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_unittest.cc

Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData

2019-04-26 12:58:14 +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_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_unittest.cc

Delete deprecated PlatformThread looping

2019-05-03 08:35:42 +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

call_statistics_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +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