Move reencode logic for screenshare bitrate overshoot from generic

webrtc/modules/video_coding/generic_encoder.cc

 /*
 *  Copyright (c) 2012 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 69 matching lines @@
 
   for (FrameType frame_type : frame_types)
     RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta);
 
   for (size_t i = 0; i < streams_or_svc_num_; ++i)
     vcm_encoded_frame_callback_->OnEncodeStarted(frame.render_time_ms(), i);
   int32_t result = encoder_->Encode(frame, codec_specific, &frame_types);
 
   if (is_screenshare_ &&
       result == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT) {
-    // Target bitrate exceeded, encoder state has been reset - try again.
-    return encoder_->Encode(frame, codec_specific, &frame_types);
+    RTC_NOTREACHED();

[ilnik, 2017/09/12 12:52:00]

Should we leave this here? What if libvpx return overshoot again on the
reencoding attempt?

[sprang_webrtc, 2017/09/12 14:17:11]

It should not, and we don't want to get stuck in a loop. VP9 also already does
this drop-re-encode logic internally.
I suggest removing this if-statement entirely and just return Encode() result
immediately.

Also, if we do that, check if is_screenshare_ is now unused and can be removed.

[ilnik, 2017/09/12 14:34:58]

Done.

   }
 
   return result;
 }
 
 void VCMGenericEncoder::SetEncoderParameters(const EncoderParameters& params) {
   RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
   bool channel_parameters_have_changed;
   bool rates_have_changed;
   {
@@ skipping 243 matching lines @@
     media_opt_->UpdateWithEncodedData(encoded_image);
     if (internal_source_) {
       // Signal to encoder to drop next frame.
       result.drop_next_frame = media_opt_->DropFrame();
     }
   }
   return result;
 }
 
 }  // namespace webrtc