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webrtc/modules/audio_coding/acm2
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Niels Möller 87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType 
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +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 relative packet arrival delay metric in stats API. 2019-03-06 16:35:16 +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 Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +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 Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
acm_send_test.h Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
audio_coding_module.cc Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
audio_coding_module_unittest.cc Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +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