content/renderer/media/webrtc_audio_capturer.cc - Issue 14430003: Using native sampling rate and optimal buffer size for audio on Android.

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Unified Diff: content/renderer/media/webrtc_audio_capturer.cc

Issue 14430003: Using native sampling rate and optimal buffer size for audio on Android. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master

Patch Set: Created 7 years, 8 months ago

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Index: content/renderer/media/webrtc_audio_capturer.cc

diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc

index b07f398515181b9b569abed8da1e6413b5bab443..e5aa3281e2018eac7d7d6a6cee877ccb46c64a43 100644

--- a/content/renderer/media/webrtc_audio_capturer.cc

+++ b/content/renderer/media/webrtc_audio_capturer.cc

@@ -26,7 +26,7 @@ static int kValidInputRates[] = {96000, 48000, 44100, 32000, 16000, 8000};

#elif defined(OS_LINUX) || defined(OS_OPENBSD)

static int kValidInputRates[] = {48000, 44100};

#elif defined(OS_ANDROID)

-static int kValidInputRates[] = {48000, 44100, 16000};

+static int kValidInputRates[] = {48000, 44100};

#else

static int kValidInputRates[] = {44100};

#endif

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