Delete unneeded includes of common_types.h and gn deps on webrtc_common.

Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
This commit is contained in:
Niels Möller 2018-11-20 15:09:53 +01:00 committed by Commit Bot
parent 38332cdcb1
commit 2222a80e79
56 changed files with 0 additions and 74 deletions

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@ -231,7 +231,6 @@ rtc_source_set("ortc_api") {
deps = [
":libjingle_peerconnection_api",
"..:webrtc_common",
"//third_party/abseil-cpp/absl/types:optional",
]
}

View file

@ -27,7 +27,6 @@
#include "api/array_view.h"
#include "api/rtcerror.h"
#include "api/video/encoded_image.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/networkroute.h"

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@ -18,7 +18,6 @@
#include "api/rtcerror.h"
#include "api/rtp_headers.h"
#include "api/rtpparameters.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {

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@ -18,7 +18,6 @@
#include "absl/types/optional.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"

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@ -45,7 +45,6 @@ rtc_static_library("audio") {
}
deps = [
"..:webrtc_common",
"../api:array_view",
"../api:call_api",
"../api:libjingle_peerconnection_api",

View file

@ -24,7 +24,6 @@
#include "audio/conversion.h"
#include "call/rtp_config.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"

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@ -26,7 +26,6 @@
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_audio/vad/include/vad.h"
#include "common_types.h" // NOLINT(build/include)
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"

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@ -25,7 +25,6 @@
#include "api/rtpreceiverinterface.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence

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@ -23,7 +23,6 @@
#include "api/crypto/frameencryptorinterface.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"

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@ -21,7 +21,6 @@ rtc_static_library("audio_frame_operations") {
]
deps = [
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",

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@ -196,7 +196,6 @@ rtc_static_library("call") {
":rtp_sender",
":simulated_network",
":video_stream_api",
"..:webrtc_common",
"../api:callfactory_api",
"../api:simulated_network_api",
"../api:transport_api",
@ -240,7 +239,6 @@ rtc_source_set("video_stream_api") {
]
deps = [
":rtp_interfaces",
"../:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/video:video_frame",
@ -290,7 +288,6 @@ rtc_source_set("fake_network") {
":call_interfaces",
":simulated_network",
":simulated_packet_receiver",
"..:webrtc_common",
"../api:simulated_network_api",
"../api:transport_api",
"../modules:module_api",
@ -330,7 +327,6 @@ if (rtc_include_tests) {
":rtp_receiver",
":rtp_sender",
":simulated_network",
"..:webrtc_common",
"../api:array_view",
"../api:fake_media_transport",
"../api:fake_media_transport",
@ -389,7 +385,6 @@ if (rtc_include_tests) {
":call_interfaces",
":simulated_network",
":video_stream_api",
"..:webrtc_common",
"../api:simulated_network_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video:builtin_video_bitrate_allocator_factory",

View file

@ -24,7 +24,6 @@
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {

View file

@ -24,7 +24,6 @@
#include "call/rtp_transport_controller_send_interface.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/network/sent_packet.h"

View file

@ -23,7 +23,6 @@
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/simulated_packet_receiver.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"

View file

@ -16,7 +16,6 @@
#include <vector>
#include "api/mediatypes.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/copyonwritebuffer.h"
namespace webrtc {

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@ -13,7 +13,6 @@
#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "call/rtp_rtcp_demuxer_helper.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/checks.h"
namespace webrtc {

View file

@ -16,7 +16,6 @@
#include "absl/memory/memory.h"
#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"

View file

@ -17,7 +17,6 @@
#include "absl/memory/memory.h"
#include "call/ssrc_binding_observer.h"
#include "call/test/mock_rtp_packet_sink_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"

View file

@ -16,7 +16,6 @@
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"

View file

@ -20,7 +20,6 @@
#include "call/rtp_bitrate_configurator.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
#include "modules/pacing/packet_router.h"
#include "modules/utility/include/process_thread.h"

View file

@ -23,7 +23,6 @@
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"

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@ -26,7 +26,6 @@
#include "api/video/video_timing.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

