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webrtc_m130/test
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philipel 32134470ca Actually input packets into the PacketBuffer. 
Also prepared it for comming changes to RTPVideoHeader.

Bug: none
Change-Id: I60e57f779aee4ee98b6eec3d6631e43ac682c37f
Reviewed-on: https://webrtc-review.googlesource.com/87825
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23911}
2018-07-10 15:16:30 +00:00
..
android
Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199)
2018-02-22 13:58:58 +00:00
fuzzers
Actually input packets into the PacketBuffer.
2018-07-10 15:16:30 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call_test.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Remove voe_auto_test and add new tests to cover the missing cases.
2017-09-15 16:56:08 +00:00
constants.h
Remove voe_auto_test and add new tests to cover the missing cases.
2017-09-15 16:56:08 +00:00
DEPS
Reland "Add multiplex case to webrtc_perf_tests"
2018-03-10 01:21:04 +00:00
direct_transport_unittest.cc
Moving demux from FakeNetworkPipe to DirectTransport.
2018-04-25 10:13:03 +00:00
direct_transport.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
direct_transport.h
Moving demux from FakeNetworkPipe to DirectTransport.
2018-04-25 10:13:03 +00:00
drifting_clock.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
drifting_clock.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
encoder_proxy_factory.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
fake_texture_frame.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
gtest.h
Reland "Adding gtest-spi.h in webrtc/test/gtest.h"
2018-04-05 08:21:23 +00:00
layer_filtering_transport.cc
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mock_audio_decoder.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Removed Die mock from MockAudioEncoder
2018-02-22 12:53:38 +00:00
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
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
2018-03-09 16:04:35 +00:00
mock_transport.h
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets"
2018-03-09 16:04:35 +00:00
null_platform_renderer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
null_transport.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
null_transport.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Remove pbos@webrtc.org from all OWNERS.
2017-11-01 08:03:46 +00:00
rtcp_packet_parser.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtcp_packet_parser.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_file_reader_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_file_writer.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_file_writer.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
single_threaded_task_queue.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
single_threaded_task_queue.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
statistics.cc
Reland "Updated analysis in videoprocessor."
2018-01-18 08:37:27 +00:00
statistics.h
Reland "Updated analysis in videoprocessor."
2018-01-18 08:37:27 +00:00
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
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
video_renderer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00