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webrtc_m130/modules/audio_coding/acm2
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Danil Chapovalov b4100ad06a Avoid using legacy rtp parser in neteq test::Packet
Bug: None
Change-Id: I9184954d9c99f0a34ae335d03843171864071e5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222648
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34316}
2021-06-17 08:38:14 +00:00
..
acm_receive_test.cc
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acm_receive_test.h
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acm_receiver_unittest.cc
Reland "Send absolute capture time through audio coding module."
2020-01-27 13:18:27 +00:00
acm_receiver.cc
Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz.
2021-06-09 18:41:47 +00:00
acm_receiver.h
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
acm_remixing_unittest.cc
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acm_remixing.cc
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acm_remixing.h
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acm_resampler.cc
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acm_resampler.h
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acm_send_test.cc
Avoid using legacy rtp parser in neteq test::Packet
2021-06-17 08:38:14 +00:00
acm_send_test.h
Reland "Send absolute capture time through audio coding module."
2020-01-27 13:18:27 +00:00
audio_coding_module_unittest.cc
Reland "Refactor the PlatformThread API."
2021-05-07 14:14:43 +00:00
audio_coding_module.cc
Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
2020-07-07 14:35:58 +00:00
call_statistics_unittest.cc
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call_statistics.cc
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call_statistics.h
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