Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 266 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/logging/BUILD.gn b/third_party/libwebrtc/webrtc/logging/BUILD.gn
new file mode 100644
index 0000000000..3acc7943d2
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/logging/BUILD.gn
@@ -0,0 +1,266 @@
+# Copyright (c) 2016 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 (!build_with_mozilla) {
+  import("//third_party/protobuf/proto_library.gni")
+}
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+group("logging") {
+  public_deps = [
+    ":rtc_event_log_impl",
+  ]
+  if (rtc_enable_protobuf) {
+    public_deps += [ ":rtc_event_log_parser" ]
+  }
+}
+
+rtc_source_set("rtc_event_log_api") {
+  sources = [
+    "rtc_event_log/events/rtc_event.h",
+    "rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
+    "rtc_event_log/events/rtc_event_audio_network_adaptation.h",
+    "rtc_event_log/events/rtc_event_audio_playout.cc",
+    "rtc_event_log/events/rtc_event_audio_playout.h",
+    "rtc_event_log/events/rtc_event_audio_receive_stream_config.cc",
+    "rtc_event_log/events/rtc_event_audio_receive_stream_config.h",
+    "rtc_event_log/events/rtc_event_audio_send_stream_config.cc",
+    "rtc_event_log/events/rtc_event_audio_send_stream_config.h",
+    "rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
+    "rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
+    "rtc_event_log/events/rtc_event_bwe_update_loss_based.cc",
+    "rtc_event_log/events/rtc_event_bwe_update_loss_based.h",
+    "rtc_event_log/events/rtc_event_logging_started.cc",
+    "rtc_event_log/events/rtc_event_logging_started.h",
+    "rtc_event_log/events/rtc_event_logging_stopped.cc",
+    "rtc_event_log/events/rtc_event_logging_stopped.h",
+    "rtc_event_log/events/rtc_event_probe_cluster_created.cc",
+    "rtc_event_log/events/rtc_event_probe_cluster_created.h",
+    "rtc_event_log/events/rtc_event_probe_result_failure.cc",
+    "rtc_event_log/events/rtc_event_probe_result_failure.h",
+    "rtc_event_log/events/rtc_event_probe_result_success.cc",
+    "rtc_event_log/events/rtc_event_probe_result_success.h",
+    "rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
+    "rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
+    "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc",
+    "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h",
+    "rtc_event_log/events/rtc_event_rtp_packet_incoming.cc",
+    "rtc_event_log/events/rtc_event_rtp_packet_incoming.h",
+    "rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc",
+    "rtc_event_log/events/rtc_event_rtp_packet_outgoing.h",
+    "rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
+    "rtc_event_log/events/rtc_event_video_receive_stream_config.h",
+    "rtc_event_log/events/rtc_event_video_send_stream_config.cc",
+    "rtc_event_log/events/rtc_event_video_send_stream_config.h",
+    "rtc_event_log/output/rtc_event_log_output_file.cc",
+    "rtc_event_log/output/rtc_event_log_output_file.h",
+    "rtc_event_log/rtc_event_log.h",
+    "rtc_event_log/rtc_event_log_factory_interface.h",
+    "rtc_event_log/rtc_stream_config.cc",
+    "rtc_event_log/rtc_stream_config.h",
+  ]
+
+  deps = [
+    "..:webrtc_common",
+    "../api:array_view",
+    "../call:video_stream_api",
+    "../modules/audio_coding:audio_network_adaptor_config",
+    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+    "../rtc_base:rtc_base_approved",
+    "../system_wrappers",
+  ]
+
+  if (!build_with_mozilla) {
+    deps += [
+      "../api:libjingle_logging_api",
+      "../api:libjingle_peerconnection_api"
+    ]
+  }
+
+  # TODO(eladalon): Remove this.
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+rtc_static_library("rtc_event_log_impl") {
+  sources = [
+    "rtc_event_log/encoder/rtc_event_log_encoder.h",
+    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
+    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
+    "rtc_event_log/rtc_event_log.cc",
+    "rtc_event_log/rtc_event_log_factory.cc",
+    "rtc_event_log/rtc_event_log_factory.h",
+  ]
+
+  defines = []
+
+  deps = [
+    ":rtc_event_log_api",
+    "..:webrtc_common",
+    "../modules/audio_coding:audio_network_adaptor",
+    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+    "../modules/rtp_rtcp",
+    "../rtc_base:protobuf_utils",
+    "../rtc_base:rtc_base_approved",
+    "../rtc_base:rtc_task_queue",
+    "../rtc_base:sequenced_task_checker",
+    "../system_wrappers",
+  ]
+
+  if (rtc_enable_protobuf) {
+    defines += [ "ENABLE_RTC_EVENT_LOG" ]
+    deps += [ ":rtc_event_log_proto" ]
+  }
+
+  # TODO(eladalon): Remove this.
