diff options
| author | xians@chromium.org <xians@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-06-20 20:17:34 +0000 |
|---|---|---|
| committer | xians@chromium.org <xians@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-06-20 20:17:34 +0000 |
| commit | 004aabce2f9aac00c6d1bf6f86a0c4e283a4caef (patch) | |
| tree | 1734113171e77f59386d394821940396e4e0e50f /content/renderer/media/webrtc_audio_renderer.cc | |
| parent | 2ef49b370b61080c96ffcfb57a5ed4bebb89cd2a (diff) | |
| download | chromium_src-004aabce2f9aac00c6d1bf6f86a0c4e283a4caef.zip chromium_src-004aabce2f9aac00c6d1bf6f86a0c4e283a4caef.tar.gz chromium_src-004aabce2f9aac00c6d1bf6f86a0c4e283a4caef.tar.bz2 | |
Fixed the UMA for webrtc sample rates. We need to map the sample rate to media::AudioSampleRate before we add the stat to the history.
How the old ADM did, please look at AddHistogramSampleRate() in
https://chromiumcodereview.appspot.com/11270012/diff/34001/content/renderer/media/webrtc_audio_device_impl.cc
Review URL: https://chromiumcodereview.appspot.com/17465009
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@207534 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/renderer/media/webrtc_audio_renderer.cc')
| -rw-r--r-- | content/renderer/media/webrtc_audio_renderer.cc | 9 |
1 files changed, 7 insertions, 2 deletions
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc index 9f9a1d6..caa4ab8 100644 --- a/content/renderer/media/webrtc_audio_renderer.cc +++ b/content/renderer/media/webrtc_audio_renderer.cc @@ -136,8 +136,13 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) { DVLOG(1) << "Resampling from 48000 to 192000 is required"; sample_rate = 48000; } - UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate", - sample_rate, media::kUnexpectedAudioSampleRate); + media::AudioSampleRate asr = media::AsAudioSampleRate(sample_rate); + if (asr != media::kUnexpectedAudioSampleRate) { + UMA_HISTOGRAM_ENUMERATION( + "WebRTC.AudioOutputSampleRate", asr, media::kUnexpectedAudioSampleRate); + } else { + UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected", sample_rate); + } // Verify that the reported output hardware sample rate is supported // on the current platform. |