Reland of Adding the ability to create an RtpSender without a track.
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index b0c9129..63bd87c 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -31,7 +31,7 @@
namespace webrtc {
-static const char kAudioTrackKind[] = "audio";
+const char MediaStreamTrackInterface::kAudioKind[] = "audio";
AudioTrack::AudioTrack(const std::string& label,
AudioSourceInterface* audio_source)
@@ -40,7 +40,7 @@
}
std::string AudioTrack::kind() const {
- return kAudioTrackKind;
+ return kAudioKind;
}
rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
