Karl Wiberg 053c371552 Audio coding: Don't choke when RTP timestamp rate > sample rate ... Bug: webrtc:10631 Change-Id: If0422786172502f039acc2cac5e8c13b637af54c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137048 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27998}		2019-05-21 03:10:49 +00:00
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acm_receive_test.cc	Update ACM to use RTPHeader instead of WebRtcRTPHeader	2019-02-18 08:01:31 +00:00
acm_receive_test.h	2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883	2019-02-22 09:59:01 +00:00
acm_receiver.cc	Expose new audio stats on the API	2019-05-03 10:10:15 +00:00
acm_receiver.h	Update ACM to use RTPHeader instead of WebRtcRTPHeader	2019-02-18 08:01:31 +00:00
acm_receiver_unittest.cc	Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData	2019-04-26 12:58:14 +00:00
acm_resampler.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
acm_resampler.h	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
acm_send_test.cc	Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData	2019-04-26 12:58:14 +00:00
acm_send_test.h	Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData	2019-04-26 12:58:14 +00:00
audio_coding_module.cc	Audio coding: Don't choke when RTP timestamp rate > sample rate	2019-05-21 03:10:49 +00:00
audio_coding_module_unittest.cc	Delete deprecated PlatformThread looping	2019-05-03 08:35:42 +00:00
call_statistics.cc	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
call_statistics.h	Move NetworkStatistics and AudioDecodingCallStats from common_types.h	2018-11-19 11:55:34 +00:00
call_statistics_unittest.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00