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@ -27,7 +27,6 @@
#include "api/video/video_stream_encoder_settings.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

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@ -47,7 +47,6 @@ rtc_static_library("common_audio") {
deps = [
":common_audio_c",
":sinc_resampler",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:gtest_prod",
"../rtc_base:rtc_base_approved",
@ -182,7 +181,6 @@ rtc_source_set("common_audio_c") {
deps = [
":common_audio_c_arm_asm",
":common_audio_cc",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:compile_assert_c",
"../rtc_base:rtc_base_approved",
@ -202,7 +200,6 @@ rtc_source_set("common_audio_cc") {
]
deps = [
"..:webrtc_common",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
]
@ -213,7 +210,6 @@ rtc_source_set("sinc_resampler") {
"resampler/sinc_resampler.h",
]
deps = [
"..:webrtc_common",
"../rtc_base:gtest_prod",
"../rtc_base:rtc_base_approved",
"../rtc_base/memory:aligned_malloc",
@ -382,7 +378,6 @@ if (rtc_include_tests) {
":fir_filter",
":fir_filter_factory",
":sinc_resampler",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",

View file

@ -747,7 +747,6 @@ if (is_linux || is_win) {
"peerconnection/server/utils.h",
]
deps = [
"..:webrtc_common",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_tools:command_line_parser",

View file

@ -54,7 +54,6 @@ rtc_source_set("rtc_stream_config") {
deps = [
":rtc_event_log_api",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
]
}
@ -301,7 +300,6 @@ if (rtc_enable_protobuf) {
":rtc_event_log_impl_encoder",
":rtc_event_log_proto",
":rtc_stream_config",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor",

View file

@ -18,7 +18,6 @@
#include "api/rtpparameters.h"
#include "api/video_codecs/sdp_video_format.h"
#include "common_types.h" // NOLINT(build/include)
#include "media/base/mediaconstants.h"
#include "rtc_base/system/rtc_export.h"

View file

@ -16,7 +16,6 @@
#include "absl/types/optional.h"
#include "api/video_codecs/sdp_video_format.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {

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@ -11,8 +11,6 @@
#ifndef MEDIA_ENGINE_ADM_HELPERS_H_
#define MEDIA_ENGINE_ADM_HELPERS_H_
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
class AudioDeviceModule;

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@ -14,7 +14,6 @@
#include "absl/strings/ascii.h"
#include "api/audio_codecs/audio_format.h"
#include "common_types.h" // NOLINT(build/include)
#include "media/base/mediaconstants.h"
namespace cricket {

View file

@ -18,7 +18,6 @@
#include "absl/strings/match.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
#include "common_types.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"

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@ -215,7 +215,6 @@ rtc_source_set("audio_device_impl") {
":audio_device_buffer",
":audio_device_default",
":audio_device_generic",
"../..:webrtc_common",
"../../api:array_view",
"../../common_audio",
"../../common_audio:common_audio_c",

View file

@ -37,7 +37,6 @@ rtc_static_library("audio_mixer_impl") {
deps = [
":audio_frame_manipulator",
"../..:webrtc_common",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../api/audio:audio_mixer_api",

View file

@ -29,7 +29,6 @@ rtc_static_library("primitives") {
]
deps = [
"../..:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/memory",
@ -97,7 +96,6 @@ if (rtc_include_tests) {
":desktop_capture",
":desktop_capture_mock",
":primitives",
"../..:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:cpu_features_api",
@ -419,7 +417,6 @@ rtc_static_library("desktop_capture_generic") {
deps = [
":primitives",
"../..:webrtc_common",
"../../api:refcountedbase",
"../../rtc_base:checks",
"../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.

