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ccd4b2aec88c79c531254fd31611ec741c77738f
platform-external-webrtc/webrtc/modules/rtp_rtcp/source
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stefan@webrtc.org ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. 
BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 15:58:23 +00:00
..
H264
Removing remaining WebRtc_Word32 not in typedefs.h
2013-04-10 17:59:17 +00:00
mock
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
Android.mk
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
bitrate.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
bitrate.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
dtmf_queue.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
dtmf_queue.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
fec_private_tables_bursty.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
fec_private_tables_random.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
fec_test_helper.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
fec_test_helper.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
forward_error_correction_internal.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
forward_error_correction_internal.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
forward_error_correction.cc
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
2013-04-11 17:48:02 +00:00
forward_error_correction.h
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
2013-04-11 17:48:02 +00:00
nack_rtx_unittest.cc
Adding a payload type for RTX.
2013-04-12 14:55:46 +00:00
producer_fec_unittest.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
producer_fec.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
producer_fec.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
receiver_fec_unittest.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
receiver_fec.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
receiver_fec.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_format_remb_unittest.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_receiver_help.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_receiver_help.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_receiver_unittest.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_receiver.cc
Add a default RTT to CallStats and use different values for buffered/real-time mode.
2013-04-23 15:58:23 +00:00
rtcp_receiver.h
Add a default RTT to CallStats and use different values for buffered/real-time mode.
2013-04-23 15:58:23 +00:00
rtcp_sender_unittest.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_sender.cc
Change capture interface to use NTP capture time.
2013-04-12 11:56:23 +00:00
rtcp_sender.h
Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps."
2013-04-09 14:56:29 +00:00
rtcp_utility.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtcp_utility.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_fec_unittest.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtp_format_video_generic.h
Generic video-codec support.
2013-03-18 16:39:03 +00:00
rtp_format_vp8_test_helper.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_format_vp8_test_helper.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtp_format_vp8_unittest.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtp_format_vp8.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_format_vp8.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_header_extension_unittest.cc
Make RtpHeaderExtensionMap::Register and ::Deregister idempotent.
2013-03-01 17:03:02 +00:00
rtp_header_extension.cc
Make RtpHeaderExtensionMap::Register and ::Deregister idempotent.
2013-03-01 17:03:02 +00:00
rtp_header_extension.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtp_packet_history_unittest.cc
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
2013-01-21 07:42:11 +00:00
rtp_packet_history.cc
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
2013-01-21 07:42:11 +00:00
rtp_packet_history.h
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
2013-01-21 07:42:11 +00:00
rtp_payload_registry_unittest.cc
Permit arbitrary payload names for kVideoCodecGeneric.
2013-04-05 13:27:38 +00:00
rtp_payload_registry.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_payload_registry.h
Adding a payload type for RTX.
2013-04-12 14:55:46 +00:00
rtp_receiver_audio.cc
More trace events
2013-04-09 19:54:10 +00:00
rtp_receiver_audio.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_receiver_strategy.cc
Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.
2013-01-14 10:01:55 +00:00
rtp_receiver_strategy.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_receiver_video.cc
More trace events
2013-04-09 19:54:10 +00:00
rtp_receiver_video.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_receiver.cc
Add a default RTT to CallStats and use different values for buffered/real-time mode.
2013-04-23 15:58:23 +00:00
rtp_receiver.h
Add a default RTT to CallStats and use different values for buffered/real-time mode.
2013-04-23 15:58:23 +00:00
rtp_rtcp_config.h
Add a default RTT to CallStats and use different values for buffered/real-time mode.
2013-04-23 15:58:23 +00:00
rtp_rtcp_impl.cc
Add a default RTT to CallStats and use different values for buffered/real-time mode.
2013-04-23 15:58:23 +00:00
rtp_rtcp_impl.h
Adding a payload type for RTX.
2013-04-12 14:55:46 +00:00
rtp_rtcp_tests.gypi
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
rtp_rtcp.gypi
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
rtp_sender_audio.cc
More trace events
2013-04-09 19:54:10 +00:00
rtp_sender_audio.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_sender_unittest.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_sender_video.cc
More trace events
2013-04-09 19:54:10 +00:00
rtp_sender_video.h
More trace events
2013-04-09 19:54:10 +00:00
rtp_sender.cc
Adding a payload type for RTX.
2013-04-12 14:55:46 +00:00
rtp_sender.h
Adding a payload type for RTX.
2013-04-12 14:55:46 +00:00
rtp_utility_unittest.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
rtp_utility.cc
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
2013-04-11 17:48:02 +00:00
rtp_utility.h
Change capture interface to use NTP capture time.
2013-04-12 11:56:23 +00:00
ssrc_database.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
ssrc_database.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
tmmbr_help.cc
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
tmmbr_help.h
WebRtc_Word32 => int32_t for rtp_rtcp/
2013-04-08 11:08:41 +00:00
video_codec_information.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
vp8_partition_aggregator_unittest.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
vp8_partition_aggregator.cc
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
vp8_partition_aggregator.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00