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 549a63c4a3f2ecefd3ceeae25d0649e61f7589d4..edc014a1cbb09d72a2ba1bef7d1fd662ad4dc499 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);