webrtc_m130/webrtc/voice_engine/BUILD.gn

# Copyright (c) 2014 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")

rtc_static_library("audio_coder") {
  sources = [
    "coder.cc",
    "coder.h",
  ]
  deps = [
    "..:webrtc_common",
    "../api/audio_codecs:builtin_audio_decoder_factory",
    "../modules:module_api",
    "../modules/audio_coding",
    "../modules/audio_coding:audio_encoder_factory_interface",
    "../modules/audio_coding:audio_format_conversion",
    "../modules/audio_coding:builtin_audio_encoder_factory",
    "../modules/audio_coding:rent_a_codec",
  ]

  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("file_player") {
  sources = [
    "file_player.cc",
    "file_player.h",
  ]
  deps = [
    ":audio_coder",
    "..:webrtc_common",
    "../base:rtc_base_approved",
    "../common_audio",
    "../modules:module_api",
    "../modules/media_file",
  ]

  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("file_recorder") {
  sources = [
    "file_recorder.cc",
    "file_recorder.h",
  ]
  deps = [
    ":audio_coder",
    "..:webrtc_common",
    "../base:rtc_base_approved",
    "../common_audio",
    "../modules:module_api",
    "../modules/media_file:media_file",
    "../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" ]
  }
}

rtc_static_library("voice_engine") {
  sources = [
    "channel.cc",
    "channel.h",
    "channel_manager.cc",
    "channel_manager.h",
    "channel_proxy.cc",
    "channel_proxy.h",
    "include/voe_base.h",
    "include/voe_codec.h",
    "include/voe_errors.h",
    "include/voe_file.h",
    "include/voe_network.h",
    "include/voe_rtp_rtcp.h",
    "monitor_module.h",
    "output_mixer.cc",
    "output_mixer.h",
    "shared_data.cc",
    "shared_data.h",
    "statistics.cc",
    "statistics.h",
    "transmit_mixer.cc",
    "transmit_mixer.h",
    "transport_feedback_packet_loss_tracker.cc",
    "transport_feedback_packet_loss_tracker.h",
    "utility.cc",
    "utility.h",
    "voe_base_impl.cc",
    "voe_base_impl.h",
    "voe_codec_impl.cc",
    "voe_codec_impl.h",
    "voe_file_impl.cc",
    "voe_file_impl.h",
    "voe_network_impl.cc",
    "voe_network_impl.h",
    "voe_rtp_rtcp_impl.cc",
    "voe_rtp_rtcp_impl.h",
    "voice_engine_defines.h",
    "voice_engine_impl.cc",
    "voice_engine_impl.h",
  ]

  if (is_win) {
    defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]

    cflags = [
      # TODO(kjellander): Bug 261: fix this warning.
      "/wd4373",  # Virtual function override.
    ]
  }

  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" ]
  }

  public_deps = [
    "../modules/audio_coding",
  ]
  deps = [
    ":audio_level",
    ":file_player",
    ":file_recorder",
    "..:webrtc_common",
    "../api:audio_mixer_api",
    "../api:call_api",
    "../api:libjingle_peerconnection_api",
    "../api:transport_api",
    "../api/audio_codecs:audio_codecs_api",
    "../api/audio_codecs:builtin_audio_decoder_factory",
    "../audio/utility:audio_frame_operations",
    "../base:rtc_base_approved",
    "../base:rtc_task_queue",
    "../modules:module_api",

    # TODO(nisse): Delete when declaration of RtpTransportController
    # and related interfaces move to api/.
    "../call:call_interfaces",
    "../common_audio",
    "../logging:rtc_event_log_api",
    "../modules/audio_coding:audio_encoder_interface",
    "../modules/audio_coding:audio_format_conversion",
    "../modules/audio_coding:rent_a_codec",
    "../modules/audio_conference_mixer",
    "../modules/audio_device",
    "../modules/audio_processing",
    "../modules/bitrate_controller",
    "../modules/media_file",
    "../modules/pacing",
    "../modules/rtp_rtcp",
    "../modules/utility",
    "../system_wrappers",
  ]
}

rtc_static_library("audio_level") {
  sources = [
    "audio_level.cc",
    "audio_level.h",
  ]

  deps = [
    "..:webrtc_common",
    "../base:rtc_base_approved",
    "../common_audio",
    "../modules:module_api",
  ]
}

