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webrtc/modules/audio_coding/acm2
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Alex Loiko 50b8c399c9 Generalize the C-language Opus interface. 
Switch to explicit channel mappings (RFC 7845) when creating
multi-stream Opus en/de-coders. The responsibility of setting up the
channel mappings will shift from WebRTC to the WebRTC user.

See https://webrtc-review.googlesource.com/c/src/+/121764 for the
current vision. See also the first child CL
https://webrtc-review.googlesource.com/c/src/+/129768
that sets up the Decoder to use this code.

Bug: webrtc:8649
Change-Id: I55959a293d54bb4c982eff68ec107c5ef8666c5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129767
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27452}
2019-04-04 14:06:44 +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 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h 2019-03-19 16:59:27 +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 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h 2019-03-19 16:59:27 +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 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h 2019-03-19 16:59:27 +00:00
audio_coding_module_unittest.cc Generalize the C-language Opus interface. 2019-04-04 14:06:44 +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