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11d583f41484913fd1e7b3e283966eb7b7e11ed2
platform-external-webrtc/webrtc/modules/audio_coding
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henrik.lundin 11d583f414 Fix a bug in RtpFileSource related to RTCP packets in rtpdump files
According to http://www.cs.columbia.edu/irt/software/rtptools/#rtpdump,
RTCP packets are marked with plen==0. In this class, plen is mapped to
original_length, not length.

Review URL: https://codereview.webrtc.org/1356543002

Cr-Commit-Position: refs/heads/master@{#9981}
2015-09-18 08:28:14 +00:00
..
codecs
Move AudioDecoderG722 next to AudioEncoderG722
2015-09-17 10:12:38 +00:00
main
Added support for logging the SSRC corresponding to AudioPlayout events.
2015-09-17 14:34:15 +00:00
neteq
Fix a bug in RtpFileSource related to RTCP packets in rtpdump files
2015-09-18 08:28:14 +00:00
audio_codec_speed_tests.isolate
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
audio_coding_tests.gypi
Rename targets to use lower case format.
2015-06-10 20:45:12 +00:00
audio_coding.gypi
iSAC: Make separate AudioEncoder and AudioDecoder objects
2015-08-24 09:03:28 +00:00
BUILD.gn
Move AudioDecoderG722 next to AudioEncoderG722
2015-09-17 10:12:38 +00:00
OWNERS
Simplify OWNERS structure in modules/audio_coding
2015-06-29 11:54:50 +00:00
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