This won't be perfect since the peeked value will be noisy, but since we
cap it with the starting rate, it should only improve things.
Bug: webrtc:9718
Change-Id: Id2cf42fb85c8d7126f6d538a3982d65caa7a75b7
Reviewed-on: https://webrtc-review.googlesource.com/c/109926
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25604}
Make averaging test in NtpEstimator less sensitive.
TESTED=Locally patched into chrome and tested on 1st party software and in video_loopback. All produced parameters looked reasonable.
Bug: webrtc:9698
Change-Id: Idc5e80c657ef190dc95da1e27d1288ff9eddd139
Reviewed-on: https://webrtc-review.googlesource.com/c/110500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25603}
When delta encoding, use ToUnsigned() and ToSigned() on
timestamp_ms, since it's a signed type. This is only relevant
for delta-encoding/decoding.
Bug: webrtc:8111
Change-Id: I1fabfcb2be64793c281f5bc0d38a2f8035dd0d18
Reviewed-on: https://webrtc-review.googlesource.com/c/110504
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25602}
Since the number of channels is always greater than 0, smaller
deltas can be accomplished by encoding a sequence of (1, 2, 1)
as if the sequence were (0, 1, 0). This way, wrap around to the
first value is a delta of 1, rahter than a delta of 3.
For simplicity's sake, though at the cost of consistency, we still
encode the base event's number of channels unshifted. We do so
because there are no bits to be gained by doing it otherwise, and
the value there is more likely to be manually inspected, than are
the deltas, so a simpler scheme has merit.
Bug: webrtc:8111
Change-Id: I2d4def67da85c42802fe13cd0494fdd9f2b38f7a
Reviewed-on: https://webrtc-review.googlesource.com/c/110242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25601}
After this CL, all audio encoders have to be injected by the caller.
This means that there is no special "built-in" set of codecs, and
users won't have to pay the binary size and security costs of codecs
they aren't using.
Bug: webrtc:8396
Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6
Reviewed-on: https://webrtc-review.googlesource.com/c/103821
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25600}
This is a preparation for deleting ChannelReceiveProxy, Changes
signature of some methods, and demotes methods OnData and
OnReceivedPayloadData to private.
Bug: webrtc:9801
Change-Id: Ib00a80c6482ed5238f3cc8233860c70f11484df9
Reviewed-on: https://webrtc-review.googlesource.com/c/110606
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25599}
Fake Ice currently does not signal the network route change. Also, it is not aware of the network thread, so added a setter for a network thread.
Bug: None
Change-Id: I25326282f32d36229422eca7368b53ee7b52ec72
Reviewed-on: https://webrtc-review.googlesource.com/c/110363
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25585}
The new event log format makes use of delta encoding to compress
parts of the log.
Bug: webrtc:8111
Change-Id: I7bec839555323a7537dcec831d4ac1d5eb109932
Reviewed-on: https://webrtc-review.googlesource.com/c/109161
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25584}
Remove some redundant lines and Tina who has left the project :(
Bug: None
Change-Id: I8a8cba3c2b13d93e668754fbe6b06daa09095534
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/c/110503
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25583}
The RTC event log analyzer would previously only plot network latency
for incoming video streams. (The latency is computed from the capture
time in the RTP header, and the packet receive time.) This CL adds
support for audio packets, which requires estimating the RTP clock
frequency for the incoming packets.
Bug: None
Change-Id: Idf1ff9febfdd4097976b22a61f1c5679deb6068c
Reviewed-on: https://webrtc-review.googlesource.com/c/108784
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25580}
In a handful of places we wait for the old IceConnectionState to reach some value and then we assume that the new connection states have also been updated. However those are updated in response to different events that might not have fired yet, so sometimes these tests will fail.
This change makes us wait explicitly for those states to update.
Bug: webrtc:9983
Change-Id: I5cb6652ee29c0b86c0834174442140a3863e08e4
Reviewed-on: https://webrtc-review.googlesource.com/c/110441
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25579}
Some targets used to be included transitively via ortc.
Since ortc module has been removed (Bug: webrtc:9824),
this CL explicitly add them in main //:webrtc target.
As a result, the following functions are exposed again:
CreateBuiltinVideoDecoderFactory()
CreateBuiltinVideoEncoderFactory()
CreatePeerConnectionFactory()
[...]
Bug: webrtc:9824
Bug: webrtc:9973
Change-Id: Iebfae582f8887bf76338c73fc85c4608e96c3f0d
Reviewed-on: https://webrtc-review.googlesource.com/c/110248
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25576}
This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25575}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
GetEncoderInfo() is now called every frame, so we should not do
expensive parsing or logging in there. Instead, prepare an EncoderInfo
instance in InitEncode() and just return that in GetEncoderInfo().
Bug: webrtc:9890
Change-Id: Idc9e79e681c6f7ff4f9b446aa298c156f25bc6f6
Reviewed-on: https://webrtc-review.googlesource.com/c/110161
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25569}
This will allow clients to include only the software codecs they need
rather than being forced to bundle them all.
- libjingle_peerconnection_jni keeps its allow_poison for now, until
dependent targets bundle their own codecs explicitly.
- native_api_codecs and native_api_video lose their allow_poison
because dependent targets are already bundling codecs explicitly.
- libjingle_peerconnection_metrics_default_jni and
native_api_peerconnection lose their allow_poison because they
were not actually poisoned.
legacy_hwcodecs_jni and default_video_codec_factory_jni exist for
clients that want to continue bundling the same codecs they get by
default today.
Bug: webrtc:7925
Change-Id: Idf853a6bc77f43decd35ad2a0f467937fec8f8b5
Reviewed-on: https://webrtc-review.googlesource.com/c/108221
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25564}
