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webrtc/modules/audio_coding/acm2
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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
..
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