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![]() This ensure the absolute capture timestamp from the first audio sample encoded in the payload is used for the corresponding rtp header. Bug: webrtc:42226041 Change-Id: Ib8f2e3a5df5c82c5806171bd5b36a26d92fbea72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349265 Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42281} |
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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 |