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a8eb756a346682a4af66f9a9c0b87cfe1ed5b114
platform-external-webrtc/webrtc/modules/rtp_rtcp/source
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aleloi a8eb756a34 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. 
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.

Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.

transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.

NOTRY=True

BUG=webrtc:5589, webrtc:5878, webrtc:6785

Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 15:02:19 +00:00
..
rtcp_packet
Add RTCP packet class for signaling encoder target bitrate.
2016-11-01 09:50:17 +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
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
dtmf_queue.h
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +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_test_helper.cc
Add FlexfecPacketGenerator.
2016-10-04 06:47:11 +00:00
fec_test_helper.h
Add FlexfecPacketGenerator.
2016-10-04 06:47:11 +00:00
flexfec_header_reader_writer_unittest.cc
Update header formatters to FlexFEC draft 03.
2016-11-14 12:28:59 +00:00
flexfec_header_reader_writer.cc
Update header formatters to FlexFEC draft 03.
2016-11-14 12:28:59 +00:00
flexfec_header_reader_writer.h
Update header formatters to FlexFEC draft 03.
2016-11-14 12:28:59 +00:00
flexfec_receiver_unittest.cc
Make FlexfecReceiver a concrete class.
2016-11-03 15:18:33 +00:00
flexfec_receiver.cc
RtpPacket::payload() return rtc::ArrayView instead of raw pointer
2016-11-21 09:35:33 +00:00
flexfec_sender_unittest.cc
Make FlexFEC packets paceable through RTPSender.
2016-11-14 13:14:54 +00:00
flexfec_sender.cc
Wire up FlexfecSender in RTPSenderVideo.
2016-11-10 13:01:16 +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
Add FlexFEC header formatters.
2016-10-03 07:43:31 +00:00
forward_error_correction.h
Add FlexFEC header formatters.
2016-10-03 07:43:31 +00:00
nack_rtx_unittest.cc
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
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
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
Remove RtcpPacket dependency on rtcp_utility
2016-10-17 08:44:49 +00:00
rtcp_packet.h
Remove RtcpPacket dependency on rtcp_utility
2016-10-17 08:44:49 +00:00
rtcp_receiver_unittest.cc
Allow max 1 block per type in RTCP Extended Reports
2016-10-31 15:40:55 +00:00
rtcp_receiver.cc
Allow max 1 block per type in RTCP Extended Reports
2016-10-31 15:40:55 +00:00
rtcp_receiver.h
Use NtpTime in RtcpReceiver instead of pair of uints
2016-10-06 16:24:51 +00:00
rtcp_sender_unittest.cc
Use rtcp::Bye instead of RTCPUtility parser for rtcp_sender_unittest
2016-11-01 13:38:43 +00:00
rtcp_sender.cc
Allow max 1 block per type in RTCP Extended Reports
2016-10-31 15:40:55 +00:00
rtcp_sender.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +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
Finish reformatting RTCPUtility::RTCPParserV2::Validate()
2016-11-07 18:46:03 +00:00
rtcp_utility.h
RTCPReceiver store cname as std::string.
2016-09-27 14:05:39 +00:00
rtp_fec_unittest.cc
Add FlexFEC header formatters.
2016-10-03 07:43:31 +00:00
rtp_format_h264_unittest.cc
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
2016-11-12 16:54:50 +00:00
rtp_format_h264.cc
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
2016-11-12 16:54:50 +00:00
rtp_format_h264.h
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
2016-11-12 16:54:50 +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
Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
2016-11-12 16:54:50 +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
Simplify creating RtpHeaderExtensionMap in EventLogAnalyzer
2016-11-15 17:21:03 +00:00
rtp_header_extension.cc
Simplify creating RtpHeaderExtensionMap in EventLogAnalyzer
