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webrtc/modules/audio_coding/neteq/mock
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Jakob Ivarsson 10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay. 
Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
2018-11-27 19:49:48 +00:00
..
mock_buffer_level_filter.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_decoder_database.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_delay_manager.h Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
mock_delay_peak_detector.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_dtmf_buffer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_dtmf_tone_generator.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_expand.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_external_decoder_pcm16b.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
mock_packet_buffer.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
mock_red_payload_splitter.h NetEq: Handle nested RED packets 2018-07-03 20:27:57 +00:00
mock_statistics_calculator.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00