if (rtc_include_tests) {
  rtc_test("voice_engine_unittests") {
    deps = [
      ":file_player",
      ":voice_engine",
      "../base:rtc_base_approved",
      "../base:rtc_base_tests_utils",
      "../modules:module_api",
      "../test:test_common",
      "//testing/gmock",
      "//testing/gtest",
      "//third_party/gflags",
      "//webrtc/common_audio",
      "//webrtc/modules/audio_coding",
      "//webrtc/modules/audio_conference_mixer",
      "//webrtc/modules/audio_device",
      "//webrtc/modules/audio_processing",
      "//webrtc/modules/media_file",
      "//webrtc/modules/rtp_rtcp",
      "//webrtc/modules/utility",
      "//webrtc/modules/video_capture:video_capture",
      "//webrtc/system_wrappers",
      "//webrtc/test:test_main",
      "//webrtc/test:video_test_common",
    ]

    if (is_android) {
      deps += [ "//testing/android/native_test:native_test_native_code" ]
      shard_timeout = 900
    }

    sources = [
      "channel_unittest.cc",
      "file_player_unittests.cc",
      "transport_feedback_packet_loss_tracker_unittest.cc",
      "utility_unittest.cc",
      "voe_base_unittest.cc",
      "voe_codec_unittest.cc",
      "voe_network_unittest.cc",
      "voice_engine_fixture.cc",
      "voice_engine_fixture.h",
    ]

    data = [
      "//resources/utility/encapsulated_pcm16b_8khz.wav",
      "//resources/utility/encapsulated_pcmu_8khz.wav",
    ]

    if (is_win) {
      defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]

      cflags = [
        # TODO(kjellander): Bug 261: fix this warning.
        "/wd4373",  # Virtual function override.
      ]
    }

    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 (!is_ios) {
    rtc_executable("voe_auto_test") {
      testonly = true

      deps = [
        ":voice_engine",
        "..:webrtc_common",
        "../base:rtc_base_approved",
        "../modules:module_api",
        "../modules/audio_device:audio_device",
        "../modules/audio_processing:audio_processing",
        "../modules/rtp_rtcp:rtp_rtcp",
        "//testing/gmock",
        "//testing/gtest",
        "//third_party/gflags",
        "//webrtc/logging:rtc_event_log_api",
        "//webrtc/modules/video_capture",
        "//webrtc/system_wrappers",
        "//webrtc/system_wrappers/:system_wrappers_default",
        "//webrtc/test/:test_common",
        "//webrtc/test/:test_support",
        "//webrtc/test/:video_test_common",
      ]

      sources = [
        "test/auto_test/automated_mode.cc",
        "test/auto_test/fakes/conference_transport.cc",
        "test/auto_test/fakes/conference_transport.h",
        "test/auto_test/fakes/loudest_filter.cc",
        "test/auto_test/fakes/loudest_filter.h",
        "test/auto_test/fixtures/after_initialization_fixture.cc",
        "test/auto_test/fixtures/after_initialization_fixture.h",
        "test/auto_test/fixtures/after_streaming_fixture.cc",
        "test/auto_test/fixtures/after_streaming_fixture.h",
        "test/auto_test/fixtures/before_initialization_fixture.cc",
        "test/auto_test/fixtures/before_initialization_fixture.h",
        "test/auto_test/fixtures/before_streaming_fixture.cc",
        "test/auto_test/fixtures/before_streaming_fixture.h",
        "test/auto_test/standard/codec_before_streaming_test.cc",
        "test/auto_test/standard/codec_test.cc",
        "test/auto_test/standard/dtmf_test.cc",
        "test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
        "test/auto_test/standard/rtp_rtcp_extensions.cc",
        "test/auto_test/standard/rtp_rtcp_test.cc",
        "test/auto_test/voe_conference_test.cc",
        "test/auto_test/voe_standard_test.cc",
        "test/auto_test/voe_standard_test.h",
        "test/auto_test/voe_test_defines.h",
      ]

      defines = []

      if (rtc_enable_protobuf) {
        defines = [ "ENABLE_RTC_EVENT_LOG" ]
      }

      if (is_win) {
        defines += [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]

        cflags = [
          "/wd4267",  # size_t to int truncation.
          "/wd4373",  # Virtual function override.
                      # TODO(kjellander): Bug 261: fix this warning.
        ]
      }

      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" ]
      }
    }
  }
}