Well, in fact we need to return both. But return codec sample rate separately and let the SdpAudioFormat contain the RTP clockrate, otherwise we're essentially lying to our callers. Bug: webrtc:11028 Change-Id: I40f36cb9db6b9824404ade6b0515a8312ff97009 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156307 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29444} |
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| .. | ||
| acm_receive_test.cc | ||
| acm_receive_test.h | ||
| acm_receiver_unittest.cc | ||
| acm_receiver.cc | ||
| acm_receiver.h | ||
| acm_resampler.cc | ||
| acm_resampler.h | ||
| acm_send_test.cc | ||
| acm_send_test.h | ||
| audio_coding_module_unittest.cc | ||
| audio_coding_module.cc | ||
| call_statistics_unittest.cc | ||
| call_statistics.cc | ||
| call_statistics.h | ||