From 004aabce2f9aac00c6d1bf6f86a0c4e283a4caef Mon Sep 17 00:00:00 2001 From: "xians@chromium.org" Date: Thu, 20 Jun 2013 20:17:34 +0000 Subject: 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 --- content/renderer/media/webrtc_audio_capturer.cc | 9 +++++++-- content/renderer/media/webrtc_audio_renderer.cc | 9 +++++++-- 2 files changed, 14 insertions(+), 4 deletions(-) (limited to 'content/renderer') diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc index ecb9656..90e2482 100644 --- a/content/renderer/media/webrtc_audio_capturer.cc +++ b/content/renderer/media/webrtc_audio_capturer.cc @@ -137,8 +137,13 @@ bool WebRtcAudioCapturer::Initialize(int render_view_id, } DVLOG(1) << "Audio input hardware sample rate: " << sample_rate; - UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputSampleRate", - sample_rate, media::kUnexpectedAudioSampleRate); + media::AudioSampleRate asr = media::AsAudioSampleRate(sample_rate); + if (asr != media::kUnexpectedAudioSampleRate) { + UMA_HISTOGRAM_ENUMERATION( + "WebRTC.AudioInputSampleRate", asr, media::kUnexpectedAudioSampleRate); + } else { + UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected", sample_rate); + } // Verify that the reported input hardware sample rate is supported // on the current platform. 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. -- cgit v1.1