mirror of
https://github.com/klzgrad/naiveproxy.git
synced 2024-11-28 08:16:09 +03:00
435 lines
14 KiB
Plaintext
435 lines
14 KiB
Plaintext
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# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../webrtc.gni")
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rtc_source_set("rtp_rtcp_format") {
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public = [
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"include/rtp_cvo.h",
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"include/rtp_header_extension_map.h",
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"include/rtp_rtcp_defines.h",
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"source/byte_io.h",
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"source/rtcp_packet.h",
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"source/rtcp_packet/app.h",
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"source/rtcp_packet/bye.h",
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"source/rtcp_packet/common_header.h",
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"source/rtcp_packet/compound_packet.h",
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"source/rtcp_packet/dlrr.h",
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"source/rtcp_packet/extended_jitter_report.h",
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"source/rtcp_packet/extended_reports.h",
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"source/rtcp_packet/fir.h",
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"source/rtcp_packet/nack.h",
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"source/rtcp_packet/pli.h",
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"source/rtcp_packet/psfb.h",
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"source/rtcp_packet/rapid_resync_request.h",
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"source/rtcp_packet/receiver_report.h",
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"source/rtcp_packet/remb.h",
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"source/rtcp_packet/report_block.h",
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"source/rtcp_packet/rrtr.h",
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"source/rtcp_packet/rtpfb.h",
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"source/rtcp_packet/sdes.h",
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"source/rtcp_packet/sender_report.h",
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"source/rtcp_packet/target_bitrate.h",
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"source/rtcp_packet/tmmb_item.h",
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"source/rtcp_packet/tmmbn.h",
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"source/rtcp_packet/tmmbr.h",
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"source/rtcp_packet/transport_feedback.h",
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"source/rtcp_packet/voip_metric.h",
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"source/rtp_header_extensions.h",
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"source/rtp_packet.h",
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"source/rtp_packet_received.h",
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"source/rtp_packet_to_send.h",
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]
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sources = [
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"include/rtp_rtcp_defines.cc",
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"source/rtcp_packet.cc",
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"source/rtcp_packet/app.cc",
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"source/rtcp_packet/bye.cc",
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"source/rtcp_packet/common_header.cc",
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"source/rtcp_packet/compound_packet.cc",
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"source/rtcp_packet/dlrr.cc",
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"source/rtcp_packet/extended_jitter_report.cc",
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"source/rtcp_packet/extended_reports.cc",
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"source/rtcp_packet/fir.cc",
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"source/rtcp_packet/nack.cc",
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"source/rtcp_packet/pli.cc",
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"source/rtcp_packet/psfb.cc",
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"source/rtcp_packet/rapid_resync_request.cc",
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"source/rtcp_packet/receiver_report.cc",
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"source/rtcp_packet/remb.cc",
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"source/rtcp_packet/report_block.cc",
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"source/rtcp_packet/rrtr.cc",
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"source/rtcp_packet/rtpfb.cc",
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"source/rtcp_packet/sdes.cc",
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"source/rtcp_packet/sender_report.cc",
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"source/rtcp_packet/target_bitrate.cc",
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"source/rtcp_packet/tmmb_item.cc",
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"source/rtcp_packet/tmmbn.cc",
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"source/rtcp_packet/tmmbr.cc",
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"source/rtcp_packet/transport_feedback.cc",
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"source/rtcp_packet/voip_metric.cc",
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"source/rtp_header_extension_map.cc",
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"source/rtp_header_extensions.cc",
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"source/rtp_packet.cc",
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"source/rtp_packet_received.cc",
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"source/rtp_packet_to_send.cc",
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]
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deps = [
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"..:module_api",
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"../..:webrtc_common",
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"../../:typedefs",
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"../../api:array_view",
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"../../api:libjingle_peerconnection_api",
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"../../api:optional",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/video:video_frame",
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"../../common_video",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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"../../system_wrappers",
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]
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}
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rtc_static_library("rtp_rtcp") {
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visibility = [ "*" ]
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sources = [
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"include/flexfec_receiver.h",
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"include/flexfec_sender.h",
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"include/receive_statistics.h",
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"include/remote_ntp_time_estimator.h",
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"include/rtp_header_parser.h",
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"include/rtp_payload_registry.h",
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"include/rtp_receiver.h",
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"include/rtp_rtcp.h",
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"include/ulpfec_receiver.h",
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"source/dtmf_queue.cc",
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"source/dtmf_queue.h",
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"source/fec_private_tables_bursty.cc",
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"source/fec_private_tables_bursty.h",
