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webrtc_m130/webrtc/modules/video_coding
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Henrik Kjellander 1bc3146e08 Disable more VideoProcessorIntegrationTest tests on Linux 32-bit 
The previously disabled tests on TSan and UBSan are also failing on
Linux 32-bit.

BUG=webrtc:6781
R=ehmaldonado@webrtc.org
TBR=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/2535913002 .

Cr-Commit-Position: refs/heads/master@{#15276}
2016-11-28 18:34:43 +00:00
..
codecs
Disable more VideoProcessorIntegrationTest tests on Linux 32-bit
2016-11-28 18:34:43 +00:00
include
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
2016-11-25 12:37:06 +00:00
test
Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
2016-11-25 14:40:30 +00:00
utility
Fix perf regression in screenshare temporal layer bitrate allocation
2016-11-25 16:09:51 +00:00
BUILD.gn
Move smoothing filter to common audio and exp_filter to base/analytics.
2016-11-24 19:01:14 +00:00
codec_database.cc
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
2016-11-25 18:06:35 +00:00
codec_database.h
Fix stats for encoder target bitrate when target rate is zero.
2016-07-05 15:34:08 +00:00
codec_timer.cc
Reland of VCMCodecTimer: Change filter from max to 95th percentile (patchset #1 id:1 of https://codereview.webrtc.org/1808693002/ )
2016-03-22 12:12:12 +00:00
codec_timer.h
Reland of VCMCodecTimer: Change filter from max to 95th percentile (patchset #1 id:1 of https://codereview.webrtc.org/1808693002/ )
2016-03-22 12:12:12 +00:00
decoding_state_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
decoding_state.cc
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
decoding_state.h
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
DEPS
Pass selected cricket::VideoCodec down to internal H264 encoder
2016-11-28 15:20:26 +00:00
encoded_frame.cc
Delete unused support for vp8 partitions.
2016-09-30 10:43:07 +00:00
encoded_frame.h
Delete unused support for vp8 partitions.
2016-09-30 10:43:07 +00:00
fec_tables_xor.h
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
frame_buffer2_unittest.cc
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
frame_buffer2.cc
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
frame_buffer2.h
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
frame_buffer.cc
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
frame_buffer.h
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
frame_object.cc
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
frame_object.h
RtpFrameObject::GetCodecHeader now return rtc::Optional<RTPVideoTypeHeader>
2016-11-03 15:57:03 +00:00
generic_decoder.cc
Change DCHECK in VCMDecodedFrameCallback back to just logging.
2016-10-03 15:54:44 +00:00
generic_decoder.h
Deprecate VideoDecoder::Reset() and remove calls.
2016-02-02 14:40:13 +00:00
generic_encoder.cc
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
generic_encoder.h
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
h264_sps_pps_tracker_unittest.cc
H264SpsPpsTracker now return PacketAction on CopyAndFixBitstream.
2016-11-07 13:11:44 +00:00
h264_sps_pps_tracker.cc
H264SpsPpsTracker now return PacketAction on CopyAndFixBitstream.
2016-11-07 13:11:44 +00:00
h264_sps_pps_tracker.h
H264SpsPpsTracker now return PacketAction on CopyAndFixBitstream.
2016-11-07 13:11:44 +00:00
histogram_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
histogram.cc
Moved sequence number specific operations from mod_ops.h
2016-03-22 09:03:55 +00:00
histogram.h
Revert "Revert of Implement the NackModule as part of the new jitter buffer. (patchset #19 id:360001 of https://codereview.webrtc.org/1715673002/ )"
2016-03-08 11:36:22 +00:00
inter_frame_delay.cc
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
inter_frame_delay.h
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
internal_defines.h
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
jitter_buffer_common.h
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
jitter_buffer_unittest.cc
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
jitter_buffer.cc
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
jitter_buffer.h
Avoid race in VideoReceiveStream shutdown
2016-07-13 17:57:20 +00:00
jitter_estimator_tests.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
jitter_estimator.cc
Prevent zero division in VCMJitterEstimator.
