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Remove sigslot dependency from RtpTransceiver

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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}
This commit is contained in:
Tommi 2023-03-01 12:08:09 +01:00 committed by WebRTC LUCI CQ
parent ae61aca9b1
commit c1dc4fd0ef
1 changed files with 1 additions and 7 deletions
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8
pc/rtp_transceiver.h
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@ -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 @@ class PeerConnectionSdpMethods;
// 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 @@ class RtpTransceiver : public RtpTransceiverInterface,
// 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;