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+if (rtc_enable_protobuf) {
+  proto_library("rtc_event_log_proto") {
+    sources = [
+      "rtc_event_log/rtc_event_log.proto",
+    ]
+    proto_out_dir = "logging/rtc_event_log"
+  }
+
+  rtc_static_library("rtc_event_log_parser") {
+    sources = [
+      "rtc_event_log/rtc_event_log_parser.cc",
+      "rtc_event_log/rtc_event_log_parser.h",
+    ]
+
+    public_deps = [
+      ":rtc_event_log_api",
+      ":rtc_event_log_proto",
+      "..:webrtc_common",
+      "../modules/audio_coding:audio_network_adaptor",
+      "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+      "../modules/rtp_rtcp:rtp_rtcp",
+      "../system_wrappers",
+    ]
+
+    if (!build_with_chromium && is_clang) {
+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+    deps = [
+      "../call:video_stream_api",
+      "../rtc_base:protobuf_utils",
+      "../rtc_base:rtc_base_approved",
+    ]
+  }
+
+  if (rtc_include_tests) {
+    rtc_source_set("rtc_event_log_tests") {
+      testonly = true
+      assert(rtc_enable_protobuf)
+      defines = [ "ENABLE_RTC_EVENT_LOG" ]
+      if (rtc_use_memcheck) {
+        defines += [ "WEBRTC_USE_MEMCHECK" ]
+      }
+      sources = [
+        "rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
+        "rtc_event_log/output/rtc_event_log_output_file_unittest.cc",
+        "rtc_event_log/rtc_event_log_unittest.cc",
+        "rtc_event_log/rtc_event_log_unittest_helper.cc",
+        "rtc_event_log/rtc_event_log_unittest_helper.h",
+      ]
+      deps = [
+        ":rtc_event_log_impl",
+        ":rtc_event_log_parser",
+        "../call",
+        "../modules/audio_coding:audio_network_adaptor",
+        "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+        "../modules/rtp_rtcp",
+        "../rtc_base:rtc_base_approved",
+        "../rtc_base:rtc_base_tests_utils",
+        "../system_wrappers:metrics_default",
+        "../test:test_support",
+        "//testing/gmock",
+        "//testing/gtest",
+      ]
+      if (!build_with_chromium && is_clang) {
+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+      }
+    }
+    rtc_test("rtc_event_log2rtp_dump") {
+      testonly = true
+      sources = [
+        "rtc_event_log/rtc_event_log2rtp_dump.cc",
+      ]
+      deps = [
+        ":rtc_event_log_api",
+        ":rtc_event_log_impl",
+        ":rtc_event_log_parser",
+        "../modules/rtp_rtcp:rtp_rtcp",
+        "../rtc_base:rtc_base_approved",
+        "../system_wrappers:field_trial_default",
+        "../system_wrappers:metrics_default",
+        "../test:rtp_test_utils",
+      ]
+      if (!build_with_chromium && is_clang) {
+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+      }
+    }
+  }
+  if (rtc_include_tests) {
+    rtc_executable("rtc_event_log2text") {
+      testonly = true
+      sources = [
+        "rtc_event_log/rtc_event_log2text.cc",
+      ]
+      deps = [
+        ":rtc_event_log_api",
+        ":rtc_event_log_impl",
+        ":rtc_event_log_parser",
+        "../call:video_stream_api",
+        "../rtc_base:rtc_base_approved",
+
+        # TODO(kwiberg): Remove this dependency.
+        "../api/audio_codecs:audio_codecs_api",
+        "../modules/audio_coding:audio_network_adaptor_config",
+        "../modules/rtp_rtcp:rtp_rtcp",
+      ]
+      if (!build_with_chromium && is_clang) {
+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+      }
+    }
+  }
+  if (rtc_include_tests) {
+    rtc_executable("rtc_event_log2stats") {
+      testonly = true
+      sources = [
+        "rtc_event_log/rtc_event_log2stats.cc",
+      ]
+      deps = [
+        ":rtc_event_log_api",
+        ":rtc_event_log_impl",
+        ":rtc_event_log_proto",
+        "../rtc_base:rtc_base_approved",
+      ]
+      if (!build_with_chromium && is_clang) {
+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+      }
+    }
+  }
+}