Delete some unused sigslot dependencies
Bug: webrtc:11943 Change-Id: Idc0d7aa0f63088810131ed0eebef2f165e66d646 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266495 Auto-Submit: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37314}
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@ -446,7 +446,6 @@ rtc_source_set("rtp_transport") {
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"../rtc_base:logging",
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"../rtc_base:socket",
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"../rtc_base/network:sent_packet",
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"../rtc_base/third_party/sigslot",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/strings",
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@ -608,7 +607,6 @@ rtc_source_set("srtp_transport") {
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"../rtc_base:safe_conversions",
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"../rtc_base:zero_memory",
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"../rtc_base/third_party/base64",
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"../rtc_base/third_party/sigslot",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/strings",
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@ -1134,7 +1132,6 @@ rtc_source_set("sdp_offer_answer") {
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"../rtc_base:stringutils",
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"../rtc_base:threading",
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"../rtc_base:weak_ptr",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers:metrics",
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]
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absl_deps = [
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@ -23,7 +23,6 @@
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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@ -28,7 +28,6 @@
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/socket.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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namespace rtc {
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@ -58,7 +58,6 @@
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#include "rtc_base/checks.h"
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#include "rtc_base/operations_chain.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/unique_id_generator.h"
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@ -77,8 +76,7 @@ namespace webrtc {
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// - Parsing and interpreting SDP.
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// - Generating offers and answers based on the current state.
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// This class lives on the signaling thread.
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class SdpOfferAnswerHandler : public SdpStateProvider,
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public sigslot::has_slots<> {
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class SdpOfferAnswerHandler : public SdpStateProvider {
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public:
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~SdpOfferAnswerHandler();
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@ -28,7 +28,6 @@
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/third_party/base64/base64.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/trace_event.h"
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#include "rtc_base/zero_memory.h"
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@ -175,8 +175,6 @@ void PeerConnectionTestWrapper::OnIceCandidate(
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const IceCandidateInterface* candidate) {
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std::string sdp;
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EXPECT_TRUE(candidate->ToString(&sdp));
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// Give the user a chance to modify sdp for testing.
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SignalOnIceCandidateCreated(&sdp);
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SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
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sdp);
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}
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@ -196,9 +194,6 @@ void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
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<< webrtc::SdpTypeToString(desc->GetType())
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<< " sdp created: " << sdp;
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// Give the user a chance to modify sdp for testing.
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SignalOnSdpCreated(&sdp);
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SetLocalDescription(desc->GetType(), sdp);
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SignalOnSdpReady(sdp);
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@ -99,10 +99,8 @@ class PeerConnectionTestWrapper
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bool video);
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// sigslots
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sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
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sigslot::signal3<const std::string&, int, const std::string&>
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SignalOnIceCandidateReady;
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sigslot::signal1<std::string*> SignalOnSdpCreated;
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sigslot::signal1<const std::string&> SignalOnSdpReady;
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sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
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