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webrtc/call
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Niels Möller 74e5f804e4 Make BitrateAllocator::transmission_max_bitrate_multiplier_ const. 
To make this work, also make the private method
GetTransmissionMaxBitrateMultiplier static.

Bug: None
Change-Id: I1777602b89d404ecbc9956acf9a9c7f852d69911
Reviewed-on: https://webrtc-review.googlesource.com/72440
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23023}
2018-04-25 13:28:13 +00:00
..
test Moving demux from FakeNetworkPipe to DirectTransport. 2018-04-25 10:13:03 +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 Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_send_stream.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
audio_send_stream.h Wire up MID send value to the PeerConnection API 2018-03-26 18:14:30 +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 chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
bitrate_allocator.cc Make BitrateAllocator::transmission_max_bitrate_multiplier_ const. 2018-04-25 13:28:13 +00:00
bitrate_allocator.h Make BitrateAllocator::transmission_max_bitrate_multiplier_ const. 2018-04-25 13:28:13 +00:00
bitrate_allocator_unittest.cc Uses config struct with bitrate allocator. 2018-04-19 09:15:33 +00:00
bitrate_constraints.cc Reland "Moved BitrateConfig out of Call::Config." 2018-02-21 11:38:42 +00:00
bitrate_constraints.h Reland "Moved BitrateConfig out of Call::Config." 2018-02-21 11:38:42 +00:00
bitrate_estimator_tests.cc Reland "Move creating encoder to VideoStreamEncoder." 2018-04-19 08:48:58 +00:00
BUILD.gn Moving demux from FakeNetworkPipe to DirectTransport. 2018-04-25 10:13:03 +00:00
call.cc Reland "Reland "Move rtp-specific config out of EncoderSettings."" 2018-04-05 14:30:09 +00:00
call.h Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
call_config.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
call_config.h Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
call_perf_tests.cc Move BitrateAllocation to api/ and rename it VideoBitrateAllocation 2018-04-23 15:31:27 +00:00
call_unittest.cc Removed fake rtp transport controller send. 2018-02-21 13:39:16 +00:00
callfactory.cc Reland: Add ability to emulate degraded network in Call via field trial 2018-03-14 17:03:25 +00:00
callfactory.h Rename Call::Config to CallConfig, keep old name as alias. 2018-02-14 15:14:39 +00:00
degraded_call.cc Fix race in DegradedCall::DestroyVideoSendStream 2018-03-16 10:23:47 +00:00
degraded_call.h Reland: Add ability to emulate degraded network in Call via field trial 2018-03-14 17:03:25 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Using microseconds in FakeNetworkPipe. 2018-04-25 10:32:23 +00:00
fake_network_pipe.h Using microseconds in FakeNetworkPipe. 2018-04-25 10:32:23 +00:00
flexfec_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
flexfec_receive_stream.h Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
flexfec_receive_stream_impl.cc replace stringstream in call/ 2018-04-04 16:09:15 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Added nisse@webrtc.org as owner in call. 2018-02-20 09:39:51 +00:00
rampup_tests.cc Reland "Reland "Move rtp-specific config out of EncoderSettings."" 2018-04-05 14:30:09 +00:00
rampup_tests.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
receive_time_calculator.cc Fix of ReceiveTimeCalculator field trial parsing. 2018-03-24 12:36:17 +00:00
receive_time_calculator.h Added receive time calculator under field trial. 2018-03-21 15:40:39 +00:00
receive_time_calculator_unittest.cc Added receive time calculator under field trial. 2018-03-21 15:40:39 +00:00
rtcp_demuxer.cc Fix circular dependencies in webrtc_common. 2017-12-15 14:33:26 +00:00
rtcp_demuxer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcp_demuxer_unittest.cc Fix circular dependencies in webrtc_common. 2017-12-15 14:33:26 +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.cc Extracted bitrate configuration from call class. 2018-02-21 12:33:02 +00:00
rtp_bitrate_configurator.h Moved bitrate config interface from Call class. 2018-02-21 15:03:45 +00:00
rtp_bitrate_configurator_unittest.cc Extracted bitrate configuration from call class. 2018-02-21 12:33:02 +00:00
rtp_config.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
rtp_config.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_demuxer.cc Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_demuxer.h Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_demuxer_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +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_rtcp_demuxer_helper.cc Optional: Use nullopt and implicit construction in /call 2018-01-11 12:00:00 +00:00
rtp_rtcp_demuxer_helper.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_stream_receiver_controller.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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_transport_controller_send.cc Log SendSideCongestionController experiment 2018-04-13 12:33:48 +00:00
rtp_transport_controller_send.h Fix of race on access to send side congestion controller. 2018-03-26 13:48:10 +00:00
rtp_transport_controller_send_interface.h Reporting feedback availability to congestion controller. 2018-03-22 14:11:54 +00:00
rtx_receive_stream.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtx_receive_stream_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ssrc_binding_observer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_config.cc Add member VideoEncoderConfig::SdpVideoFormat. 2018-04-17 11:38:23 +00:00
video_config.h Add member VideoEncoderConfig::SdpVideoFormat. 2018-04-17 11:38:23 +00:00
video_receive_stream.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
video_receive_stream.h Move videosinkinterface.h to common_video to solve a circular dep. 2018-01-04 13:19:49 +00:00
video_send_stream.cc Reland "Move creating encoder to VideoStreamEncoder." 2018-04-19 08:48:58 +00:00
video_send_stream.h Delete unused VideoSendStream::Config::EncoderSettings::encoder. 2018-04-19 13:53:46 +00:00