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"source/fec_private_tables_random.cc",
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"source/fec_private_tables_random.h",
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"source/flexfec_header_reader_writer.cc",
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"source/flexfec_header_reader_writer.h",
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"source/flexfec_receiver.cc",
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"source/flexfec_sender.cc",
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"source/forward_error_correction.cc",
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"source/forward_error_correction.h",
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"source/forward_error_correction_internal.cc",
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"source/forward_error_correction_internal.h",
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"source/packet_loss_stats.cc",
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"source/packet_loss_stats.h",
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"source/playout_delay_oracle.cc",
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"source/playout_delay_oracle.h",
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"source/receive_statistics_impl.cc",
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"source/receive_statistics_impl.h",
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"source/remote_ntp_time_estimator.cc",
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"source/rtcp_nack_stats.cc",
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"source/rtcp_nack_stats.h",
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"source/rtcp_receiver.cc",
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"source/rtcp_receiver.h",
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"source/rtcp_sender.cc",
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"source/rtcp_sender.h",
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"source/rtp_format.cc",
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"source/rtp_format.h",
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"source/rtp_format_h264.cc",
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"source/rtp_format_h264.h",
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"source/rtp_format_video_generic.cc",
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"source/rtp_format_video_generic.h",
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"source/rtp_format_vp8.cc",
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"source/rtp_format_vp8.h",
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"source/rtp_format_vp9.cc",
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"source/rtp_format_vp9.h",
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"source/rtp_header_parser.cc",
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"source/rtp_packet_history.cc",
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"source/rtp_packet_history.h",
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"source/rtp_payload_registry.cc",
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"source/rtp_receiver_audio.cc",
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"source/rtp_receiver_audio.h",
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"source/rtp_receiver_impl.cc",
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"source/rtp_receiver_impl.h",
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"source/rtp_receiver_strategy.cc",
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"source/rtp_receiver_strategy.h",
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"source/rtp_receiver_video.cc",
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"source/rtp_receiver_video.h",
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"source/rtp_rtcp_config.h",
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"source/rtp_rtcp_impl.cc",
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"source/rtp_rtcp_impl.h",
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"source/rtp_sender.cc",
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"source/rtp_sender.h",
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"source/rtp_sender_audio.cc",
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"source/rtp_sender_audio.h",
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"source/rtp_sender_video.cc",
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"source/rtp_sender_video.h",
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"source/rtp_utility.cc",
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"source/rtp_utility.h",
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"source/time_util.cc",
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"source/time_util.h",
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"source/tmmbr_help.cc",
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"source/tmmbr_help.h",
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"source/ulpfec_generator.cc",
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"source/ulpfec_generator.h",
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"source/ulpfec_header_reader_writer.cc",
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"source/ulpfec_header_reader_writer.h",
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"source/ulpfec_receiver_impl.cc",
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"source/ulpfec_receiver_impl.h",
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]
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if (rtc_enable_bwe_test_logging) {
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defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
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} else {
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defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
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}
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deps = [
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":rtp_rtcp_format",
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"..:module_api",
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"../..:webrtc_common",
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"../../:typedefs",
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"../../api:array_view",
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"../../api:libjingle_peerconnection_api",
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"../../api:optional",
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"../../api:transport_api",
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"../../api/audio_codecs:audio_codecs_api",
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"../../api/video:video_bitrate_allocation",
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"../../common_video",
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"../../logging:rtc_event_audio",
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"../../logging:rtc_event_log_api",
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"../../logging:rtc_event_rtp_rtcp",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:gtest_prod",
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"../../rtc_base:rate_limiter",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:rtc_numerics",
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"../../rtc_base:safe_minmax",
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"../../rtc_base:sequenced_task_checker",
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"../../rtc_base:stringutils",
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"../../rtc_base/system:fallthrough",
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"../../rtc_base/time:timestamp_extrapolator",
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"../../system_wrappers",
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"../../system_wrappers:field_trial_api",
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"../../system_wrappers:metrics_api",
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"../audio_coding:audio_format_conversion",
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"../remote_bitrate_estimator",
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]
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# TODO(jschuh): Bug 1348: fix this warning.
