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914351de5c
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1) Original cl description: Always offer transport sequence number header extension for audio If the extension is negotiated, it will only be used if the field trial WebRTC-Audio-SendSideBwe is enabled. This allows simpler experimentation if it should be used or not. Patchset 3 contain the only change: Add the field trial WebRTC-Audio-SendSideBwe to call/rampup_tests.cc TBR: srte@webrtc.org,ossu@webrtc.org Bug: webrtc:10309 webrtc:10286 Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f Reviewed-on: https://webrtc-review.googlesource.com/c/123183 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26706} |
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| .. | ||
| alr_experiment.cc | ||
| alr_experiment.h | ||
| audio_allocation_settings.cc | ||
| audio_allocation_settings.h | ||
| BUILD.gn | ||
| cpu_speed_experiment.cc | ||
| cpu_speed_experiment.h | ||
| cpu_speed_experiment_unittest.cc | ||
| DEPS | ||
| field_trial_parser.cc | ||
| field_trial_parser.h | ||
| field_trial_parser_unittest.cc | ||
| field_trial_units.cc | ||
| field_trial_units.h | ||
| field_trial_units_unittest.cc | ||
| jitter_upper_bound_experiment.cc | ||
| jitter_upper_bound_experiment.h | ||
| normalize_simulcast_size_experiment.cc | ||
| normalize_simulcast_size_experiment.h | ||
| normalize_simulcast_size_experiment_unittest.cc | ||
| OWNERS | ||
| quality_scaling_experiment.cc | ||
| quality_scaling_experiment.h | ||
| quality_scaling_experiment_unittest.cc | ||
| rate_control_settings.cc | ||
| rate_control_settings.h | ||
| rate_control_settings_unittest.cc | ||
| rtt_mult_experiment.cc | ||
| rtt_mult_experiment.h | ||
| rtt_mult_experiment_unittest.cc | ||