Avoid unnecessary HW video encoder reconfiguration

webrtc/config.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 124 matching lines @@
   std::vector<VideoStream> streams;
   std::vector<SpatialLayer> spatial_layers;
   ContentType content_type;
   void* encoder_specific_settings;
 
   // Padding will be used up to this bitrate regardless of the bitrate produced
   // by the encoder. Padding above what's actually produced by the encoder helps
   // maintaining a higher bitrate estimate. Padding will however not be sent
   // unless the estimated bandwidth indicates that the link can handle it.
   int min_transmit_bitrate_bps;
+  bool expect_encode_from_texture;
 };
 
 // Controls the capacity of the packet buffer in NetEq. The capacity is the
 // maximum number of packets that the buffer can contain. If the limit is
 // exceeded, the buffer will be flushed. The capacity does not affect the actual
 // audio delay in the general case, since this is governed by the target buffer
 // level (calculated from the jitter profile). It is only in the rare case of
 // severe network freezes that a higher capacity will lead to a (transient)
 // increase in audio delay.
 struct NetEqCapacityConfig {
@@ skipping 14 matching lines @@
 struct VoicePacing {
   VoicePacing() : enabled(false) {}
   explicit VoicePacing(bool value) : enabled(value) {}
   static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
   bool enabled;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CONFIG_H_