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configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
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}
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rtc_source_set("rtcp_transceiver") {
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visibility = [ "*" ]
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public = [
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"source/rtcp_transceiver.h",
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"source/rtcp_transceiver_config.h",
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"source/rtcp_transceiver_impl.h",
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]
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sources = [
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"source/rtcp_transceiver.cc",
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"source/rtcp_transceiver_config.cc",
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"source/rtcp_transceiver_impl.cc",
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]
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deps = [
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":rtp_rtcp",
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":rtp_rtcp_format",
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"../../:webrtc_common",
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"../../api:array_view",
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"../../api:libjingle_peerconnection_api",
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"../../api:optional",
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"../../api:transport_api",
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"../../api/video:video_bitrate_allocation",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:rtc_task_queue",
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"../../rtc_base:weak_ptr",
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"../../system_wrappers",
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]
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}
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rtc_source_set("fec_test_helper") {
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testonly = true
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sources = [
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"source/fec_test_helper.cc",
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"source/fec_test_helper.h",
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]
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deps = [
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":rtp_rtcp",
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":rtp_rtcp_format",
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"..:module_api",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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]
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# TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
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configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
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}
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rtc_source_set("mock_rtp_rtcp") {
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testonly = true
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sources = [
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"mocks/mock_recovered_packet_receiver.cc",
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"mocks/mock_rtcp_bandwidth_observer.cc",
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"mocks/mock_rtcp_rtt_stats.cc",
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"mocks/mock_rtp_rtcp.cc",
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]
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public = [
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"mocks/mock_recovered_packet_receiver.h",
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"mocks/mock_rtcp_bandwidth_observer.h",
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"mocks/mock_rtcp_rtt_stats.h",
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"mocks/mock_rtp_rtcp.h",
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]
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deps = [
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":rtp_rtcp",
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":rtp_rtcp_format",
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"..:module_api",
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"../../api:optional",
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"../../api/video:video_bitrate_allocation",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../test:test_support",
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]
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}
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if (rtc_include_tests) {
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rtc_executable("test_packet_masks_metrics") {
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testonly = true
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sources = [
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"test/testFec/average_residual_loss_xor_codes.h",
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"test/testFec/test_packet_masks_metrics.cc",
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]
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deps = [
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":rtp_rtcp",
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"../../test:fileutils",
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"../../test:test_main",
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"//testing/gtest",
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]