2016-11-15 17:21:03 +00:00
rtp_header_extension.h
Simplify creating RtpHeaderExtensionMap in EventLogAnalyzer
2016-11-15 17:21:03 +00:00
rtp_header_extensions.cc
Reland of Simplify and extend RtpHeaderExtensionMap (patchset #1 id:1 of https://codereview.webrtc.org/2484863007/ )
2016-11-09 13:46:45 +00:00
rtp_header_extensions.h
Reland of Simplify and extend RtpHeaderExtensionMap (patchset #1 id:1 of https://codereview.webrtc.org/2484863007/ )
2016-11-09 13:46:45 +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
RtpPacket::payload() return rtc::ArrayView instead of raw pointer
2016-11-21 09:35:33 +00:00
rtp_packet.cc
RtpPacket::payload() return rtc::ArrayView instead of raw pointer
2016-11-21 09:35:33 +00:00
rtp_packet.h
RtpPacket::payload() return rtc::ArrayView instead of raw pointer
2016-11-21 09:35:33 +00:00
rtp_payload_registry_unittest.cc
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
rtp_payload_registry.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
rtp_receiver_audio.cc
Send audio and video codecs to RTPPayloadRegistry
2016-11-24 17:34:53 +00:00
rtp_receiver_audio.h
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
rtp_receiver_impl.cc
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
rtp_receiver_impl.h
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
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
Send audio and video codecs to RTPPayloadRegistry
2016-11-24 17:34:53 +00:00
rtp_receiver_video.cc
Send audio and video codecs to RTPPayloadRegistry
2016-11-24 17:34:53 +00:00
rtp_receiver_video.h
Send audio and video codecs to RTPPayloadRegistry
2016-11-24 17:34:53 +00:00
rtp_rtcp_config.h
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
rtp_rtcp_impl_unittest.cc
Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 16:38:26 +00:00
rtp_rtcp_impl.cc
Add overhead per packet observer to the rtp_sender.
2016-11-17 09:38:48 +00:00
rtp_rtcp_impl.h
Make FlexFEC packets paceable through RTPSender.
2016-11-14 13:14:54 +00:00
rtp_sender_audio.cc
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
rtp_sender_audio.h
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
rtp_sender_unittest.cc
Cleanup RtpSender hiding RtpHeaderExtensionLength function.
2016-11-28 10:55:01 +00:00
rtp_sender_video.cc
RtpPacket::payload() return rtc::ArrayView instead of raw pointer
2016-11-21 09:35:33 +00:00
rtp_sender_video.h
Make FlexFEC packets paceable through RTPSender.
2016-11-14 13:14:54 +00:00
rtp_sender.cc
Cleanup RtpSender hiding RtpHeaderExtensionLength function.
2016-11-28 10:55:01 +00:00
rtp_sender.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
rtp_utility_unittest.cc
Fix parsing padding byte in rtp header extension
2016-11-24 17:06:10 +00:00
rtp_utility.cc
Fix parsing padding byte in rtp header extension
2016-11-24 17:06:10 +00:00
rtp_utility.h
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
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_generator_unittest.cc
Simplify SetFecParameters signature.
2016-11-07 11:36:14 +00:00
ulpfec_generator.cc
Simplify SetFecParameters signature.
2016-11-07 11:36:14 +00:00
ulpfec_generator.h
Fix forward declaration typo in UlpfecGenerator.
2016-11-08 08:29:06 +00:00
ulpfec_header_reader_writer_unittest.cc
Style fixes in FecReceiver and ProducerFec unit tests.
2016-10-03 12:11:31 +00:00
ulpfec_header_reader_writer.cc
Add FlexFEC header formatters.
2016-10-03 07:43:31 +00:00
ulpfec_header_reader_writer.h
Add FlexFEC header formatters.
2016-10-03 07:43:31 +00:00
ulpfec_receiver_impl.cc
Rename FecReceiver to UlpfecReceiver.
2016-10-31 11:51:38 +00:00
ulpfec_receiver_impl.h
Rename FecReceiver to UlpfecReceiver.
2016-10-31 11:51:38 +00:00
ulpfec_receiver_unittest.cc
Rename FecReceiver to UlpfecReceiver.
2016-10-31 11:51:38 +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