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webrtc_m130/webrtc/modules/rtp_rtcp/source
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Peter Boström a7d88d3844 Remove audio/video distinction for probe packets. 
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.

BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2061193002 .

Cr-Commit-Position: refs/heads/master@{#13210}
2016-06-20 08:51:20 +00:00
..
mock
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtcp_packet
[rtcp] ExtendedJitterReports::Parse updated not to use RTCPUtility
2016-06-02 12:07:00 +00:00
bitrate.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
bitrate.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
byte_io_unittest.cc
Fix ubsan warning in byteio_unittest
2016-02-26 13:13:51 +00:00
byte_io.h
Fix a bug in webrtc::ByteReader
2016-01-25 07:47:59 +00:00
CPPLINT.cfg
[rtp_rtcp] Lint errors cleaned from rtp_utility
2015-12-28 18:18:52 +00:00
dtmf_queue.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
dtmf_queue.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +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
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +00:00
fec_receiver_impl.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
fec_receiver_unittest.cc
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +00:00
fec_test_helper.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
fec_test_helper.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
forward_error_correction_internal.cc
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
forward_error_correction_internal.h
Add include guards to forward_error_correction_internal.h
2013-12-24 05:06:12 +00:00
forward_error_correction.cc
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +00:00
forward_error_correction.h
Replace RefCountImpl with rtc::RefCountedObject.
2016-03-21 15:44:41 +00:00
nack_rtx_unittest.cc
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
packet_loss_stats_unittest.cc
Add statistics gathering for packet loss.
2015-07-11 01:10:08 +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
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
playout_delay_oracle.cc
Remove thread_checker in playout_delay_oracle
2016-06-14 17:55:46 +00:00
playout_delay_oracle.h
Remove thread_checker in playout_delay_oracle
2016-06-14 17:55:46 +00:00
producer_fec_unittest.cc
Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ )
2016-05-02 00:19:13 +00:00
producer_fec.cc
Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ )
2016-05-02 00:19:13 +00:00
producer_fec.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
receive_statistics_impl.cc
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
receive_statistics_impl.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
receive_statistics_unittest.cc
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
remote_ntp_time_estimator_unittest.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
remote_ntp_time_estimator.cc
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +00:00
rtcp_format_remb_unittest.cc
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +00:00
rtcp_packet_unittest.cc
Remove "This file includes unit tests" comments.
2016-04-26 11:44:02 +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
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_receiver_help.cc
Fixed memleak when two voip blocks present in single rtcp packet.
2016-04-18 17:54:13 +00:00
rtcp_receiver_help.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
rtcp_receiver_unittest.cc
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +00:00
rtcp_receiver.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtcp_receiver.h
TMMBRHelp moved from member object/base class to stack object,
2016-05-24 20:25:35 +00:00
rtcp_sender_unittest.cc
_boundingSetToSend moved out of tmmbr_help_ into tmmbn_to_send_
2016-05-09 17:59:55 +00:00
rtcp_sender.cc
TMMBRHelp moved from member object/base class to stack object,
2016-05-24 20:25:35 +00:00
rtcp_sender.h
TMMBRHelp moved from member object/base class to stack object,
2016-05-24 20:25:35 +00:00
rtcp_utility_unittest.cc
Add a ParseHeader method to RtcpPacket, for parsing common RTCP header.
2015-09-14 19:50:49 +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
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
rtp_fec_unittest.cc
Remove unnecessary redefinition of PacketLists in rtp_fec_unittest.
2016-06-13 07:37:06 +00:00
rtp_format_h264_unittest.cc
Fix issue with parsing of incorrect (empty) Stap-A H264 NAL units.
2016-06-07 16:01:38 +00:00
rtp_format_h264.cc
Fix issue with parsing of incorrect (empty) Stap-A H264 NAL units.
2016-06-07 16:01:38 +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
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +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
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +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
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +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
Remove "This file includes unit tests" comments.
2016-04-26 11:44:02 +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
RtpPacket class introduced.
2016-04-20 12:25:19 +00:00
rtp_header_extensions.h
RtpPacket class introduced.
2016-04-20 12:25:19 +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
Remove "This file includes unit tests" comments.
2016-04-26 11:44:02 +00:00
rtp_packet_history.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_packet_history.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_packet_received.h
RtpPacket class introduced.
2016-04-20 12:25:19 +00:00
rtp_packet_to_send.h
RtpPacket class introduced.
2016-04-20 12:25:19 +00:00
rtp_packet_unittest.cc
Fix crash parsing malformed rtp packet
2016-06-17 11:18:58 +00:00
rtp_packet.cc
Fix crash parsing malformed rtp packet
2016-06-17 11:18:58 +00:00
rtp_packet.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
rtp_payload_registry_unittest.cc
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
2016-04-27 08:20:08 +00:00
rtp_payload_registry.cc
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
rtp_receiver_audio.cc
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_receiver_audio.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
rtp_receiver_impl.cc
Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
2016-05-11 11:05:13 +00:00
rtp_receiver_impl.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +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
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_receiver_video.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_receiver_video.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +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
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_rtcp_impl.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_rtcp_impl.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_sender_audio.cc
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
rtp_sender_audio.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_sender_unittest.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_sender_video.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_sender_video.h
Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
2016-04-14 10:05:37 +00:00
rtp_sender.cc
Remove audio/video distinction for probe packets.
2016-06-20 08:51:20 +00:00
rtp_sender.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
rtp_utility.cc
Fix buffer overflow parsing malformed rtp packet
2016-06-16 13:57:26 +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
rtt calculation handles time go backwards
2016-03-09 14:14:45 +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
TMMBRHelp moved from member object/base class to stack object,
2016-05-24 20:25:35 +00:00
tmmbr_help.h
TMMBRHelp moved from member object/base class to stack object,
2016-05-24 20:25:35 +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
in rtp_rtcp module:
2015-12-10 20:39:16 +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