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webrtc_m130/webrtc/modules/rtp_rtcp/source
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danilchap 7411061982 Use RtpPacketToSend in RtpSenderVideo. 
This reduce reparsing rtp packet while sending.

BUG=webrtc:5261

Review-Url: https://codereview.webrtc.org/2217383002
Cr-Commit-Position: refs/heads/master@{#14465}
2016-10-02 17:54:52 +00:00
..
mock
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
rtcp_packet
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
byte_io_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
byte_io.h
Fix a bug in webrtc::ByteReader
2016-01-25 07:47:59 +00:00
CPPLINT.cfg
Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver
2016-09-23 17:36:12 +00:00
dtmf_queue.cc
- Make RtpSenderAudio not inherit from DtmfQueue.
2016-09-23 20:10:50 +00:00
dtmf_queue.h
- Make RtpSenderAudio not inherit from DtmfQueue.
2016-09-23 20:10:50 +00:00
fec_private_tables_bursty.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
fec_private_tables_random.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
fec_receiver_impl.cc
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
fec_receiver_impl.h
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
fec_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
fec_test_helper.cc
Generalize FEC unit tests and rename GenerateFec.
2016-09-21 06:16:36 +00:00
fec_test_helper.h
Generalize FEC unit tests and rename GenerateFec.
2016-09-21 06:16:36 +00:00
forward_error_correction_internal.cc
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
forward_error_correction_internal.h
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
forward_error_correction.cc
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
forward_error_correction.h
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
nack_rtx_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
OWNERS
…
packet_loss_stats_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
packet_loss_stats.cc
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
packet_loss_stats.h
Add statistics gathering for packet loss.
2015-07-11 01:10:08 +00:00
playout_delay_oracle_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
playout_delay_oracle.cc
Merge min_ms and max_ms accessors in PlayoutDelayOracle
2016-09-07 11:30:00 +00:00
playout_delay_oracle.h
Merge min_ms and max_ms accessors in PlayoutDelayOracle
2016-09-07 11:30:00 +00:00
producer_fec_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
producer_fec.cc
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
producer_fec.h
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
receive_statistics_impl.cc
Fixig issues in BWE dynamics plot scripts.
2016-09-30 13:29:57 +00:00
receive_statistics_impl.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
receive_statistics_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
remote_ntp_time_estimator_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
remote_ntp_time_estimator.cc
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +00:00
rtcp_packet_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
rtcp_packet.cc
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_packet.h
Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ )
2016-09-27 16:27:52 +00:00
rtcp_receiver_help.cc
Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
2016-09-29 22:28:12 +00:00
rtcp_receiver_help.h
Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
2016-09-29 22:28:12 +00:00
rtcp_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
rtcp_receiver.cc
Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
2016-09-29 22:28:12 +00:00
rtcp_receiver.h
Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
2016-09-29 22:28:12 +00:00
rtcp_sender_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
rtcp_sender.cc
Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
2016-09-28 09:54:30 +00:00
rtcp_sender.h
Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
2016-09-28 09:54:30 +00:00
rtcp_utility_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
rtcp_utility.cc
Fixed undefined shift in parsing Tmmbr, Tmmbn and Remb
2016-04-19 10:15:21 +00:00
rtcp_utility.h
RTCPReceiver store cname as std::string.
2016-09-27 14:05:39 +00:00
rtp_fec_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
rtp_format_h264_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
rtp_format_h264.cc
Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )
2016-09-30 16:06:43 +00:00
rtp_format_h264.h
Add H264 bitstream rewriting to limit frame reordering marker in header
2016-06-02 09:43:38 +00:00
rtp_format_video_generic.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_video_generic.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
rtp_format_vp8_test_helper.cc
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
rtp_format_vp8_test_helper.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_vp8_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
rtp_format_vp8.cc
Use webrtc/base/logging.h for rtp_rtcp.
2015-10-28 15:39:43 +00:00
rtp_format_vp8.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_vp9_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
rtp_format_vp9.cc
Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
2015-11-10 15:17:26 +00:00
rtp_format_vp9.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format.cc
Add support for VP9 packetization/depacketization.
2015-07-28 11:02:58 +00:00
rtp_format.h
Add H264 bitstream rewriting to limit frame reordering marker in header
2016-06-02 09:43:38 +00:00
rtp_header_extension_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
rtp_header_extension.cc
RtpPacket class introduced.
2016-04-20 12:25:19 +00:00
rtp_header_extension.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_header_extensions.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
rtp_header_extensions.h
Implement PlayoutDelay extension as a trait
2016-09-07 13:08:20 +00:00
rtp_header_parser.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_packet_history_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
rtp_packet_history.cc
RtpPacketHistory rewritten to use RtpPacket class.
2016-08-03 16:27:50 +00:00
rtp_packet_history.h
RtpPacketHistory rewritten to use RtpPacket class.
2016-08-03 16:27:50 +00:00
rtp_packet_received.h
RtpPacket class introduced.
2016-04-20 12:25:19 +00:00
rtp_packet_to_send.h
RtpPacketHistory rewritten to use RtpPacket class.
2016-08-03 16:27:50 +00:00
rtp_packet_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
rtp_packet.cc
Use RtpPacketToSend in RtpSenderVideo.
2016-10-02 17:54:52 +00:00
rtp_packet.h
Introduce helpers to RtpSender to propagate RtpPacketToSend.
2016-09-02 17:16:08 +00:00
rtp_payload_registry_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
rtp_payload_registry.cc
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
rtp_receiver_audio.cc
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
rtp_receiver_audio.h
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
rtp_receiver_impl.cc
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
rtp_receiver_impl.h
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
rtp_receiver_strategy.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_receiver_strategy.h
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
rtp_receiver_video.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
rtp_receiver_video.h
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
rtp_rtcp_config.h
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00
rtp_rtcp_impl_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
rtp_rtcp_impl.cc
Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
2016-09-28 09:54:30 +00:00
rtp_rtcp_impl.h
Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
2016-09-28 09:54:30 +00:00
rtp_sender_audio.cc
- Make RtpSenderAudio not inherit from DtmfQueue.
2016-09-23 20:10:50 +00:00
rtp_sender_audio.h
- Make RtpSenderAudio not inherit from DtmfQueue.
2016-09-23 20:10:50 +00:00
rtp_sender_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
rtp_sender_video.cc
Use RtpPacketToSend in RtpSenderVideo.
2016-10-02 17:54:52 +00:00
rtp_sender_video.h
Use RtpPacketToSend in RtpSenderVideo.
2016-10-02 17:54:52 +00:00
rtp_sender.cc
Use RtpPacketToSend in RtpSenderVideo.
2016-10-02 17:54:52 +00:00
rtp_sender.h
Deprecate RtpSender::SendPadData with provided timestamps.
2016-09-19 12:38:05 +00:00
rtp_utility.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
rtp_utility.h
Remove deprecated functions in rtp_rtcp module
2016-04-29 10:01:33 +00:00
ssrc_database.cc
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
ssrc_database.h
Refactor RtpSender and SSRCDatabase.
2016-02-02 16:34:16 +00:00
time_util_unittest.cc
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
time_util.cc
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
time_util.h
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
tmmbr_help.cc
Remove TMMBRSet class
2016-08-18 13:17:48 +00:00
tmmbr_help.h
Remove TMMBRSet class
2016-08-18 13:17:48 +00:00
ulpfec_header_reader_writer_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
ulpfec_header_reader_writer.cc
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
ulpfec_header_reader_writer.h
Generalize FEC header formatting.
2016-09-21 07:19:42 +00:00
video_codec_information.h
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00
vp8_partition_aggregator_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
vp8_partition_aggregator.cc
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
vp8_partition_aggregator.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00