Revert^2 "Delete pc/peerconnection build target"
This reverts commit 771b524606f43e682d63aa3a0724b21e8d14aac0.
Reason for revert: Downstream usage removed
Original change's description:
> Revert "Delete pc/peerconnection build target"
>
> This reverts commit 18a42e3272a6a25a23042fd39e67de02def8cafb.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Delete pc/peerconnection build target
> >
> > It is not useful any more.
> >
> > Bug: webrtc:13634, b/238176207
> > Change-Id: I3dd4ebca355bb828c6c3c30392333d9fe03a478c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267821
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41427}
>
> Bug: webrtc:13634, b/238176207
> Change-Id: Ib53e0b0cc81ac218e3c19e4c652ffe0b19155c22
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332220
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Christoffer Dewerin <jansson@google.com>
> Commit-Queue: Christoffer Dewerin <jansson@google.com>
> Cr-Commit-Position: refs/heads/main@{#41430}
Bug: webrtc:13634, b/238176207
Change-Id: I3e99aa0ae37350b56e5f33be932f78903d1d4969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334120
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41543}
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index baab6f7..e351748 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -16,7 +16,6 @@
# - rtc_pc
# - session_description
# - simulcast_description
-# - peerconnection
# - sdp_utils
# - media_stream_observer
# - video_track_source
@@ -736,142 +735,6 @@
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
-rtc_source_set("peerconnection") {
- # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
- visibility = [ "*" ] # Used by Chromium and others
- allow_poison = [ "environment_construction" ]
- cflags = []
- sources = []
-
- deps = [
- ":audio_rtp_receiver",
- ":audio_track",
- ":connection_context",
- ":data_channel_controller",
- ":data_channel_utils",
- ":dtmf_sender",
- ":ice_server_parsing",
- ":jitter_buffer_delay",
- ":jsep_ice_candidate",
- ":jsep_session_description",
- ":legacy_stats_collector",
- ":legacy_stats_collector_interface",
- ":local_audio_source",
- ":media_protocol_names",
- ":media_stream",
- ":media_stream_observer",
- ":peer_connection",
- ":peer_connection_factory",
- ":peer_connection_internal",
- ":peer_connection_message_handler",
- ":proxy",
- ":remote_audio_source",
- ":rtc_stats_collector",
- ":rtc_stats_traversal",
- ":rtp_parameters_conversion",
- ":rtp_receiver",
- ":rtp_sender",
- ":rtp_transceiver",
- ":rtp_transmission_manager",
- ":sctp_data_channel",
- ":sdp_offer_answer",
- ":sdp_state_provider",
- ":sdp_utils",
- ":session_description",
- ":simulcast_description",
- ":simulcast_sdp_serializer",
- ":stream_collection",
- ":track_media_info_map",
- ":transceiver_list",
- ":usage_pattern",
- ":video_rtp_receiver",
- ":video_track",
- ":video_track_source",
- ":webrtc_sdp",
- ":webrtc_session_description_factory",
- "../api:array_view",
- "../api:async_dns_resolver",
- "../api:audio_options_api",
- "../api:call_api",
- "../api:fec_controller_api",
- "../api:field_trials_view",
- "../api:frame_transformer_interface",
- "../api:ice_transport_factory",
- "../api:libjingle_logging_api",
- "../api:libjingle_peerconnection_api",
- "../api:media_stream_interface",
- "../api:network_state_predictor_api",
- "../api:packet_socket_factory",
- "../api:priority",
- "../api:rtc_error",
- "../api:rtc_event_log_output_file",
- "../api:rtc_stats_api",
- "../api:rtp_parameters",
- "../api:rtp_transceiver_direction",
- "../api:scoped_refptr",
- "../api:sequence_checker",
- "../api/adaptation:resource_adaptation_api",
- "../api/audio_codecs:audio_codecs_api",
- "../api/crypto:frame_decryptor_interface",
- "../api/crypto:options",
- "../api/neteq:neteq_api",
- "../api/rtc_event_log",
- "../api/task_queue",
- "../api/task_queue:pending_task_safety_flag",
- "../api/transport:bitrate_settings",
- "../api/transport:datagram_transport_interface",
- "../api/transport:enums",
- "../api/transport:field_trial_based_config",
- "../api/transport:network_control",
- "../api/transport:sctp_transport_factory_interface",
- "../api/units:data_rate",
- "../api/video:builtin_video_bitrate_allocator_factory",
- "../api/video:video_bitrate_allocator_factory",
- "../api/video:video_codec_constants",
- "../api/video:video_frame",
- "../api/video:video_rtp_headers",
- "../api/video_codecs:video_codecs_api",
- "../call:call_interfaces",
- "../call:rtp_interfaces",
- "../call:rtp_sender",
- "../common_video",
- "../logging:ice_log",
- "../media:rtc_data_sctp_transport_internal",
- "../media:rtc_media_base",
- "../media:rtc_media_config",
- "../modules/audio_processing:audio_processing_statistics",
- "../modules/rtp_rtcp:rtp_rtcp_format",
- "../p2p:rtc_p2p",
- "../rtc_base:callback_list",
- "../rtc_base:checks",
- "../rtc_base:ip_address",
- "../rtc_base:network_constants",
- "../rtc_base:rtc_operations_chain",
- "../rtc_base:safe_minmax",
- "../rtc_base:socket_address",
- "../rtc_base:threading",
- "../rtc_base:weak_ptr",
- "../rtc_base/experiments:field_trial_parser",
- "../rtc_base/network:sent_packet",
- "../rtc_base/synchronization:mutex",
- "../rtc_base/system:file_wrapper",
- "../rtc_base/system:no_unique_address",
- "../rtc_base/system:rtc_export",
- "../rtc_base/system:unused",
- "../rtc_base/third_party/base64",
- "../rtc_base/third_party/sigslot",
- "../stats",
- "../system_wrappers",
- "../system_wrappers:field_trial",
- "../system_wrappers:metrics",
- ]
- absl_deps = [
- "//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
-}
-
rtc_library("sctp_data_channel") {
visibility = [ ":*" ]
sources = [
@@ -2119,7 +1982,6 @@
":media_protocol_names",
":media_session",
":pc_test_utils",
- ":peerconnection",
":rtc_pc",
":rtcp_mux_filter",
":rtp_media_utils",
@@ -2220,7 +2082,6 @@
deps = [
":pc_test_utils",
":peer_connection",
- ":peerconnection",
":peerconnection_wrapper",
"../api:audio_options_api",
"../api:create_peerconnection_factory",
@@ -2273,7 +2134,6 @@
]
deps = [
":pc_test_utils",
- ":peerconnection",
":sdp_utils",
"../api:function_view",
"../api:libjingle_peerconnection_api",
@@ -2631,7 +2491,6 @@
":peer_connection",
":peer_connection_factory",
":peer_connection_proxy",
- ":peerconnection",
":remote_audio_source",
":rtp_media_utils",
":rtp_parameters_conversion",
@@ -2790,7 +2649,6 @@
":jitter_buffer_delay",
":libjingle_peerconnection",
":peer_connection_internal",
- ":peerconnection",
":rtp_receiver",
":rtp_sender",
":sctp_data_channel",
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 88fd1e7..1f67266 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -828,7 +828,6 @@
":generated_metrics_jni",
":native_api_jni",
":peerconnection_jni",
- "../../pc:peerconnection",
"../../rtc_base:stringutils",
"../../system_wrappers:metrics",
]
