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webrtc_m130/media/base
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Åsa Persson fcf79cca7b Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. 
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp

Partial implementation: currently only populated when a/v sync is enabled.

Bug: webrtc:7065
Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29581}
2019-10-23 07:46:39 +00:00
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adapted_video_track_source.cc
Cleanup video frame metadata copying
2019-06-05 15:05:57 +00:00
adapted_video_track_source.h
…
audio_source.h
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codec_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
codec.cc
Add support for enabling and negotiating raw RTP packetization.
2019-06-04 14:35:54 +00:00
codec.h
Add support for enabling and negotiating raw RTP packetization.
2019-06-04 14:35:54 +00:00
delayable.h
…
fake_frame_source.cc
…
fake_frame_source.h
Format almost everything.
2019-07-08 13:45:15 +00:00
fake_media_engine.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
fake_media_engine.h
Add/remove receive streams with SSRC 0 from media channels
2019-10-07 23:01:28 +00:00
fake_network_interface.h
Only handle each RTCP once.
2019-09-18 16:54:39 +00:00
fake_rtp.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
fake_rtp.h
Format almost everything.
2019-07-08 13:45:15 +00:00
fake_video_renderer.cc
Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.
2019-06-20 10:24:29 +00:00
fake_video_renderer.h
Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.
2019-06-20 10:24:29 +00:00
h264_profile_level_id.cc
…
h264_profile_level_id.h
…
media_channel.cc
…
media_channel.h
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
media_config.h
…
media_constants.cc
Add support for enabling and negotiating raw RTP packetization.
2019-06-04 14:35:54 +00:00
media_constants.h
Add support for enabling and negotiating raw RTP packetization.
2019-06-04 14:35:54 +00:00
media_engine.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
media_engine.h
Delete CoreAudio include from media_engine.h
2019-09-13 12:27:19 +00:00
rid_description.cc
…
rid_description.h
…
rtp_data_engine_unittest.cc
Move MediaTransportInterface out of the libjingle_peerconnection_api target
2019-09-13 10:49:56 +00:00
rtp_data_engine.cc
Add/remove receive streams with SSRC 0 from media channels
2019-10-07 23:01:28 +00:00
rtp_data_engine.h
Add/remove receive streams with SSRC 0 from media channels
2019-10-07 23:01:28 +00:00
rtp_utils_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_utils.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_utils.h
…
stream_params_unittest.cc
…
stream_params.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
stream_params.h
Format almost everything.
2019-07-08 13:45:15 +00:00
test_utils.cc
…
test_utils.h
…
turn_utils_unittest.cc
…
turn_utils.cc
…
turn_utils.h
…
video_adapter_unittest.cc
Add option to make first scale factor depend on input resolution.
2019-10-18 14:19:04 +00:00
video_adapter.cc
Add option to make first scale factor depend on input resolution.
2019-10-18 14:19:04 +00:00
video_adapter.h
Add option to make first scale factor depend on input resolution.
2019-10-18 14:19:04 +00:00
video_broadcaster_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
video_broadcaster.cc
…
video_broadcaster.h
…
video_common_unittest.cc
…
video_common.cc
…
video_common.h
…
video_source_base.cc
…
video_source_base.h
…
vp9_profile.cc
…
vp9_profile.h
…