a31c428307
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary resampling in VoE, allowing us to pass in the native 44.1 kHz. Our ALSA interface always requires 48 kHz, allowing ALSA to handle resampling. This also removes WEBRTC_PA_GTALK which was not defined anywhere. BUG=webrtc:1395 TESTED=Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good. Testing AEC was difficult as I can't find a way to change the sample rate of an individual device in PulseAudio. Using a webcam at 32 kHz, other problems were the overriding contribution to quality degradation (delay issues, possible clock drift from the camera). At least I verified that the quality got no worse with this patch. R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1384004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3955 4adac7df-926f-26a2-2b94-8c16560cd09d |
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data | ||
samples/js | ||
third_party | ||
third_party_mods | ||
tools | ||
webrtc | ||
.gitignore | ||
android-webrtc.mk | ||
Android.mk | ||
AUTHORS | ||
codereview.settings | ||
DEPS | ||
libvpx.mk | ||
LICENSE | ||
license_template.txt | ||
LICENSE_THIRD_PARTY | ||
OWNERS | ||
PATENTS | ||
PRESUBMIT.py | ||
WATCHLISTS | ||
webrtc.gyp |