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); |