412a31bbf8
Insert frame transformer between Depacketizer and Decoder.
...
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.
The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130 .
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30654}
2020-03-02 08:33:44 +00:00
a837030f8f
Split out RtpSource from libjingle_peerconnection_api
...
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 14:04:47 +00:00
015c3cbf51
Remove deprecated constructors of RtpSource
...
Bug: webrtc:10650
Change-Id: I1dee27252068ad33e62978ee3a3b3f60b266a2c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149220
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28883}
2019-08-16 20:56:56 +00:00
b5d918324c
Add RTP timestamp to contributing sources
...
RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.
Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28020}
2019-05-22 08:53:08 +00:00
428dcb2517
Remove SetLatency/GetLatency from MediaSourceInterface API level
...
Bug: webrtc:10287
Change-Id: I74fad31db98b75791085688438064f9510b0b6fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165
Commit-Queue: Ruslan Burakov <kuddai@google.com >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27692}
2019-04-18 19:11:31 +00:00
4bac79ece2
Add SetJitterBufferMinimumDelay method to RtpReceiverInterface.
...
This change is required to allow modification of Jitter Buffer delay
in javascript via Origin Trial Experiment.
Link to experiment description:
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/Tgm4qiNepJc
Bug: webrtc:10287
Change-Id: I4f21380aad5982a4a60c55683b5173ce72ce0392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131144
Commit-Queue: Ruslan Burakov <kuddai@google.com >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27444}
2019-04-04 09:00:16 +00:00
4a7b3acfcf
Add DTLSTransport info into sender/receiver state.
...
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26289}
2019-01-17 10:21:32 +00:00
10542f21c8
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
...
Mechanically generated by running this command:
tools_webrtc/do-renames.sh update all-renames.txt && git cl format
Then manually updating:
tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
1c05765831
(3) Rename files to snake_case: move the files
...
Mechanically generated with this command:
tools_webrtc/do-rename.sh move all-renames.txt
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
