":audio_end_to_end_test", |
| ":channel_receive_unittest", |
| "../api:array_view", |
| "../api:bitrate_allocation", |
| "../api:call_api", |
| "../api:field_trials", |
| "../api:frame_transformer_factory", |
| "../api:frame_transformer_interface", |
| "../api:function_view", |
| "../api:location", |
| "../api:make_ref_counted", |
| "../api:mock_audio_mixer", |
| "../api:mock_frame_decryptor", |
| "../api:mock_frame_encryptor", |
| "../api:mock_frame_transformer", |
| "../api:mock_transformable_audio_frame", |
| "../api:ref_count", |
| "../api:rtc_error_matchers", |
| "../api:rtp_headers", |
| "../api:rtp_parameters", |
| "../api:scoped_refptr", |
| "../api:simulated_network_api", |
| "../api:transport_api", |
| "../api/audio:audio_frame_api", |
| "../api/audio:audio_frame_processor", |
| "../api/audio:audio_mixer_api", |
| "../api/audio:audio_processing_statistics", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../api/crypto:frame_decryptor_interface", |
| "../api/crypto:frame_encryptor_interface", |
| "../api/crypto:options", |
| "../api/environment", |
| "../api/environment:environment_factory", |
| "../api/task_queue", |
| "../api/task_queue/test:mock_task_queue_base", |
| "../api/transport:bitrate_settings", |
| "../api/transport:network_control", |
| "../api/transport/rtp:rtp_source", |
| "../api/units:data_rate", |
| "../api/units:data_size", |
| "../api/units:time_delta", |
| "../api/units:timestamp", |
| "../call:bitrate_allocator", |
| "../call:call_interfaces", |
| "../call:mock_bitrate_allocator", |
| "../call:mock_call_interfaces", |
| "../call:mock_rtp_interfaces", |
| "../call:rtp_interfaces", |
| "../call:rtp_receiver", |
| "../call:rtp_sender", |
| "../common_audio", |
| "../modules/async_audio_processing", |
| "../modules/audio_coding:audio_coding_module_typedefs", |
| "../modules/audio_device:mock_audio_device", |
| "../modules/audio_mixer:audio_mixer_impl", |
| "../modules/audio_mixer:audio_mixer_test_utils", |
| "../modules/audio_processing:mocks", |
| "../modules/pacing", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:mock_rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:logging", |
| "../rtc_base:safe_compare", |
| "../rtc_base:task_queue_for_test", |
| "../rtc_base:threading", |
| "../rtc_base:timeutils", |
| "../system_wrappers", |
| "../test:audio_codec_mocks", |
| "../test:create_test_environment", |
| "../test:create_test_field_trials", |
| "../test:mock_transport", |
| "../test:rtp_test_utils", |
| "../test:run_loop", |
| "../test:test_common", |
| "../test:test_support", |
| "../test:video_test_constants", |
| "../test:wait_until", |
| "../test/time_controller", |
| "utility:utility_tests", |
| "//third_party/abseil-cpp/absl/functional:any_invocable", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings:string_view", |
| ] |
| } |
| |
| rtc_library("channel_receive_unittest") { |
| testonly = true |
| sources = [ "channel_receive_unittest.cc" ] |
| deps = [ |
| ":audio", |
| "../api:array_view", |
| "../api:make_ref_counted", |
| "../api:mock_frame_transformer", |
| "../api:scoped_refptr", |
| "../api:transport_api", |
| "../api/audio:audio_device", |
| "../api/audio:audio_frame_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/audio_codecs:builtin_audio_decoder_factory", |
| "../api/crypto:frame_decryptor_interface", |
| "../api/crypto:options", |
| "../api/environment:environment_factory", |
| "../api/task_queue:default_task_queue_factory", |
| "../api/units:time_delta", |
| "../api/units:timestamp", |
| "../logging:mocks", |
| "../modules/audio_device:mock_audio_device", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:ntp_time_util", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:logging", |
| "../rtc_base:stringutils", |
| "../rtc_base:threading", |
| "../rtc_base:timeutils", |
| "../system_wrappers", |
| "../test:audio_codec_mocks", |
| "../test:mock_transport", |
| "../test:test_support", |
| "../test/time_controller", |
| "//third_party/abseil-cpp/absl/strings", |
| ] |
| } |
| } |
