Danil Chapovalov b3179c75ed Remove RTPSender::SetSendPayloadType ...

Bug: None
Change-Id: Id99c9eda5e377de68c8bff053511534c66bd60a0
Reviewed-on: https://webrtc-review.googlesource.com/63801
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22559}

2018-03-22 10:48:34 +00:00

..

audio_coding

Move asm_defines.h to rtc_base/system/

2018-03-22 03:12:13 +00:00

audio_device

Delete unneeded includes of wav_file.h and file_wrapper.h.

2018-03-20 15:59:27 +00:00

audio_mixer

Change place of UMA logging in AudioMixer.

2018-03-19 14:10:51 +00:00

audio_processing

Further headset mode robustification based on linear filter convergence

2018-03-22 09:51:14 +00:00

bitrate_controller

Preparing for task queue in congenstion controller

2018-02-20 12:35:15 +00:00

congestion_controller

Disabling periodic tasks on SSCC in unit tests.

2018-03-21 13:50:38 +00:00

desktop_capture

Revert "Add ScopedAllowBaseSyncPrimitives for DesktopConfigurationMonitor."

2018-01-31 08:32:19 +00:00

include

Revert "Revert "Enables PeerConnectionFactory using external fec controller""

2018-02-20 12:41:55 +00:00

pacing

Added congestion control functionality to pacer.

2018-03-09 17:40:24 +00:00

remote_bitrate_estimator

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

rtp_rtcp

Remove RTPSender::SetSendPayloadType

2018-03-22 10:48:34 +00:00

utility

Delete unused MediaFile module.

2018-01-29 11:18:18 +00:00

video_capture

Update gn files to support Mozilla build

2018-01-16 07:51:23 +00:00

video_coding

Add field VideoEncoderConfig::codec_type.

2018-03-22 09:39:13 +00:00

video_processing

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

BUILD.gn

Cleaning up modules_tests resources.

2018-03-09 12:57:24 +00:00

module_common_types_unittest.cc

Remove various IDs:

2017-09-28 14:37:11 +00:00

OWNERS

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