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webrtc/call
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Sebastian Jansson 464a5576ea Adds audio priority bitrate field trial parameter. 
This replaces the functionality provided by
AudioPriorityBitrateAllocationStrategy, removing the need provide that
component via injection in all clients using audio bitrate priority.

Bug: webrtc:10286
Change-Id: I3bafab56d24459d9d27dc07abffdc8538440a346
Reviewed-on: https://webrtc-review.googlesource.com/c/121402
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26651}
2019-02-12 16:03:22 +00:00
..
test Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
audio_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Propagate base minimum delay to audio_receiver_stream 2019-02-06 11:07:42 +00:00
audio_send_stream.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_send_stream.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
audio_state.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
bitrate_allocator_unittest.cc Adds audio priority bitrate field trial parameter. 2019-02-12 16:03:22 +00:00
bitrate_allocator.cc Adds audio priority bitrate field trial parameter. 2019-02-12 16:03:22 +00:00
bitrate_allocator.h Adds audio priority bitrate field trial parameter. 2019-02-12 16:03:22 +00:00
bitrate_estimator_tests.cc Include absl/memory/memory.h if absl::make_unique is used 2019-01-08 20:08:32 +00:00
BUILD.gn Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
call_config.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
call_config.h Remove comments about using std::shared_ptr. 2019-01-15 13:15:58 +00:00
call_factory.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
call_factory.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
call_perf_tests.cc Reduce webrtc_perf_tests duration on buildbots 2019-01-22 11:48:33 +00:00
call_unittest.cc Remove Transport implementation from ChannelSend 2018-12-18 09:34:52 +00:00
call.cc Plumbing of feedback on request setting 2019-02-08 10:54:21 +00:00
call.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
degraded_call.cc Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
degraded_call.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
fake_network_pipe.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
flexfec_receive_stream_impl.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
flexfec_receive_stream_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
flexfec_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
OWNERS Makes srte owner in call/ and test/ 2018-07-13 08:39:41 +00:00
packet_receiver.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rampup_tests.cc [clang-tidy] Apply performance-for-range-copy fixes. 2019-01-28 09:53:50 +00:00
rampup_tests.h Refactor bitrate configuration in CallTest 2018-11-13 16:03:00 +00:00
receive_time_calculator_unittest.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
receive_time_calculator.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
receive_time_calculator.h [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtcp_demuxer_unittest.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtcp_demuxer.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtcp_demuxer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_bitrate_configurator_unittest.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rtp_config.cc Adding rtcp report interval into RTCConfiguration. 2018-11-12 20:00:00 +00:00
rtp_config.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rtp_demuxer_unittest.cc Delete unneeded includes of common_types.h and gn deps on webrtc_common. 2018-11-20 16:28:39 +00:00
rtp_demuxer.cc Allow all "token" chars from RFC 4566 when checking for legal mid names. 2018-08-01 18:20:42 +00:00
rtp_demuxer.h Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +00:00
rtp_payload_params.cc H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +00:00
rtp_payload_params.h Pass explicit frame dependency information to RtpPayloadParams 2019-01-29 08:59:48 +00:00
rtp_rtcp_demuxer_helper_unittest.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtp_rtcp_demuxer_helper.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtp_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtp_stream_receiver_controller.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rtp_transport_controller_send_interface.h Enables feature to account for unacknowledged data. 2019-01-23 10:00:52 +00:00
rtp_transport_controller_send.cc Removes all const Clock*. 2019-01-30 13:03:37 +00:00
rtp_transport_controller_send.h Removes all const Clock*. 2019-01-30 13:03:37 +00:00
rtp_video_sender_interface.h Move FecController to RtpVideoSender. 2018-10-05 14:39:01 +00:00
rtp_video_sender_unittest.cc Removes usages of TaskQueueCongestionControl field trial. 2019-02-01 09:46:59 +00:00
rtp_video_sender.cc Move FrameCountObserver from RTPSender to RtpVideoSender 2019-01-29 09:31:11 +00:00
rtp_video_sender.h Move FrameCountObserver from RTPSender to RtpVideoSender 2019-01-29 09:31:11 +00:00
rtx_receive_stream_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulated_network.cc Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
simulated_network.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
simulated_packet_receiver.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
ssrc_binding_observer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
video_receive_stream.cc Enable use of MediaTransportInterface for video streams. 2019-01-11 14:06:15 +00:00
video_receive_stream.h Expose video freeze metrics in GetStats. 2019-02-04 09:58:08 +00:00
video_send_stream.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
video_send_stream.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00