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 204 matching lines @@
   // WebRTC does not yet support higher rates than 96000 on the client side
   // and 48000 is the preferred sample rate. Therefore, if 192000 is detected,
   // we change the rate to 48000 instead. The consequence is that the native
   // layer will be opened up at 192kHz but WebRTC will provide data at 48kHz
   // which will then be resampled by the audio converted on the browser side
   // to match the native audio layer.
   if (sample_rate == 192000) {
     DVLOG(1) << "Resampling from 48000 to 192000 is required";
     sample_rate = 48000;
   }
-  media::AudioSampleRate asr = media::AsAudioSampleRate(sample_rate);
-  if (asr != media::kUnexpectedAudioSampleRate) {
+  media::AudioSampleRate asr;
+  if (media::ToAudioSampleRate(sample_rate, &asr)) {
     UMA_HISTOGRAM_ENUMERATION(
-        "WebRTC.AudioOutputSampleRate", asr, media::kUnexpectedAudioSampleRate);
+        "WebRTC.AudioOutputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected", sample_rate);
   }
 
   // Verify that the reported output hardware sample rate is supported
   // on the current platform.
   if (std::find(&kValidOutputRates[0],
                 &kValidOutputRates[0] + arraysize(kValidOutputRates),
                 sample_rate) ==
                     &kValidOutputRates[arraysize(kValidOutputRates)]) {
@@ skipping 84 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_MAX + 1);
   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