Remove simulcast bitrate modes.

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 678 matching lines @@
     video_options_.unsignalled_recv_stream_limit.Get(&stream_limit);
     stream_limit = std::min(kMaxUnsignalledRecvStreams, stream_limit);
     stream_limit = std::max(0, stream_limit);
     video_options_.unsignalled_recv_stream_limit.Set(stream_limit);
   }
 
   SetOptionFromOptionalConstraint(constraints,
       MediaConstraintsInterface::kHighStartBitrate,
       &video_options_.video_start_bitrate);
 
-  if (FindConstraint(
-      constraints,
-      MediaConstraintsInterface::kVeryHighBitrate,
-      &value,
-      NULL)) {
-    video_options_.video_highest_bitrate.Set(
-        cricket::VideoOptions::VERY_HIGH);
-  } else if (FindConstraint(
-      constraints,
-      MediaConstraintsInterface::kHighBitrate,
-      &value,
-      NULL)) {
-    video_options_.video_highest_bitrate.Set(
-        cricket::VideoOptions::HIGH);
-  }
-
   SetOptionFromOptionalConstraint(constraints,
       MediaConstraintsInterface::kCombinedAudioVideoBwe,
       &audio_options_.combined_audio_video_bwe);
 
   audio_options_.audio_jitter_buffer_max_packets.Set(
       rtc_configuration.audio_jitter_buffer_max_packets);
 
   audio_options_.audio_jitter_buffer_fast_accelerate.Set(
       rtc_configuration.audio_jitter_buffer_fast_accelerate);
 
@@ skipping 1486 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc