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webrtc_m130/webrtc/modules/video_coding
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johan 62d02c328d Unit test out of band H264 SPS,PPS within RtpStreamReceiver. 
This CL introduces a dedicated unit test for webrtc::RtpStreamReceiver.
Focus of this CL is testing RtpStreamReciver::OnReceivedPayloadData().
Dependencies with virtual interfaces are (g)mocked, non-virtual
dependencies are instantiated.

This CL is chained to https://codereview.webrtc.org/2638933002/ .

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2641463002
Cr-Commit-Position: refs/heads/master@{#16240}
2017-01-24 12:38:27 +00:00
..
codecs
Revert of Add experimental simulcast screen content mode (patchset #5 id:80001 of https://codereview.webrtc.org/2636443002/ )
2017-01-18 13:19:13 +00:00
include
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +00:00
test
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
utility
Rename base/analytics/ to base/numerics/
2017-01-13 17:14:33 +00:00
BUILD.gn
GN: Refactor modules_unittests to eliminate package boundary violations.
2017-01-19 16:27:11 +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
…
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
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_rate_table.h
Rename fec_tables_xor.h -> fec_rate_table.h
2017-01-12 14:16:24 +00:00
frame_buffer2_unittest.cc
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +00:00
frame_buffer2.cc
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +00:00
frame_buffer2.h
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +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
Set |_encodedWidth| and |_encodedHeight| in RtpFrameObject.
2017-01-12 11:26:04 +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
…
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
Decode h264 fmtp sprop-parameter-sets to binary.
2016-12-08 11:57:25 +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
…
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
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +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
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_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
Fix memory leak in video_coding::PacketBuffer::InsertPacket.
2016-11-30 09:32:11 +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
…
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
…
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 Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +00:00
video_codec_initializer.cc
Revert of Add experimental simulcast screen content mode (patchset #5 id:80001 of https://codereview.webrtc.org/2636443002/ )
2017-01-18 13:19:13 +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 Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +00:00
video_sender_unittest.cc
Periodically update channel parameters and send TargetBitrate message.
2017-01-16 14:24:02 +00:00
video_sender.cc
Periodically update channel parameters and send TargetBitrate message.
2017-01-16 14:24:02 +00:00