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webrtc/modules/audio_coding/acm2
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Alex Loiko e5b94160b5 Decoder for multistream Opus. 
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.

This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.

E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"

Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 16:15:37 +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 Decoder for multistream Opus. 2019-04-08 16:15:37 +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