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![]() This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k). Bug: none Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38536} |
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acm_receive_test.cc | ||
acm_receive_test.h | ||
acm_receiver.cc | ||
acm_receiver.h | ||
acm_receiver_unittest.cc | ||
acm_remixing.cc | ||
acm_remixing.h | ||
acm_remixing_unittest.cc | ||
acm_resampler.cc | ||
acm_resampler.h | ||
acm_send_test.cc | ||
acm_send_test.h | ||
audio_coding_module.cc | ||
audio_coding_module_unittest.cc | ||
call_statistics.cc | ||
call_statistics.h | ||
call_statistics_unittest.cc |