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webrtc_m130/webrtc/modules/video_coding
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sprang 5271ea6571 Switch temporal layer impl used for screenshare upper simulcast stream. 
Use default temporal layers instead of the RealtimeTemporalLayers one.
When using low fps, having a single stream with much higher bitrate than
the lower simulcast stream causes poor rampup behavior.

Tested manually.

BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2723983002
Cr-Commit-Position: refs/heads/master@{#16934}
2017-03-01 09:58:17 +00:00
..
codecs
Switch temporal layer impl used for screenshare upper simulcast stream.
2017-03-01 09:58:17 +00:00
include
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
test
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
utility
Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ )
2017-02-07 15:02:22 +00:00
BUILD.gn
Enable include check for webrtc_h264 target
2017-02-28 08:28:04 +00:00
codec_database.cc
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +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
Rename base/analytics/ to base/numerics/
2017-01-13 17:14:33 +00:00
decoding_state_unittest.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
decoding_state.cc
Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error.
2016-11-29 15:01:29 +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
Add support for creating HW codecs in the VideoProcessor tests.
2017-02-20 12:35:52 +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_rate_table.h
Rename fec_tables_xor.h -> fec_rate_table.h
2017-01-12 14:16:24 +00:00
frame_buffer2_unittest.cc
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
frame_buffer2.cc
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
frame_buffer2.h
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
frame_buffer.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +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
Don't update the jitter estimate with frames containing retransmitted packets.
2017-01-25 16:56:23 +00:00
frame_object.h
Don't update the jitter estimate with frames containing retransmitted packets.
2017-01-25 16:56:23 +00:00
generic_decoder.cc
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
generic_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
generic_encoder.cc
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
generic_encoder.h
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
h264_sprop_parameter_sets_unittest.cc
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
2017-01-02 16:42:32 +00:00
h264_sprop_parameter_sets.cc
Unit test out of band H264 SPS,PPS within RtpStreamReceiver.
2017-01-24 12:38:27 +00:00
h264_sprop_parameter_sets.h
Delete unneeded includes of base/common.h.
2017-01-25 09:47:24 +00:00
h264_sps_pps_tracker_unittest.cc
Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
2017-01-24 10:38:17 +00:00
h264_sps_pps_tracker.cc
Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
2017-01-24 10:38:17 +00:00
h264_sps_pps_tracker.h
Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
2017-01-24 10:38:17 +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
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
2016-11-28 23:21:51 +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
…
inter_frame_delay.h
…
internal_defines.h
…
jitter_buffer_common.h
…
jitter_buffer_unittest.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +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
…
jitter_estimator.h
Wire up VCMTiming in FrameBuffer2.
2016-08-03 08:59:47 +00:00
media_opt_util.cc
Rename fec_tables_xor.h -> fec_rate_table.h
2017-01-12 14:16:24 +00:00
media_opt_util.h
Rename base/analytics/ to base/numerics/
2017-01-13 17:14:33 +00:00
media_optimization.cc
Remove unused arguments and variable in MediaOptimization.
2016-12-08 10:19:49 +00:00
media_optimization.h
Remove unused arguments and variable in MediaOptimization.
2016-12-08 10:19:49 +00:00
nack_fec_tables.h
…
nack_module_unittest.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
nack_module.cc
Remove |running_| state from NackModule to avoid running a tight loop.
2017-02-27 09:59:36 +00:00
nack_module.h
Remove |running_| state from NackModule to avoid running a tight loop.
2017-02-27 09:59:36 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
packet_buffer.cc
video_coding::PacketBuffer now group all H264 packets with the same timestamp into the same frame.
2017-02-02 16:51:29 +00:00
packet_buffer.h
Don't detect a new frame if a previous packet is used in a previous frame.
2017-01-11 10:01:56 +00:00
packet.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
packet.h
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +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
Advance picture id of keyframe if the stream has been continuous without a new keyframe for a while.
2017-02-20 15:53:23 +00:00
rtp_frame_reference_finder.cc
Advance picture id of keyframe if the stream has been continuous without a new keyframe for a while.
2017-02-20 15:53:23 +00:00
rtp_frame_reference_finder.h
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
rtt_filter.cc
…
rtt_filter.h
…
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
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
session_info.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +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
…
timestamp_map.h
…
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
Do not call OnDecoderTiming before timing values are set.
2016-12-22 09:26:18 +00:00
timing.h
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
video_codec_initializer_unittest.cc
Reland of Add experimental simulcast screen content mode
2017-01-26 14:12:26 +00:00
video_codec_initializer.cc
Minor style fix to please internal style tool.
2017-01-26 15:57:15 +00:00
video_coding_impl.cc
Create VideoReceiver with external VCMTiming object.
2016-12-15 15:11:01 +00:00
video_coding_impl.h
Create VideoReceiver with external VCMTiming object.
2016-12-15 15:11:01 +00:00
video_coding_robustness_unittest.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
video_packet_buffer_unittest.cc
Don't detect a new frame if a previous packet is used in a previous frame.
2017-01-11 10:01:56 +00:00
video_receiver_unittest.cc
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
video_receiver.cc
Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
2017-02-22 13:30:39 +00:00
video_sender_unittest.cc
Replace test::FrameGenerator::ChromaGenerator with new FrameGenerator::SquareGenerator The problem with the ChromaGenerator is that the VP8 encoder produce a too low bitrate for each frame. The squaregenerator make the VP8 encoder produce about 600kbit/s at VGA.
2017-02-27 14:52:10 +00:00
video_sender.cc
Periodically update channel parameters and send TargetBitrate message.
2017-01-16 14:24:02 +00:00