d2aa8f97f1
The frame transformer is passed from RTPSenderInterface through the library to be eventually set in ChannelSend, where the frame transformation will occur in the follow-up CL. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30946}
292 lines
8.8 KiB
Plaintext
292 lines
8.8 KiB
Plaintext
# 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_send.cc",
|
|
"channel_send.h",
|
|
"conversion.h",
|
|
"null_audio_poller.cc",
|
|
"null_audio_poller.h",
|
|
"remix_resample.cc",
|
|
"remix_resample.h",
|
|
]
|
|
|
|
deps = [
|
|
"../api:array_view",
|
|
"../api:call_api",
|
|
"../api:frame_transformer_interface",
|
|
"../api:function_view",
|
|
"../api:rtp_headers",
|
|
"../api:rtp_parameters",
|
|
"../api:scoped_refptr",
|
|
"../api:transport_api",
|
|
"../api/audio:aec3_factory",
|
|
"../api/audio:audio_frame_api",
|
|
"../api/audio:audio_mixer_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/crypto:frame_decryptor_interface",
|
|
"../api/crypto:frame_encryptor_interface",
|
|
"../api/crypto:options",
|
|
"../api/neteq:neteq_api",
|
|
"../api/rtc_event_log",
|
|
"../api/task_queue",
|
|
"../api/transport/rtp:rtp_source",
|
|
"../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",
|
|
"../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_device",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../modules/audio_processing:audio_frame_proxies",
|
|
"../modules/audio_processing:rms_level",
|
|
"../modules/pacing",
|
|
"../modules/remote_bitrate_estimator",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../modules/utility",
|
|
"../rtc_base",
|
|
"../rtc_base:audio_format_to_string",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rate_limiter",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:safe_minmax",
|
|
"../rtc_base/experiments:field_trial_parser",
|
|
"../rtc_base/synchronization:sequence_checker",
|
|
"../system_wrappers",
|
|
"../system_wrappers:field_trial",
|
|
"../system_wrappers:metrics",
|
|
"utility:audio_frame_operations",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
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/task_queue",
|
|
"../call:fake_network",
|
|
"../call:simulated_network",
|
|
"../system_wrappers",
|
|
"../test:test_common",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
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",
|
|
"mock_voe_channel_proxy.h",
|
|
"remix_resample_unittest.cc",
|
|
"test/audio_stats_test.cc",
|
|
]
|
|
deps = [
|
|
":audio",
|
|
":audio_end_to_end_test",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:mock_audio_mixer",
|
|
"../api:mock_frame_decryptor",
|
|
"../api:mock_frame_encryptor",
|
|
"../api/audio:audio_frame_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../api/audio_codecs/opus:audio_decoder_opus",
|
|
"../api/audio_codecs/opus:audio_encoder_opus",
|
|
"../api/rtc_event_log",
|
|
"../api/task_queue:default_task_queue_factory",
|
|
"../api/units:time_delta",
|
|
"../call:mock_bitrate_allocator",
|
|
"../call:mock_call_interfaces",
|
|
"../call:mock_rtp_interfaces",
|
|
"../call:rtp_interfaces",
|
|
"../call:rtp_receiver",
|
|
"../call:rtp_sender",
|
|
"../common_audio",
|
|
"../logging:mocks",
|
|
"../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule
|
|
"../modules/audio_device:mock_audio_device",
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
|
"../modules/audio_mixer:audio_mixer_test_utils",
|
|
"../modules/audio_processing:audio_processing_statistics",
|
|
"../modules/audio_processing:mocks",
|
|
"../modules/pacing",
|
|
