Clean up histogram'd media enum max values.

content/renderer/media/webrtc_audio_renderer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 312 matching lines @@
   DCHECK_GE(session_id_, 0);
   sink_->InitializeUnifiedStream(sink_params, this, session_id_);
 
   sink_->Start();
 
   // User must call Play() before any audio can be heard.
   state_ = PAUSED;
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputChannelLayout",
                             source_params.channel_layout(),
-                            media::CHANNEL_LAYOUT_MAX);
+                            media::CHANNEL_LAYOUT_HISTOGRAM_MAX);
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
                             source_params.frames_per_buffer(),
                             kUnexpectedAudioBufferSize);
   AddHistogramFramesPerBuffer(source_params.frames_per_buffer());
 
   return true;
 }
 
 scoped_refptr<MediaStreamAudioRenderer>
 WebRtcAudioRenderer::CreateSharedAudioRendererProxy() {
@@ skipping 136 matching lines @@
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(buffer_.get(),
                              audio_bus->frames(),
                              sizeof(buffer_[0]));
 }
 
 }  // namespace content