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platform-external-webrtc/media/base
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Ruslan Burakov 8af8896596 Expose jitter buffer flushes metric in new getStats api. 
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}
2018-11-23 11:41:43 +00:00
..
adaptedvideotracksource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
adaptedvideotracksource.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
audiosource.h
…
codec_unittest.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.cc
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-25 14:13:44 +00:00
codec.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
device.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakeframesource.cc
Add sanity checks to UpdateDelayStatistics and patch unit tests.
2018-09-24 23:13:02 +00:00
fakeframesource.h
Add sanity checks to UpdateDelayStatistics and patch unit tests.
2018-09-24 23:13:02 +00:00
fakemediaengine.cc
Removes templating from CompositeMediaEngine.
2018-11-16 10:31:57 +00:00
fakemediaengine.h
Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
2018-11-20 14:03:18 +00:00
fakenetworkinterface.h
Add implicit conversion between rtc:PacketTime and int64_t.
2018-11-01 14:28:24 +00:00
fakertp.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertp.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakevideocapturer.cc
Fix flaky unit test in rtc_unittests
2018-10-23 12:23:16 +00:00
fakevideocapturer.h
Introduce TaskQueueForTest.
2018-02-14 15:32:49 +00:00
fakevideorenderer.cc
Enable clang::find_bad_constructs for sdk/ (part 1).
2018-07-26 12:16:31 +00:00
fakevideorenderer.h
Enable clang::find_bad_constructs for sdk/ (part 1).
2018-07-26 12:16:31 +00:00
h264_profile_level_id.cc
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
h264_profile_level_id.h
Export symbols needed by the Chromium component build (part 4).
2018-10-18 08:42:22 +00:00
mediachannel.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
mediachannel.h
Expose jitter buffer flushes metric in new getStats api.
2018-11-23 11:41:43 +00:00
mediaconfig.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
mediaconstants.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediaconstants.h
Export symbols needed by the Chromium component build (part 4).
2018-10-18 08:42:22 +00:00
mediaengine.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
mediaengine.h
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
rtpdataengine_unittest.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtpdataengine.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtpdataengine.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.h
Reland "Export symbols needed by the Chromium component build (part 1)."
2018-10-11 09:50:21 +00:00
streamparams_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.cc
Delete unused code from media/base/testutils.{cc,h}
2018-10-18 16:06:33 +00:00
testutils.h
Delete unused code from media/base/testutils.{cc,h}
2018-10-18 16:06:33 +00:00
turnutils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
turnutils.cc
…
turnutils.h
Reland "Export symbols needed by the Chromium component build (part 1)."
2018-10-11 09:50:21 +00:00
videoadapter_unittest.cc
Add support to adapt video without preserving aspect ratio
2018-10-26 12:30:32 +00:00
videoadapter.cc
Add support to adapt video without preserving aspect ratio
2018-10-26 12:30:32 +00:00
videoadapter.h
Add support to adapt video without preserving aspect ratio
2018-10-26 12:30:32 +00:00
videobroadcaster_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
videocapturer_unittest.cc
Delete unused code from media/base/testutils.{cc,h}
2018-10-18 16:06:33 +00:00
videocapturer.cc
Enable video adaptation for all screenshare content
2018-08-24 12:02:03 +00:00
videocapturer.h
Export symbols needed by the Chromium component build (part 4).
2018-10-18 08:42:22 +00:00
videocapturerfactory.h
…
videocommon_unittest.cc
…
videocommon.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocommon.h
Export symbols needed by the Chromium component build (part 5).
2018-10-19 09:16:07 +00:00
videosourcebase.cc
Enable and fix chromium clang warnings in sdk/android targets.
2018-04-05 11:22:03 +00:00
videosourcebase.h
Move VideoStreamEncoderInterface to api/.
2018-05-21 19:50:37 +00:00
vp9_profile.cc
vp9: change x-google-profile-id to profile-id
2018-10-10 12:24:33 +00:00
vp9_profile.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00