mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 22:00:47 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_coding/acm2
History
Chen Xing 3e8ef940fe Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. 
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.

Bug: webrtc:10668
Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28434}
2019-07-01 15:56:40 +00:00
..
acm_receive_test.cc Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
acm_receive_test.h 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 2019-02-22 09:59:01 +00:00
acm_receiver.cc Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. 2019-07-01 15:56:40 +00:00
acm_receiver.h Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
acm_receiver_unittest.cc Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData 2019-04-26 12:58:14 +00:00
acm_resampler.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
acm_resampler.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
acm_send_test.cc Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData 2019-04-26 12:58:14 +00:00
acm_send_test.h Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData 2019-04-26 12:58:14 +00:00
audio_coding_module.cc Audio coding: Don't choke when RTP timestamp rate > sample rate 2019-05-21 03:10:49 +00:00
audio_coding_module_unittest.cc Reland "Reland "Change buffer level filter to store current level in number of samples."" 2019-06-27 09:16:27 +00:00
call_statistics.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
call_statistics.h Move NetworkStatistics and AudioDecodingCallStats from common_types.h 2018-11-19 11:55:34 +00:00
call_statistics_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00