webrtc/audio
Taylor Brandstetter 84916937b7 Update packetsLost and jitter stats any time a packet is received.
Before this CL, the packetsLost and jitter stats (as returned by
GetStats, at the API level) were only being updated when an RTCP SR or
RR is generated. According to the stats spec, "local" stats like this
should be updated any time a packet is received.

This CL also fixes some minor issues with the calculation of packetsLost
(and fractionLost):
* Packets weren't being count as lost if lost over a sequence number
  rollover.
* Temporary periods of "negative" loss (caused by duplicate or out of
  order packets) weren't being accumulated into the cumulative loss
  counter. Example:
  Period 1: Received packets 1, 2, 4
    Loss over that period: 1 (expected 4 packets, got 3)
    Reported cumulative loss: 1
  Period 2: Received packets 3, 5
    Loss over that period: -1 (expected 1 packet, got 2)
    Reported cumulative loss: 1 (should be 0!)

Landing with NOTRY because Android compile bots are broken for an
unrelated reason.
NOTRY=True

Bug: webrtc:8804
Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
Reviewed-on: https://webrtc-review.googlesource.com/50020
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23731}
2018-06-25 23:56:39 +00:00
..
test Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
utility Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_receive_stream.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
audio_receive_stream_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_send_stream.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_send_stream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_state.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_state.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_state_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_transport_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_transport_impl.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
channel.cc Update packetsLost and jitter stats any time a packet is received. 2018-06-25 23:56:39 +00:00
channel.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
channel_proxy.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
channel_proxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
null_audio_poller.cc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
null_audio_poller.h Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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 Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
transport_feedback_packet_loss_tracker.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
transport_feedback_packet_loss_tracker_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00