commit 9e6fd2bd4799e9abb8561eb552c2d87ea3e1ef32
author Henrik Boström <hbos@webrtc.org> Tue Nov 21 13:41:51 2017 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Nov 22 09:04:17 2017 +0000
tree 5b866cf0b199cbfca0929057d977d5a5a63b75df
parent dd41194ca4d5b14b6b1aa714084696cfccf31839

Add streams() to RtpReceiverInterface and implementations.

This makes the receiver know about its associated set of streams, the
equivalent of the [[AssociatedRemoteMediaStreams]] slot in the spec,
https://w3c.github.io/webrtc-pc/#dfn-x%5B%5Bassociatedremotemediastreams%5D%5D

This does not change layers below peerconnection.cc. The streams are set
upon the receiver's construction and is not modified for the duration of
its lifetime.

When we support modifying the associated set of streams of a receiver
the receiver needs to know about it. The receiver's streams() should be
used in all places where a receiver's streams need to be known.

Bug: webrtc:8473
Change-Id: I31202973aed98e61fa9b6a78b52e815227b6c17d
Reviewed-on: https://webrtc-review.googlesource.com/22922
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20825}

pc/peerconnection_rtp_unittest.cc

diff --git a/pc/peerconnection_rtp_unittest.cc b/pc/peerconnection_rtp_unittest.cc
index 0261b1a..860c6dc 100644
--- a/pc/peerconnection_rtp_unittest.cc
+++ b/pc/peerconnection_rtp_unittest.cc
@@ -72,10 +72,10 @@
   ASSERT_EQ(1u, callee->observer()->add_track_events_.size());
   // TODO(deadbeef): When no stream is handled correctly we would expect
   // |add_track_events_[0].streams| to be empty. https://crbug.com/webrtc/7933
-  ASSERT_EQ(1u, callee->observer()->add_track_events_[0].streams.size());
-  EXPECT_TRUE(
-      callee->observer()->add_track_events_[0].streams[0]->FindAudioTrack(
-          "audio_track"));
+  auto& add_track_event = callee->observer()->add_track_events_[0];
+  ASSERT_EQ(1u, add_track_event.streams.size());
+  EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
+  EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
 }
 
 TEST_F(PeerConnectionRtpTest, AddTrackWithStreamFiresOnAddTrack) {
@@ -89,12 +89,11 @@
   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
 
   ASSERT_EQ(1u, callee->observer()->add_track_events_.size());
-  ASSERT_EQ(1u, callee->observer()->add_track_events_[0].streams.size());
-  EXPECT_EQ("audio_stream",
-            callee->observer()->add_track_events_[0].streams[0]->label());
-  EXPECT_TRUE(
-      callee->observer()->add_track_events_[0].streams[0]->FindAudioTrack(
-          "audio_track"));
+  auto& add_track_event = callee->observer()->add_track_events_[0];
+  ASSERT_EQ(1u, add_track_event.streams.size());
+  EXPECT_EQ("audio_stream", add_track_event.streams[0]->label());
+  EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
+  EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
 }
 
 TEST_F(PeerConnectionRtpTest, RemoveTrackWithoutStreamFiresOnRemoveTrack) {