View file

@ -29,7 +29,6 @@ rtc_static_library("pacing") {
deps = [
":interval_budget",
"..:module_api",
"../../:webrtc_common",
"../../api/transport:network_control",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
@ -56,7 +55,6 @@ rtc_source_set("interval_budget") {
]
deps = [
"../../:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]

View file

@ -26,7 +26,6 @@ rtc_static_library("utility") {
deps = [
"..:module_api",
"../..:webrtc_common",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",

View file

@ -33,7 +33,6 @@ rtc_static_library("system_wrappers") {
libs = []
deps = [
":cpu_features_api",
"..:webrtc_common",
"../api:array_view",
"../modules:module_api_public",
"../rtc_base:checks",
@ -90,9 +89,6 @@ rtc_source_set("cpu_features_api") {
sources = [
"include/cpu_features_wrapper.h",
]
deps = [
"..:webrtc_common",
]
}
rtc_source_set("field_trial") {

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@ -15,7 +15,6 @@
#include <memory>
#include <string>
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/atomicops.h"
#include "rtc_base/checks.h"

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@ -16,7 +16,6 @@
#include <memory>
#include "api/video_codecs/vp8_temporal_layers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/sleep.h"

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@ -15,7 +15,6 @@
#include <vector>
#include "api/video_codecs/video_encoder.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/criticalsection.h"
#include "rtc_base/sequenced_task_checker.h"
#include "rtc_base/task_queue.h"

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@ -12,7 +12,6 @@
#include "api/video_codecs/create_vp8_temporal_layers.h"
#include "api/video_codecs/vp8_temporal_layers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_error_codes.h"
#include "modules/video_coding/utility/simulcast_utility.h"

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@ -13,8 +13,6 @@
#include <set>
#include <string>
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
namespace test {

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@ -12,7 +12,6 @@
#include <string>
#include "common_types.h" // NOLINT(build/include)
#include "test/rtp_file_reader.h"
namespace webrtc {

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@ -37,7 +37,6 @@ if (rtc_include_tests) {
"../:test_common",
"../:test_support",
"../:video_test_common",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:transport_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",

View file

@ -10,7 +10,6 @@
#include <stdio.h>
#include "common_types.h" // NOLINT(build/include)
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

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@ -12,7 +12,6 @@
#include <memory>
#include "common_types.h" // NOLINT(build/include)
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/video_capture/video_capture.h"
#include "rtc_base/criticalsection.h"

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@ -53,7 +53,6 @@ rtc_static_library("video") {
deps = [
":frame_dumping_decoder",
"..:webrtc_common",
"../api:fec_controller_api",
"../api:libjingle_peerconnection_api",
"../api:transport_api",
@ -376,7 +375,6 @@ if (rtc_include_tests) {
"replay.cc",
]
deps = [
"..:webrtc_common",
"../api/test/video:function_video_factory",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",

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@ -18,7 +18,6 @@
#include "absl/types/optional.h"
#include "call/video_receive_stream.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/include/video_coding_defines.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/numerics/histogram_percentile_counter.h"

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@ -14,7 +14,6 @@
#include <map>
#include <vector>
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {

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@ -16,7 +16,6 @@
#include "absl/memory/memory.h"
#include "common_types.h" // NOLINT(build/include)
#include "media/base/mediaconstants.h"
#include "modules/pacing/packet_router.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"

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@ -16,7 +16,6 @@
#include <set>
#include "call/video_send_stream.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"

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@ -15,7 +15,6 @@
#include <limits>
#include <utility>
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

View file

@ -18,7 +18,6 @@
#include "api/video/video_stream_encoder_observer.h"
#include "call/video_send_stream.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "rtc_base/criticalsection.h"

View file

@ -13,7 +13,6 @@
#include <atomic>
#include "api/call/transport.h"
#include "common_types.h" // NOLINT(build/include)
namespace webrtc {
namespace internal {

View file

@ -21,7 +21,6 @@
#include "api/video_codecs/video_decoder_factory.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "call/rtx_receive_stream.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/h264/profile_level_id.h"
#include "common_video/include/incoming_video_stream.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"

View file

@ -18,7 +18,6 @@
#include "api/video/video_stream_encoder_interface.h"
#include "call/bitrate_allocator.h"
#include "call/rtp_video_sender_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/utility/include/process_thread.h"
#include "rtc_base/weak_ptr.h"
#include "video/call_stats.h"