admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/test
History
Danil Chapovalov 916ec7dadf Add Generic frame descritpor header extension 
to list of extensions supported by RtpPacket.

Bug: webrtc:9361
Change-Id: Iabee824381be3776e17e95f32507058607fc0bc8
Reviewed-on: https://webrtc-review.googlesource.com/85346
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23788}
2018-06-29 15:02:44 +00:00
..
android
…
fuzzers
Add Generic frame descritpor header extension
2018-06-29 15:02:44 +00:00
gl
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
ios
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
linux
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mac
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testsupport
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
win
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
BUILD.gn
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
2018-06-21 15:57:43 +00:00
call_test.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
call_test.h
Reland "Move creating encoder to VideoStreamEncoder."
2018-04-19 08:48:58 +00:00
configurable_frame_size_encoder.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
configurable_frame_size_encoder.h
Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
2018-04-23 15:31:27 +00:00
constants.cc
…
constants.h
…
DEPS
…
direct_transport_unittest.cc
Moving demux from FakeNetworkPipe to DirectTransport.
2018-04-25 10:13:03 +00:00
direct_transport.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
direct_transport.h
Moving demux from FakeNetworkPipe to DirectTransport.
2018-04-25 10:13:03 +00:00
drifting_clock.cc
…
drifting_clock.h
…
encoder_proxy_factory.h
Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
2018-04-23 15:31:27 +00:00
encoder_settings.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
encoder_settings.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fake_decoder.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fake_decoder.h
Delete pre_decode_callback.
2018-06-20 07:04:09 +00:00
fake_encoder.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fake_encoder.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fake_texture_frame.cc
…
fake_texture_frame.h
…
fake_videorenderer.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
field_trial.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
field_trial.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
frame_generator_capturer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
frame_generator_capturer.h
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
frame_generator_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
frame_generator.cc
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
frame_generator.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
frame_utils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
frame_utils.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
function_video_decoder_factory.h
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl}
2018-05-31 11:48:17 +00:00
function_video_encoder_factory.h
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl}
2018-05-31 11:48:17 +00:00
gmock.h
…
gtest.h
Reland "Adding gtest-spi.h in webrtc/test/gtest.h"
2018-04-05 08:21:23 +00:00
layer_filtering_transport.cc
Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.
2018-06-21 17:53:35 +00:00
layer_filtering_transport.h
Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.
2018-06-21 17:53:35 +00:00
mock_audio_decoder_factory.h
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
mock_audio_decoder.cc
…
mock_audio_decoder.h
…
mock_audio_encoder_factory.h
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
mock_audio_encoder.cc
…
mock_audio_encoder.h
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
mock_transport.cc
…
mock_transport.h
…
null_platform_renderer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
null_transport.cc
…
null_transport.h
…
OWNERS
…
rtcp_packet_parser.cc
…
rtcp_packet_parser.h
…
rtp_file_reader_unittest.cc
…
rtp_file_reader.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_file_reader.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_file_writer_unittest.cc
…
rtp_file_writer.cc
…
rtp_file_writer.h
…
rtp_rtcp_observer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
run_loop.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
run_loop.h
…
run_test.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
run_test.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
single_threaded_task_queue_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
single_threaded_task_queue.cc
…
single_threaded_task_queue.h
…
statistics.cc
…
statistics.h
…
test_main.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
vcm_capturer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
vcm_capturer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_capturer.cc
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
video_capturer.h
Replace rtc::Optional with absl::optional in test and rtc_tools
2018-06-18 13:15:23 +00:00
video_codec_settings.h
Delete RTP-specific values from the VideoCodecType enum.
2018-06-07 07:49:27 +00:00
video_renderer.cc
…
video_renderer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00