Reland "Implement transceiver.stop()"

This is a reland of 11dc6571cb4ff3e71dee1557dfff8d9076e108d3

One fix that makes Web Platform Tests pass in debug mode is applied.

Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}

Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}

api/rtp_transceiver_interface.cc

diff --git a/api/rtp_transceiver_interface.cc b/api/rtp_transceiver_interface.cc
index e795e51..1dc0fcc 100644
--- a/api/rtp_transceiver_interface.cc
+++ b/api/rtp_transceiver_interface.cc
@@ -25,6 +25,23 @@
   return absl::nullopt;
 }
 
+bool RtpTransceiverInterface::stopping() const {
+  return false;
+}
+
+void RtpTransceiverInterface::Stop() {
+  StopInternal();
+}
+
+RTCError RtpTransceiverInterface::StopStandard() {
+  RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopStandard called";
+  return RTCError::OK();
+}
+
+void RtpTransceiverInterface::StopInternal() {
+  RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopInternal called";
+}
+
 RTCError RtpTransceiverInterface::SetCodecPreferences(
     rtc::ArrayView<RtpCodecCapability>) {
   RTC_NOTREACHED() << "Not implemented";
@@ -47,4 +64,17 @@
   return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
 }
 
+// TODO(bugs.webrtc.org/11839) Remove default implementations when clients
+// are updated.
+void RtpTransceiverInterface::SetDirection(
+    RtpTransceiverDirection new_direction) {
+  SetDirectionWithError(new_direction);
+}
+
+RTCError RtpTransceiverInterface::SetDirectionWithError(
+    RtpTransceiverDirection new_direction) {
+  RTC_NOTREACHED() << "Default implementation called";
+  return RTCError::OK();
+}
+
 }  // namespace webrtc

api/rtp_transceiver_interface.h

diff --git a/api/rtp_transceiver_interface.h b/api/rtp_transceiver_interface.h
index 13277d9..cdda34b 100644
--- a/api/rtp_transceiver_interface.h
+++ b/api/rtp_transceiver_interface.h
@@ -89,6 +89,16 @@
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
   virtual bool stopped() const = 0;
 
+  // The stopping attribute indicates that the user has indicated that the
+  // sender of this transceiver will stop sending, and that the receiver will
+  // no longer receive. It is always true if stopped() is true.
+  // If stopping() is true and stopped() is false, it means that the
+  // transceiver's stop() method has been called, but the negotiation with
+  // the other end for shutting down the transceiver is not yet done.
+  // https://w3c.github.io/webrtc-pc/#dfn-stopping-0
+  // TODO(hta): Remove default implementation.
+  virtual bool stopping() const;
+
   // The direction attribute indicates the preferred direction of this
   // transceiver, which will be used in calls to CreateOffer and CreateAnswer.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
@@ -99,7 +109,10 @@
   // CreateOffer and CreateAnswer mark the corresponding media descriptions as
   // sendrecv, sendonly, recvonly, or inactive.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
-  virtual void SetDirection(RtpTransceiverDirection new_direction) = 0;
+  // TODO(hta): Deprecate SetDirection without error and rename
+  // SetDirectionWithError to SetDirection, remove default implementations.
+  virtual void SetDirection(RtpTransceiverDirection new_direction);
+  virtual RTCError SetDirectionWithError(RtpTransceiverDirection new_direction);
 
   // The current_direction attribute indicates the current direction negotiated
   // for this transceiver. If this transceiver has never been represented in an
@@ -114,10 +127,19 @@
   // Exposed in the public interface for use by Chromium.
   virtual absl::optional<RtpTransceiverDirection> fired_direction() const;
 
-  // The Stop method irreversibly stops the RtpTransceiver. The sender of this
-  // transceiver will no longer send, the receiver will no longer receive.
+  // Initiates a stop of the transceiver.
+  // The stop is complete when stopped() returns true.
+  // A stopped transceiver can be reused for a different track.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
-  virtual void Stop() = 0;
+  // TODO(hta): Rename to Stop() when users of the non-standard Stop() are
+  // updated.
+  virtual RTCError StopStandard();
+
+  // Stops a transceiver immediately, without waiting for signalling.
+  // This is an internal function, and is exposed for historical reasons.
+  // https://w3c.github.io/webrtc-pc/#dfn-stop-the-rtcrtptransceiver
+  virtual void StopInternal();
+  RTC_DEPRECATED virtual void Stop();
 
   // The SetCodecPreferences method overrides the default codec preferences used
   // by WebRTC for this transceiver.