Move deprecated code to their own build targets.
Moves the deprecated version of RtpRtcp module, and related classes in video/. Bug: webrtc:11581 Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33025}
This commit is contained in:
Niels Möller
Commit Bot
parent
844c759766
commit
3e9cb2cbf2
@ -427,6 +427,7 @@ if (rtc_include_tests) {
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../audio",
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"../modules:module_api",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer",
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"../modules/audio_mixer:audio_mixer_impl",
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@ -30,6 +30,7 @@
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#include "call/audio_state.h"
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#include "modules/audio_device/include/mock_audio_device.h"
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#include "modules/audio_processing/include/mock_audio_processing.h"
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#include "modules/include/module.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
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#include "test/fake_encoder.h"
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#include "test/gtest.h"
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@ -135,7 +135,6 @@ rtc_library("rtp_rtcp") {
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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_rtcp.h", # deprecated
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"include/ulpfec_receiver.h",
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"source/absolute_capture_time_receiver.cc",
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"source/absolute_capture_time_receiver.h",
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@ -145,8 +144,6 @@ rtc_library("rtp_rtcp") {
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"source/active_decode_targets_helper.h",
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"source/create_video_rtp_depacketizer.cc",
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"source/create_video_rtp_depacketizer.h",
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"source/deprecated/deprecated_rtp_sender_egress.cc",
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"source/deprecated/deprecated_rtp_sender_egress.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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@ -191,8 +188,6 @@ rtc_library("rtp_rtcp") {
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"source/rtp_packetizer_av1.cc",
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"source/rtp_packetizer_av1.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_rtcp_impl2.cc",
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"source/rtp_rtcp_impl2.h",
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"source/rtp_rtcp_interface.h",
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@ -248,7 +243,6 @@ rtc_library("rtp_rtcp") {
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deps = [
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":rtp_rtcp_format",
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":rtp_video_header",
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"..:module_api",
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"..:module_api_public",
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"..:module_fec_api",
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"../../api:array_view",
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@ -319,8 +313,36 @@ rtc_library("rtp_rtcp") {
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}
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rtc_source_set("rtp_rtcp_legacy") {
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# TODO(bugs.webrtc.org/11581): The files "source/rtp_rtcp_impl.cc"
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# and "source/rtp_rtcp_impl.h" should be moved to this target.
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sources = [
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"include/rtp_rtcp.h",
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"source/deprecated/deprecated_rtp_sender_egress.cc",
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"source/deprecated/deprecated_rtp_sender_egress.h",
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"source/rtp_rtcp_impl.cc",
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"source/rtp_rtcp_impl.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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"..:module_fec_api",
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"../../api:rtp_headers",
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"../../api:transport_api",
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"../../api/rtc_event_log",
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"../../api/transport:field_trial_based_config",
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"../../api/units:data_rate",
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"../../api/video:video_bitrate_allocation",
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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:rtc_base_approved",
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"../../rtc_base/synchronization:mutex",
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"../remote_bitrate_estimator",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("rtcp_transceiver") {
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@ -17,7 +17,6 @@
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#include "absl/types/optional.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "modules/include/module.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
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@ -12,8 +12,6 @@ rtc_library("video") {
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sources = [
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"buffered_frame_decryptor.cc",
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"buffered_frame_decryptor.h",
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"call_stats.cc",
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"call_stats.h",
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"call_stats2.cc",
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"call_stats2.h",
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"encoder_rtcp_feedback.cc",
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@ -22,18 +20,12 @@ rtc_library("video") {
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"quality_limitation_reason_tracker.h",
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"quality_threshold.cc",
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"quality_threshold.h",
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"receive_statistics_proxy.cc",
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"receive_statistics_proxy.h",
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"receive_statistics_proxy2.cc",
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"receive_statistics_proxy2.h",
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"report_block_stats.cc",
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"report_block_stats.h",
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"rtp_streams_synchronizer.cc",
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"rtp_streams_synchronizer.h",
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"rtp_streams_synchronizer2.cc",
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"rtp_streams_synchronizer2.h",
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"rtp_video_stream_receiver.cc",
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"rtp_video_stream_receiver.h",
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"rtp_video_stream_receiver2.cc",
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"rtp_video_stream_receiver2.h",
