Commit graph

8 commits

Author SHA1 Message Date
Danil Chapovalov
702820d087 Update rtc tools to create Call using Environment
Bug: webrtc:15656
Change-Id: I2bf17e95f4c8b26dcb735eb5198d5347a97dee69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329082
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41261}
2023-11-28 19:48:27 +00:00
Harald Alvestrand
d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces.
This CL takes away all usages except for Android code.

Low-Coverage-Reason: Refactoring old code
Bug: webrtc:15410
Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40554}
2023-08-15 13:20:21 +00:00
Jonas Oreland
6c2dae21e9 Move VideoEncoderConfig from api/ into video/config
This cl move VideoEncoderConfig from api/ to video/config.

VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.

brandt@ think that the reason these were in api/ in the
first place had to downstream project.

Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).

Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
2022-09-29 09:44:43 +00:00
Niels Möller
de95329daa Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.

Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
2020-09-29 10:19:20 +00:00
Artem Titov
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
Danil Chapovalov
83bbe91398 Delete deprecated rtc_event_log header
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 10:58:17 +00:00
Ilya Nikolaevskiy
ca160215c9 Update rtc_tools/rtp_generator to compile
This is a trivial CL, updating rtp_generator.cc according to changes in
APIs in other places.

Bug: webrtc:10807
Change-Id: Ie85c6283f2d78dcf742979378db0b4fb0914c96c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145209
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28526}
2019-07-10 15:17:29 +00:00
Benjamin Wright
87bbb91469 Add rtp_generator utility to rtc_tools.
This CL introduces a new rtp_generator tool that can be utilized to generate
.rtpdump files that can be replayed by the video_replayer. This allows
automated generation of corpus material for the new WebRTC RTP fuzzers in
addition to allowing anyone who is experimenting with a new RTP feature to
quickly debug issues.

It can be used as follows:
./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump
./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump

It works by generating squares randomly on the screen for a given duration. This
initial version is very limited and doesn't support FEC, RED and other
configurations. I plan to extend it to support these in future CLs.

Bug: webrtc:10117
Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51
Reviewed-on: https://webrtc-review.googlesource.com/c/119964
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26517}
2019-02-01 18:36:19 +00:00