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} # test_packet_masks_metrics
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||
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|
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rtc_source_set("rtp_rtcp_modules_tests") {
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testonly = true
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sources = [
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"test/testFec/test_fec.cc",
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]
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deps = [
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":rtp_rtcp",
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":rtp_rtcp_format",
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"../../rtc_base:rtc_base_approved",
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"../../test:fileutils",
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"../../test:test_support",
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]
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}
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||
|
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rtc_source_set("rtp_rtcp_unittests") {
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testonly = true
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||
|
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|
||
|
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sources = [
|
||
|
|
"source/byte_io_unittest.cc",
|
||
|
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"source/fec_private_tables_bursty_unittest.cc",
|
||
|
|
"source/flexfec_header_reader_writer_unittest.cc",
|
||
|
|
"source/flexfec_receiver_unittest.cc",
|
||
|
|
"source/flexfec_sender_unittest.cc",
|
||
|
|
"source/nack_rtx_unittest.cc",
|
||
|
|
"source/packet_loss_stats_unittest.cc",
|
||
|
|
"source/playout_delay_oracle_unittest.cc",
|
||
|
|
"source/receive_statistics_unittest.cc",
|
||
|
|
"source/remote_ntp_time_estimator_unittest.cc",
|
||
|
|
"source/rtcp_nack_stats_unittest.cc",
|
||
|
|
"source/rtcp_packet/app_unittest.cc",
|
||
|
|
"source/rtcp_packet/bye_unittest.cc",
|
||
|
|
"source/rtcp_packet/common_header_unittest.cc",
|
||
|
|
"source/rtcp_packet/compound_packet_unittest.cc",
|
||
|
|
"source/rtcp_packet/dlrr_unittest.cc",
|
||
|
|
"source/rtcp_packet/extended_jitter_report_unittest.cc",
|
||
|
|
"source/rtcp_packet/extended_reports_unittest.cc",
|
||
|
|
"source/rtcp_packet/fir_unittest.cc",
|
||
|
|
"source/rtcp_packet/nack_unittest.cc",
|
||
|
|
"source/rtcp_packet/pli_unittest.cc",
|
||
|
|
"source/rtcp_packet/rapid_resync_request_unittest.cc",
|
||
|
|
"source/rtcp_packet/receiver_report_unittest.cc",
|
||
|
|
"source/rtcp_packet/remb_unittest.cc",
|
||
|
|
"source/rtcp_packet/report_block_unittest.cc",
|
||
|
|
"source/rtcp_packet/rrtr_unittest.cc",
|
||
|
|
"source/rtcp_packet/sdes_unittest.cc",
|
||
|
|
"source/rtcp_packet/sender_report_unittest.cc",
|
||
|
|
"source/rtcp_packet/target_bitrate_unittest.cc",
|
||
|
|
"source/rtcp_packet/tmmbn_unittest.cc",
|
||
|
|
"source/rtcp_packet/tmmbr_unittest.cc",
|
||
|
|
"source/rtcp_packet/transport_feedback_unittest.cc",
|
||
|
|
"source/rtcp_packet/voip_metric_unittest.cc",
|
||
|
|
"source/rtcp_packet_unittest.cc",
|
||
|
|
"source/rtcp_receiver_unittest.cc",
|
||
|
|
"source/rtcp_sender_unittest.cc",
|
||
|
|
"source/rtcp_transceiver_impl_unittest.cc",
|
||
|
|
"source/rtcp_transceiver_unittest.cc",
|
||
|
|
"source/rtp_fec_unittest.cc",
|
||
|
|
"source/rtp_format_h264_unittest.cc",
|
||
|
|
"source/rtp_format_video_generic_unittest.cc",
|
||
|
|
"source/rtp_format_vp8_test_helper.cc",
|
||
|
|
"source/rtp_format_vp8_test_helper.h",
|
||
|
|
"source/rtp_format_vp8_unittest.cc",
|
||
|
|
"source/rtp_format_vp9_unittest.cc",
|
||
|
|
"source/rtp_header_extension_map_unittest.cc",
|
||
|
|
"source/rtp_packet_history_unittest.cc",
|
||
|
|
"source/rtp_packet_unittest.cc",
|
||
|
|
"source/rtp_payload_registry_unittest.cc",
|
||
|
|
"source/rtp_receiver_unittest.cc",
|
||
|
|
"source/rtp_rtcp_impl_unittest.cc",
|
||
|
|
"source/rtp_sender_unittest.cc",
|
||
|
|
"source/rtp_utility_unittest.cc",
|
||
|
|
"source/time_util_unittest.cc",
|
||
|
|
"source/ulpfec_generator_unittest.cc",
|
||
|
|
"source/ulpfec_header_reader_writer_unittest.cc",
|
||
|
|
"source/ulpfec_receiver_unittest.cc",
|
||
|
|
"test/testAPI/test_api.cc",
|
||
|
|
"test/testAPI/test_api.h",
|
||
|
|
"test/testAPI/test_api_audio.cc",
|
||
|
|
"test/testAPI/test_api_rtcp.cc",
|
||
|
|
"test/testAPI/test_api_video.cc",
|
||
|
|
]
|
||
|
|
deps = [
|
||
|
|
":fec_test_helper",
|
||
|
|
":mock_rtp_rtcp",
|
||
|
|
":rtcp_transceiver",
|
||
|
|
":rtp_rtcp",
|
||
|
|
":rtp_rtcp_format",
|
||
|
|
"..:module_api",
|
||
|
|
"../..:webrtc_common",
|
||
|
|
"../../:typedefs",
|
||
|
|
"../../api:array_view",
|
||
|
|
"../../api:libjingle_peerconnection_api",
|
||
|
|
"../../api:optional",
|
||
|
|
"../../api:transport_api",
|
||
|
|
"../../api/video:video_bitrate_allocation",
|
||
|
|
"../../api/video:video_frame",
|
||
|
|
"../../call:rtp_receiver",
|
||
|
|
"../../common_video:common_video",
|
||
|
|
"../../logging:mocks",
|
||
|
|
"../../logging:rtc_event_log_api",
|
||
|
|
"../../rtc_base:checks",
|
||
|
|
"../../rtc_base:rate_limiter",
|
||
|
|
"../../rtc_base:rtc_base_approved",
|
||
|
|
"../../rtc_base:rtc_base_tests_utils",
|
||
|
|
"../../rtc_base:rtc_task_queue",
|
||
|
|
"../../system_wrappers",
|
||
|
|
"../../test:field_trial",
|
||
|
|
"../../test:rtp_test_utils",
|
||
|
|
"../../test:test_common",
|
||
|
|
"../../test:test_support",
|
||
|
|
"../audio_coding:audio_format_conversion",
|
||
|
|
]
|
||
|
|
|
||
|
|
# TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
|
||
|
|
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
|
||
|
|
}
|
||
|
|
}
|