"../modules/rtp_rtcp:mock_rtp_rtcp",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../modules/utility",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:safe_compare",
|
|
"../rtc_base:task_queue_for_test",
|
|
"../rtc_base:timeutils",
|
|
"../system_wrappers",
|
|
"../test:audio_codec_mocks",
|
|
"../test:field_trial",
|
|
"../test:mock_transport",
|
|
"../test:rtp_test_utils",
|
|
"../test:test_common",
|
|
"../test:test_support",
|
|
"utility:utility_tests",
|
|
"//testing/gtest",
|
|
]
|
|
}
|
|
|
|
if (rtc_enable_protobuf) {
|
|
rtc_test("low_bandwidth_audio_test") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/low_bandwidth_audio_test.cc",
|
|
"test/low_bandwidth_audio_test_flags.cc",
|
|
"test/pc_low_bandwidth_audio_test.cc",
|
|
]
|
|
|
|
deps = [
|
|
":audio_end_to_end_test",
|
|
"../api:create_network_emulation_manager",
|
|
"../api:create_peerconnection_quality_test_fixture",
|
|
"../api:network_emulation_manager_api",
|
|
"../api:peer_connection_quality_test_fixture_api",
|
|
"../api:simulated_network_api",
|
|
"../call:simulated_network",
|
|
"../common_audio",
|
|
"../system_wrappers",
|
|
"../test:fileutils",
|
|
"../test:perf_test",
|
|
"../test:test_common",
|
|
"../test:test_main",
|
|
"../test:test_support",
|
|
"../test/pc/e2e:network_quality_metrics_reporter",
|
|
"//testing/gtest",
|
|
"//third_party/abseil-cpp/absl/flags:flag",
|
|
]
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_native_code" ]
|
|
}
|
|
data = [
|
|
"../resources/voice_engine/audio_tiny16.wav",
|
|
"../resources/voice_engine/audio_tiny48.wav",
|
|
]
|
|
}
|
|
|
|
group("low_bandwidth_audio_perf_test") {
|
|
testonly = true
|
|
|
|
deps = [
|
|
":low_bandwidth_audio_test",
|
|
"//third_party/catapult/tracing/tracing/proto:histogram_proto",
|
|
"//third_party/protobuf:py_proto_runtime",
|
|
]
|
|
|
|
data = [
|
|
"test/low_bandwidth_audio_test.py",
|
|
"../resources/voice_engine/audio_tiny16.wav",
|
|
"../resources/voice_engine/audio_tiny48.wav",
|
|
"${root_out_dir}/pyproto/tracing/tracing/proto/histogram_pb2.py",
|
|
]
|
|
|
|
# TODO(http://crbug.com/1029452): Create a cleaner target with just the
|
|
# tracing python code. We don't need Polymer for instance.
|
|
data_deps = [ "//third_party/catapult/tracing:convert_chart_json" ]
|
|
|
|
if (is_win) {
|
|
data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ]
|
|
} else {
|
|
data += [ "${root_out_dir}/low_bandwidth_audio_test" ]
|
|
}
|
|
|
|
if (is_linux || is_android) {
|
|
data += [
|
|
"../tools_webrtc/audio_quality/linux/PolqaOem64",
|
|
"../tools_webrtc/audio_quality/linux/pesq",
|
|
]
|
|
}
|
|
if (is_win) {
|
|
data += [
|
|
"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
|
|
"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
|
|
"../tools_webrtc/audio_quality/win/pesq.exe",
|
|
"../tools_webrtc/audio_quality/win/vcomp120.dll",
|
|
]
|
|
}
|
|
if (is_mac) {
|
|
data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
|
|
}
|
|
|
|
write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps"
|
|
}
|
|
}
|
|
|
|
rtc_library("audio_perf_tests") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/audio_bwe_integration_test.cc",
|
|
"test/audio_bwe_integration_test.h",
|
|
]
|
|
deps = [
|
|
"../api:simulated_network_api",
|
|
"../api/task_queue",
|
|
"../call:fake_network",
|
|
"../call:simulated_network",
|
|
"../common_audio",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:task_queue_for_test",
|
|
"../system_wrappers",
|
|
"../test:field_trial",
|
|
"../test:fileutils",
|
|
"../test:test_common",
|
|
"../test:test_main",
|
|
"../test:test_support",
|
|
"//testing/gtest",
|
|
]
|
|
|
|
data = [ "//resources/voice_engine/audio_dtx16.wav" ]
|
|
}
|
|
}
|