commit 6060df594871aee72d835700cba0c6d15029fb6a
author Harald Alvestrand <hta@webrtc.org> Tue Aug 11 07:54:02 2020
committer Commit Bot <commit-bot@chromium.org> Tue Aug 11 10:46:23 2020
tree 82aceaebd75de586db17a747a13914ea03fe90ef
parent f867ba8b5d48f4c2ff510380db287e1c6c367d4f

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}

pc/media_session.cc

diff --git a/pc/media_session.cc b/pc/media_session.cc
index 69ddb0c..0cff84d 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -1507,7 +1507,6 @@
   RtpDataCodecs offer_rtp_data_codecs;
   GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
                     &offer_video_codecs, &offer_rtp_data_codecs);
-
   if (!session_options.vad_enabled) {
     // If application doesn't want CN codecs in offer.
     StripCNCodecs(&offer_audio_codecs);
@@ -1796,15 +1795,13 @@
   switch (direction) {
     // If stream is inactive - generate list as if sendrecv.
     case RtpTransceiverDirection::kSendRecv:
+    case RtpTransceiverDirection::kStopped:
     case RtpTransceiverDirection::kInactive:
       return audio_sendrecv_codecs_;
     case RtpTransceiverDirection::kSendOnly:
       return audio_send_codecs_;
     case RtpTransceiverDirection::kRecvOnly:
       return audio_recv_codecs_;
-    case RtpTransceiverDirection::kStopped:
-      RTC_NOTREACHED();
-      return audio_sendrecv_codecs_;
   }
 }
 
@@ -1815,6 +1812,7 @@
     // For inactive and sendrecv answers, generate lists as if we were to accept
     // the offer's direction. See RFC 3264 Section 6.1.
     case RtpTransceiverDirection::kSendRecv:
+    case RtpTransceiverDirection::kStopped:
     case RtpTransceiverDirection::kInactive:
       return GetAudioCodecsForOffer(
           webrtc::RtpTransceiverDirectionReversed(offer));
@@ -1822,9 +1820,6 @@
       return audio_send_codecs_;
     case RtpTransceiverDirection::kRecvOnly:
       return audio_recv_codecs_;
-    case RtpTransceiverDirection::kStopped:
-      RTC_NOTREACHED();
-      return audio_sendrecv_codecs_;
   }
 }
 
@@ -1833,15 +1828,13 @@
   switch (direction) {
     // If stream is inactive - generate list as if sendrecv.
     case RtpTransceiverDirection::kSendRecv:
+    case RtpTransceiverDirection::kStopped:
     case RtpTransceiverDirection::kInactive:
       return video_sendrecv_codecs_;
     case RtpTransceiverDirection::kSendOnly:
       return video_send_codecs_;
     case RtpTransceiverDirection::kRecvOnly:
       return video_recv_codecs_;
-    case RtpTransceiverDirection::kStopped:
-      RTC_NOTREACHED();
-      return video_sendrecv_codecs_;
   }
 }
 
@@ -1852,6 +1845,7 @@
     // For inactive and sendrecv answers, generate lists as if we were to accept
     // the offer's direction. See RFC 3264 Section 6.1.
     case RtpTransceiverDirection::kSendRecv:
+    case RtpTransceiverDirection::kStopped:
     case RtpTransceiverDirection::kInactive:
       return GetVideoCodecsForOffer(
           webrtc::RtpTransceiverDirectionReversed(offer));
@@ -1859,9 +1853,6 @@
       return video_send_codecs_;
     case RtpTransceiverDirection::kRecvOnly:
       return video_recv_codecs_;
-    case RtpTransceiverDirection::kStopped:
-      RTC_NOTREACHED();
-      return video_sendrecv_codecs_;
   }
 }