commit bcd39d483db751aec0806262307f4db76d0a9f07
author Amit Hilbuch <amithi@webrtc.org> Sat Jan 26 01:13:56 2019
committer Commit Bot <commit-bot@chromium.org> Mon Jan 28 18:56:02 2019
tree 0e5a2227157c19ed48b43a848419ca32cdef14f1
parent e76ca612388cfc7fbd53af8be7b9be5129d29e25

Creating Simulcast offer and answer in Peer Connection.

CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.

Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26428}

pc/channel_manager.h

diff --git a/pc/channel_manager.h b/pc/channel_manager.h
index 38d79ee..b70ed2e 100644
--- a/pc/channel_manager.h
+++ b/pc/channel_manager.h
@@ -100,6 +100,7 @@
       const std::string& content_name,
       bool srtp_required,
       const webrtc::CryptoOptions& crypto_options,
+      rtc::UniqueRandomIdGenerator* ssrc_generator,
       const AudioOptions& options);
   // Destroys a voice channel created by CreateVoiceChannel.
   void DestroyVoiceChannel(VoiceChannel* voice_channel);
@@ -116,6 +117,7 @@
       const std::string& content_name,
       bool srtp_required,
       const webrtc::CryptoOptions& crypto_options,
+      rtc::UniqueRandomIdGenerator* ssrc_generator,
       const VideoOptions& options);
   // Destroys a video channel created by CreateVideoChannel.
   void DestroyVideoChannel(VideoChannel* video_channel);
@@ -126,7 +128,8 @@
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,
-      const webrtc::CryptoOptions& crypto_options);
+      const webrtc::CryptoOptions& crypto_options,
+      rtc::UniqueRandomIdGenerator* ssrc_generator);
   // Destroys a data channel created by CreateRtpDataChannel.
   void DestroyRtpDataChannel(RtpDataChannel* data_channel);