Add histogram for percentage of sent frames that are limited in resolution due to bandwidth.

webrtc/modules/video_coding/codecs/vp8/vp8_impl.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 113 matching lines @@
     return false;
   }
   for (int i = 0; i < num_streams; ++i) {
     if (codec.width * codec.simulcastStream[i].height !=
         codec.height * codec.simulcastStream[i].width) {
       return false;
     }
   }
   return true;
 }
+
+bool SendingAllStreams(std::vector<bool>& streams) {
+  for (bool stream : streams) {
+    if (!stream)
+      return false;
+  }
+  return true;

[stefan-webrtc, 2015/09/16 07:49:27]

I think you could replace this with return std::find(streams.begin(),
streams.end(), false) == streams.end();

+}
 }  // namespace
 
 const float kTl1MaxTimeToDropFrames = 20.0f;
 
 VP8EncoderImpl::VP8EncoderImpl()
     : encoded_complete_callback_(NULL),
       inited_(false),
       timestamp_(0),
       feedback_mode_(false),
       qp_max_(56),  // Setting for max quantizer.
@@ skipping 793 matching lines @@
                                 only_predicting_from_key_frame;
   temporal_layers_[stream_idx]->PopulateCodecSpecific(base_layer_sync_point,
                                                       vp8Info,
                                                       timestamp);
   // Prepare next.
   picture_id_[stream_idx] = (picture_id_[stream_idx] + 1) & 0x7FFF;
 }
 
 int VP8EncoderImpl::GetEncodedPartitions(const VideoFrame& input_image,
                                          bool only_predicting_from_key_frame) {
+  bool bw_limited_resolution = false;
+  bool bw_limited_resolution_available = encoders_.size() > 1;
+  if (bw_limited_resolution_available)
+    bw_limited_resolution = !SendingAllStreams(send_stream_);

[stefan-webrtc, 2015/09/16 07:49:27]

Same reasoning as in the quality CL. Would it be useful to track how many
resolutions are disabled here?

[åsapersson, 2015/09/17 14:12:03]

Done.

+
   int stream_idx = static_cast<int>(encoders_.size()) - 1;
   int result = WEBRTC_VIDEO_CODEC_OK;
   for (size_t encoder_idx = 0; encoder_idx < encoders_.size();
       ++encoder_idx, --stream_idx) {
     vpx_codec_iter_t iter = NULL;
     int part_idx = 0;
     encoded_images_[encoder_idx]._length = 0;
     encoded_images_[encoder_idx]._frameType = kDeltaFrame;
     RTPFragmentationHeader frag_info;
     // token_partitions_ is number of bits used.
@@ skipping 30 matching lines @@
         }
         PopulateCodecSpecific(&codec_specific, *pkt, stream_idx,
                               input_image.timestamp(),
                               only_predicting_from_key_frame);
         break;
       }
     }
     encoded_images_[encoder_idx]._timeStamp = input_image.timestamp();
     encoded_images_[encoder_idx].capture_time_ms_ =
         input_image.render_time_ms();
+    encoded_images_[encoder_idx].adapt_reason_.bw_limited_resolution_available =
+        bw_limited_resolution_available;
+    encoded_images_[encoder_idx].adapt_reason_.bw_limited_resolution =
+        bw_limited_resolution;
 
     int qp = -1;
     vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
     temporal_layers_[stream_idx]->FrameEncoded(
         encoded_images_[encoder_idx]._length,
         encoded_images_[encoder_idx]._timeStamp, qp);
     if (send_stream_[stream_idx]) {
       if (encoded_images_[encoder_idx]._length > 0) {
         TRACE_COUNTER_ID1("webrtc", "EncodedFrameSize", encoder_idx,
                           encoded_images_[encoder_idx]._length);
@@ skipping 386 matching lines @@
     return -1;
   }
   if (vpx_codec_control(copy->decoder_, VP8_SET_REFERENCE, ref_frame_)
       != VPX_CODEC_OK) {
     return -1;
   }
   return 0;
 }
 
 }  // namespace webrtc