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"rtp_video_stream_receiver_frame_transformer_delegate.cc",
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@ -48,20 +40,14 @@ rtc_library("video") {
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"stream_synchronization.h",
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"transport_adapter.cc",
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"transport_adapter.h",
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"video_quality_observer.cc",
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"video_quality_observer.h",
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"video_quality_observer2.cc",
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"video_quality_observer2.h",
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"video_receive_stream.cc",
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"video_receive_stream.h",
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"video_receive_stream2.cc",
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"video_receive_stream2.h",
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"video_send_stream.cc",
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"video_send_stream.h",
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"video_send_stream_impl.cc",
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"video_send_stream_impl.h",
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"video_stream_decoder.cc",
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"video_stream_decoder.h",
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"video_stream_decoder2.cc",
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"video_stream_decoder2.h",
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]
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@ -111,7 +97,6 @@ rtc_library("video") {
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"../modules/video_coding:nack_module",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../modules/video_coding/deprecated:nack_module",
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"../modules/video_processing",
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"../rtc_base:checks",
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"../rtc_base:rate_limiter",
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@ -155,22 +140,73 @@ rtc_library("video") {
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}
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rtc_source_set("video_legacy") {
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# TODO(bugs.webrtc.org/11581): These files should be moved to this target:
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#
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# "call_stats.cc",
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# "call_stats.h",
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# "receive_statistics_proxy.cc",
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# "receive_statistics_proxy.h",
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# "rtp_streams_synchronizer.cc",
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# "rtp_streams_synchronizer.h",
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# "rtp_video_stream_receiver.cc",
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# "rtp_video_stream_receiver.h",
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# "video_quality_observer.cc",
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# "video_quality_observer.h",
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# "video_receive_stream.cc",
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# "video_receive_stream.h",
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# "video_stream_decoder.cc",
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# "video_stream_decoder.h",
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sources = [
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"call_stats.cc",
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"call_stats.h",
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"receive_statistics_proxy.cc",
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"receive_statistics_proxy.h",
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"rtp_streams_synchronizer.cc",
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"rtp_streams_synchronizer.h",
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"rtp_video_stream_receiver.cc",
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"rtp_video_stream_receiver.h",
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"video_quality_observer.cc",
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"video_quality_observer.h",
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"video_receive_stream.cc",
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"video_receive_stream.h",
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"video_stream_decoder.cc",
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"video_stream_decoder.h",
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]
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deps = [
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":frame_dumping_decoder",
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":video",
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"../api:array_view",
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"../api:scoped_refptr",
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"../api/crypto:frame_decryptor_interface",
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"../api/task_queue",
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"../api/video:encoded_image",
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"../api/video:recordable_encoded_frame",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:rtp_interfaces",
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"../call:rtp_receiver", # For RtxReceiveStream.
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"../call:video_stream_api",
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"../common_video",
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"../media:rtc_h264_profile_id",
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"../modules:module_api",
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"../modules/pacing",
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"../modules/remote_bitrate_estimator",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/rtp_rtcp:rtp_rtcp_legacy",
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"../modules/rtp_rtcp:rtp_video_header",
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"../modules/utility",
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"../modules/video_coding",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../modules/video_coding/deprecated:nack_module",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_numerics",
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"../rtc_base:rtc_task_queue",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/experiments:keyframe_interval_settings_experiment",
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"../rtc_base/synchronization:mutex",
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"../rtc_base/system:no_unique_address",
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"../rtc_base/system:thread_registry",
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"../rtc_base/task_utils:to_queued_task",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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]
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if (!build_with_mozilla) {
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deps += [ "../media:rtc_media_base" ]
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}
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absl_deps = [
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("video_stream_decoder_impl") {
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