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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/audio/BUILD.gn
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/BUILD.gn')
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diff --git a/third_party/libwebrtc/audio/BUILD.gn b/third_party/libwebrtc/audio/BUILD.gn
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+# 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:call_api",
+ "../api:field_trials_view",
+ "../api:frame_transformer_interface",
+ "../api:function_view",
+ "../api:rtp_headers",
+ "../api:rtp_parameters",
+ "../api:scoped_refptr",
+ "../api:sequence_checker",
+ "../api:transport_api",
+ "../api/audio:aec3_factory",
+ "../api/audio:audio_frame_api",
+ "../api/audio:audio_frame_processor",
+ "../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/task_queue:pending_task_safety_flag",
+ "../api/transport/rtp:rtp_source",
+ "../api/units:time_delta",
+ "../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:rtc_media_base",
+ "../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:api",
+ "../modules/audio_processing:audio_frame_proxies",
+ "../modules/audio_processing:rms_level",
+ "../modules/pacing",
+ "../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:audio_format_to_string",
+ "../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:rtc_task_queue",
+ "../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:field_trial",
+ "../system_wrappers:metrics",
+ "utility:audio_frame_operations",
+ ]
+ absl_deps = [
+ "//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/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",
+ "../modules/audio_device:audio_device_api",
+ "../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:libjingle_peerconnection_api",
+ "../api:mock_audio_mixer",
+ "../api:mock_frame_decryptor",
+ "../api:mock_frame_encryptor",
+ "../api:scoped_refptr",
+ "../api/audio:audio_frame_api",
+ "../api/audio_codecs:audio_codecs_api",
+ "../api/audio_codecs:builtin_audio_encoder_factory",
+ "../api/audio_codecs/opus:audio_decoder_opus",
+ "../api/audio_codecs/opus:audio_encoder_opus",
+ "../api/crypto:frame_decryptor_interface",
+ "../api/rtc_event_log",
+ "../api/task_queue:default_task_queue_factory",
+ "../api/task_queue/test:mock_task_queue_base",
+ "../api/units:time_delta",
+ "../api/units:timestamp",
+ "../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_api",
+ "../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",
+ "../rtc_base:checks",
+ "../rtc_base:gunit_helpers",
+ "../rtc_base:macromagic",
+ "../rtc_base:refcount",
+ "../rtc_base:rtc_base_tests_utils",
+ "../rtc_base:safe_compare",
+ "../rtc_base:task_queue_for_test",
+ "../rtc_base:threading",
+ "../rtc_base:timeutils",
+ "../system_wrappers",
+ "../test:audio_codec_mocks",
+ "../test:field_trial",
+ "../test:mock_frame_transformer",
+ "../test:mock_transformable_frame",
+ "../test:mock_transport",
+ "../test:rtp_test_utils",
+ "../test:run_loop",
+ "../test:scoped_key_value_config",
+ "../test:test_common",
+ "../test:test_support",
+ "../test:video_test_constants",
+ "../test/time_controller:time_controller",
+ "utility:utility_tests",
+ "//testing/gtest",
+ ]
+ }
+
+ rtc_library("channel_receive_unittest") {
+ testonly = true
+ sources = [ "channel_receive_unittest.cc" ]
+ deps = [
+ ":audio",
+ "../api/audio_codecs:builtin_audio_decoder_factory",
+ "../api/crypto:frame_decryptor_interface",
+ "../api/task_queue:default_task_queue_factory",
+ "../logging:mocks",
+ "../modules/audio_device:audio_device_api",
+ "../modules/audio_device:mock_audio_device",
+ "../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:logging",
+ "../rtc_base:threading",
+ "../test:audio_codec_mocks",
+ "../test:mock_transport",
+ "../test:test_support",
+ "../test/time_controller",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+ }
+}