Remove sigslot dependency from RtpTransceiver
Bug: webrtc:11943
Change-Id: I4212c90088671150f4fe828ad238380bf71b938e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295720
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39440}
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 0844b34..530522a 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -43,7 +43,6 @@
#include "pc/rtp_sender_proxy.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread_annotations.h"
namespace cricket {
@@ -82,8 +81,7 @@
// MediaType specified in the constructor. Audio RtpTransceivers will have
// AudioRtpSenders, AudioRtpReceivers, and a VoiceChannel. Video RtpTransceivers
// will have VideoRtpSenders, VideoRtpReceivers, and a VideoChannel.
-class RtpTransceiver : public RtpTransceiverInterface,
- public sigslot::has_slots<> {
+class RtpTransceiver : public RtpTransceiverInterface {
public:
// Construct a Plan B-style RtpTransceiver with no senders, receivers, or
// channel set.
@@ -257,10 +255,6 @@
// the webrtc-pc specification, described under the stop() method.
void StopTransceiverProcedure();
- // Fired when the RtpTransceiver state changes such that negotiation is now
- // needed (e.g., in response to a direction change).
- // sigslot::signal0<> SignalNegotiationNeeded;
-
// RtpTransceiverInterface implementation.
cricket::MediaType media_type() const override;
absl::optional<std::string> mid() const override;