2016-02-01 15:31:17 +00:00
jitter_estimator.h
Wire up VCMTiming in FrameBuffer2.
2016-08-03 08:59:47 +00:00
media_opt_util.cc
Replace c-style cast and constrain value in VCMFecMethod::ProtectionFactor.
2016-11-17 06:59:42 +00:00
media_opt_util.h
Move smoothing filter to common audio and exp_filter to base/analytics.
2016-11-24 19:01:14 +00:00
media_optimization.cc
Remove all old suspension logic.
2016-07-01 07:00:19 +00:00
media_optimization.h
Remove all old suspension logic.
2016-07-01 07:00:19 +00:00
nack_fec_tables.h
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
nack_module_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
nack_module.cc
Nack count returned on OnReceivedPacket.
2016-05-13 09:12:11 +00:00
nack_module.h
Nack count returned on OnReceivedPacket.
2016-05-13 09:12:11 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
packet_buffer.cc
Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called.
2016-11-28 15:15:04 +00:00
packet_buffer.h
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
packet.cc
Remove unused constructor from VCMPacket.
2016-06-29 16:03:20 +00:00
packet.h
Remove unused constructor from VCMPacket.
2016-06-29 16:03:20 +00:00
percentile_filter_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
percentile_filter.cc
Reland of VCMCodecTimer: Change filter from max to 95th percentile (patchset #1 id:1 of https://codereview.webrtc.org/1808693002/ )
2016-03-22 12:12:12 +00:00
percentile_filter.h
Reland of VCMCodecTimer: Change filter from max to 95th percentile (patchset #1 id:1 of https://codereview.webrtc.org/1808693002/ )
2016-03-22 12:12:12 +00:00
protection_bitrate_calculator_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
protection_bitrate_calculator.cc
Remove frame rate and target bitrate from ProtectionBitrateCalculator::SetEncodingData
2016-07-11 08:47:36 +00:00
protection_bitrate_calculator.h
Remove frame rate and target bitrate from ProtectionBitrateCalculator::SetEncodingData
2016-07-11 08:47:36 +00:00
receiver_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
receiver.cc
Remove old WebRTC-NewVideoJitterBuffer used for testing the NackModule.
2016-07-06 14:10:45 +00:00
receiver.h
Remove old WebRTC-NewVideoJitterBuffer used for testing the NackModule.
2016-07-06 14:10:45 +00:00
rtp_frame_reference_finder_unittest.cc
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
rtp_frame_reference_finder.cc
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
rtp_frame_reference_finder.h
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
rtt_filter.cc
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
rtt_filter.h
Lint fix for webrtc/modules/video_coding PART 2!
2015-12-21 12:12:45 +00:00
sequence_number_util_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
sequence_number_util.h
Moved sequence number specific operations from mod_ops.h
2016-03-22 09:03:55 +00:00
session_info_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
session_info.cc
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
session_info.h
Revert "Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )"
2016-10-06 12:04:59 +00:00
timestamp_map.cc
Lint fix for webrtc/modules/video_coding PART 3!
2015-12-21 16:23:29 +00:00
timestamp_map.h
Replace scoped_ptr with unique_ptr in webrtc/modules/video_coding/
2016-02-29 13:52:06 +00:00
timing_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
timing.cc
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
timing.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
video_codec_initializer.cc
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
2016-11-25 12:37:06 +00:00
video_coding_impl.cc
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
video_coding_impl.h
Initial rate allocation should not use fps = 0
2016-11-21 13:42:04 +00:00
video_coding_robustness_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
video_packet_buffer_unittest.cc
Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called.
2016-11-28 15:15:04 +00:00
video_receiver_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
video_receiver.cc
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
video_sender_unittest.cc
Initial rate allocation should not use fps = 0
2016-11-21 13:42:04 +00:00
video_sender.cc
Initial rate allocation should not use fps = 0
2016-11-21 13:42:04 +00:00