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6 Commits

Author SHA1 Message Date
Saurav Das
7262fc29a0 Refactor Rtp Receivers to accept SSRC 0. 
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.

Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
 2019-09-16 21:29:58 +00:00
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Ruslan Burakov
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
Ruslan Burakov
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
Ruslan Burakov
493a650b1e Propagate base minimum delay from video jitter buffer to webrtc/api. 
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.


Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
 2019-02-27 15:08:34 +00:00
Ruslan Burakov
501bfba0cb Split rtp_receiver for readability. 
Bug: webrtc:10304
Change-Id: I85e421060d8560cf36cdb5970ae190efc77e7709
Reviewed-on: https://webrtc-review.googlesource.com/c/122085
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26634}
 2019-02-11 12:47:51 +00:00