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

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 130 matching lines @@
       timestamp_(0),
       feedback_mode_(false),
       qp_max_(56),  // Setting for max quantizer.
       cpu_speed_default_(-6),
       rc_max_intra_target_(0),
       token_partitions_(VP8_ONE_TOKENPARTITION),
       down_scale_requested_(false),
       down_scale_bitrate_(0),
       tl0_frame_dropper_(),
       tl1_frame_dropper_(kTl1MaxTimeToDropFrames),
-      key_frame_request_(kMaxSimulcastStreams, false) {
+      key_frame_request_(kMaxSimulcastStreams, false),
+      quality_scaler_enabled_(false) {
   uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
   srand(seed);
 
   picture_id_.reserve(kMaxSimulcastStreams);
   last_key_frame_picture_id_.reserve(kMaxSimulcastStreams);
   temporal_layers_.reserve(kMaxSimulcastStreams);
   raw_images_.reserve(kMaxSimulcastStreams);
   encoded_images_.reserve(kMaxSimulcastStreams);
   send_stream_.reserve(kMaxSimulcastStreams);
   cpu_speed_.assign(kMaxSimulcastStreams, -6);  // Set default to -6.
@@ skipping 414 matching lines @@
           stream_bitrates[stream_idx], inst->maxBitrate, inst->maxFramerate,
           &configurations_[i]);
     }
   }
 
   rps_.Init();
   quality_scaler_.Init(codec_.qpMax / QualityScaler::kDefaultLowQpDenominator,
                        false);
   quality_scaler_.ReportFramerate(codec_.maxFramerate);
 
+  // Only apply scaling to improve for single-layer streams. The scaling metrics
+  // use frame drops as a signal and is only applicable when we drop frames.
+  quality_scaler_enabled_ = encoders_.size() == 1 &&
+                            configurations_[0].rc_dropframe_thresh > 0 &&
+                            codec_.codecSpecific.VP8.automaticResizeOn;
+
   return InitAndSetControlSettings();
 }
 
 int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
 #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
   // On mobile platform, always set to -12 to leverage between cpu usage
   // and video quality.
   return -12;
 #else
   // For non-ARM, increase encoding complexity (i.e., use lower speed setting)
@@ skipping 101 matching lines @@
                            const std::vector<VideoFrameType>* frame_types) {
   TRACE_EVENT1("webrtc", "VP8::Encode", "timestamp", frame.timestamp());
 
   if (!inited_)
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   if (frame.IsZeroSize())
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   if (encoded_complete_callback_ == NULL)
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
 
-  // Only apply scaling to improve for single-layer streams. The scaling metrics
-  // use frame drops as a signal and is only applicable when we drop frames.
-  const bool use_quality_scaler = encoders_.size() == 1 &&
-                                  configurations_[0].rc_dropframe_thresh > 0 &&
-                                  codec_.codecSpecific.VP8.automaticResizeOn;
-  if (use_quality_scaler)
+  if (quality_scaler_enabled_)
     quality_scaler_.OnEncodeFrame(frame);
   const VideoFrame& input_image =
-      use_quality_scaler ? quality_scaler_.GetScaledFrame(frame) : frame;
+      quality_scaler_enabled_ ? quality_scaler_.GetScaledFrame(frame) : frame;
 
-  if (use_quality_scaler && (input_image.width() != codec_.width ||
+  if (quality_scaler_enabled_ && (input_image.width() != codec_.width ||
       input_image.height() != codec_.height)) {
     int ret = UpdateCodecFrameSize(input_image);
     if (ret < 0)
       return ret;
   }
 
   // Since we are extracting raw pointers from |input_image| to
   // |raw_images_[0]|, the resolution of these frames must match. Note that
   // |input_image| might be scaled from |frame|. In that case, the resolution of
   // |raw_images_[0]| should have been updated in UpdateCodecFrameSize.
@@ skipping 273 matching lines @@
         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);
         encoded_images_[encoder_idx]._encodedHeight =
             codec_.simulcastStream[stream_idx].height;
         encoded_images_[encoder_idx]._encodedWidth =
             codec_.simulcastStream[stream_idx].width;
+        encoded_images_[encoder_idx].adapt_reason_.quality_available =

[stefan-webrtc, 2015/09/23 15:08:07]

quality_scaling_available?

[åsapersson, 2015/09/24 08:16:17]

In other cl, adapt_reason.bw_available, keep to be similar?

[stefan-webrtc, 2015/09/28 13:44:29]

I guess it works since itäs part of adapt_reason_

+            quality_scaler_enabled_;
+        if (quality_scaler_enabled_) {
+          encoded_images_[encoder_idx]
+              .adapt_reason_.quality_resolution_downscales =
+              quality_scaler_.downscale_shift();
+        }
         encoded_complete_callback_->Encoded(encoded_images_[encoder_idx],
                                             &codec_specific, &frag_info);
       } else if (codec_.mode == kScreensharing) {
         result = WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT;
       }
     } else {
       // Required in case padding is applied to dropped frames.
       encoded_images_[encoder_idx]._length = 0;
       encoded_images_[encoder_idx]._frameType = kSkipFrame;
       codec_specific.codecType = kVideoCodecVP8;
@@ skipping 372 matching lines @@
     return -1;
   }
   if (vpx_codec_control(copy->decoder_, VP8_SET_REFERENCE, ref_frame_)
       != VPX_CODEC_OK) {
     return -1;
   }
   return 0;
 }
 
 }  // namespace webrtc