mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-20 09:07:52 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/audio
History
Sebastian Jansson b686396ec6 Makes AudioSendStream signal that it's part of allocation. 
This adds calls to the underlying RtpRtcp module to indicate when audio
is part of bitrate allocation. This information is propagated and set in
the packet info for each packet.

This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.

Bug: webrtc:9796
Change-Id: I79b024cb7f2eb8c86421cfa34d38ef68467776c3
Reviewed-on: https://webrtc-review.googlesource.com/c/104882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25086}
2018-10-10 09:47:46 +00:00
..
test Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig 2018-10-08 12:19:31 +00:00
utility Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_level.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_level.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
audio_receive_stream.cc Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
audio_receive_stream.h Split ChannelProxy into send and receive classes. 2018-10-04 06:40:27 +00:00
audio_receive_stream_unittest.cc Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
audio_send_stream.cc Makes AudioSendStream signal that it's part of allocation. 2018-10-10 09:47:46 +00:00
audio_send_stream.h Split ChannelProxy into send and receive classes. 2018-10-04 06:40:27 +00:00
audio_send_stream_tests.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_send_stream_unittest.cc Enable End-to-End Encrypted Audio Payloads. 2018-10-04 22:08:34 +00:00
audio_state.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
audio_state.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
audio_state_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_transport_impl.cc Remove echo_cancellation() and echo_control_mobile() interface access outside APM 2018-09-27 12:28:12 +00:00
audio_transport_impl.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Split voe::Channel into ChannelSend and ChannelReceive 2018-10-04 13:33:38 +00:00
channel_receive.cc Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
channel_receive.h Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
channel_receive_proxy.cc Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
channel_receive_proxy.h Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
channel_send.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
channel_send.h Enable End-to-End Encrypted Audio Payloads. 2018-10-04 22:08:34 +00:00
channel_send_proxy.cc Enable End-to-End Encrypted Audio Payloads. 2018-10-04 22:08:34 +00:00
channel_send_proxy.h Enable End-to-End Encrypted Audio Payloads. 2018-10-04 22:08:34 +00:00
conversion.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
mock_voe_channel_proxy.h Set ChannelReceive transport at construction time. 2018-10-05 10:56:40 +00:00
null_audio_poller.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
null_audio_poller.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
OWNERS
remix_resample.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
remix_resample.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
remix_resample_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
time_interval.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
time_interval_unittest.cc Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
transport_feedback_packet_loss_tracker.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
transport_feedback_packet_loss_tracker.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
transport_feedback_packet_loss_tracker_unittest.cc Cleanup modules_common_types 2018-09-18 08:08:33 +00:00