Clean up histogram'd media enum max values.

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index 9187983773fc0dfcde351842ea0386c673fb1511..298436998c26d775c34fb324df9b2ca3b9d38c64 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -238,10 +238,10 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
     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);
@@ -336,7 +336,7 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
 
   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);