Consistent 30% improvement in audio mixer running time.

webrtc/modules/audio_mixer/audio_frame_manipulator.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 11 matching lines @@
     // TODO(aleloi): This can overflow. Convert to floats.
     energy += audio_frame.data_[position] * audio_frame.data_[position];
   }
   return energy;
 }
 
 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
   RTC_DCHECK(audio_frame);
   RTC_DCHECK_GE(start_gain, 0.0f);
   RTC_DCHECK_GE(target_gain, 0.0f);
+  if (start_gain == target_gain) {

[the sun, 2017/02/07 08:53:22]

Comparing floats is fuzzy and operator == is not to be trusted to do what you
intend. See e.g.:

http://stackoverflow.com/questions/41989570/in-c-is-exactly-one-of-and-guaran...
http://stackoverflow.com/questions/17333/what-is-the-most-effective-way-for-f...

hlundin@, do you know if we have any function in the codebase (outside of gtest)
for this?

[hlundin-webrtc, 2017/02/07 21:34:51]

No, I know of no such function in our codebase.

+    return;
+  }
 
   size_t samples = audio_frame->samples_per_channel_;
   RTC_DCHECK_LT(0, samples);
   float increment = (target_gain - start_gain) / samples;
   float gain = start_gain;
   for (size_t i = 0; i < samples; ++i) {
     // If the audio is interleaved of several channels, we want to
     // apply the same gain change to the ith sample of every channel.
     for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
       audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain;
     }
     gain += increment;
   }
 }
 
 void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
   RTC_DCHECK_GE(target_number_of_channels, 1);
   RTC_DCHECK_LE(target_number_of_channels, 2);
   if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
     AudioFrameOperations::MonoToStereo(frame);
   } else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
     AudioFrameOperations::StereoToMono(frame);
   }
 }
 }  // namespace webrtc