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webrtc_m130/webrtc/modules/rtp_rtcp/source
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mflodman@webrtc.org 0abc6011b9 Remove SetCaptureDelay from the RTP module. 
This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34229004

Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 16:36:48 +00:00
..
mock
…
bitrate.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
bitrate.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
byte_io_unittest.cc
Make sure ByteReader<T>::Read* is properly constified.
2015-01-28 12:38:16 +00:00
byte_io.h
Make sure ByteReader<T>::Read* is properly constified.
2015-01-28 12:38:16 +00:00
dtmf_queue.cc
Remove potential deadlock in RTPSenderAudio.
2015-02-12 12:20:50 +00:00
dtmf_queue.h
…
fec_private_tables_bursty.h
…
fec_private_tables_random.h
…
fec_receiver_impl.cc
Add percentage of fec packets and recovered media packets to histogram stats:
2015-01-15 07:40:20 +00:00
fec_receiver_impl.h
Add percentage of fec packets and recovered media packets to histogram stats:
2015-01-15 07:40:20 +00:00
fec_receiver_unittest.cc
Add percentage of fec packets and recovered media packets to histogram stats:
2015-01-15 07:40:20 +00:00
fec_test_helper.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
fec_test_helper.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
forward_error_correction_internal.cc
…
forward_error_correction_internal.h
…
forward_error_correction.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
forward_error_correction.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
nack_rtx_unittest.cc
Allow rtp packet history to dynamically expand in size.
2015-01-29 09:09:41 +00:00
OWNERS
…
producer_fec_unittest.cc
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
2015-02-02 23:54:40 +00:00
producer_fec.cc
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
2015-02-02 23:54:40 +00:00
producer_fec.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
receive_statistics_impl.cc
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
receive_statistics_impl.h
Update StreamDataCounter with FEC bytes.
2015-01-27 12:17:29 +00:00
receive_statistics_unittest.cc
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
remote_ntp_time_estimator_unittest.cc
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
remote_ntp_time_estimator.cc
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
rtcp_format_remb_unittest.cc
Rtp-Rtcp sender cleanup.
2014-12-19 13:49:55 +00:00
rtcp_packet_unittest.cc
Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
2015-01-28 13:58:40 +00:00
rtcp_packet.cc
…
rtcp_packet.h
…
rtcp_receiver_help.cc
…
rtcp_receiver_help.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
rtcp_receiver_unittest.cc
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
rtcp_receiver.cc
Fix data race for RTCPReceiver stats callback.
2015-02-17 14:45:44 +00:00
rtcp_receiver.h
Fix data race for RTCPReceiver stats callback.
2015-02-17 14:45:44 +00:00
rtcp_sender_unittest.cc
Rtp-Rtcp sender cleanup.
2014-12-19 13:49:55 +00:00
rtcp_sender.cc
Remove SetCaptureDelay from the RTP module.
2015-02-17 16:36:48 +00:00
rtcp_sender.h
Remove SetCaptureDelay from the RTP module.
2015-02-17 16:36:48 +00:00
rtcp_utility_unittest.cc
Add stats for duplicate sent and received NACK requests.
2014-10-29 12:42:30 +00:00
rtcp_utility.cc
Add stats for duplicate sent and received NACK requests.
2014-10-29 12:42:30 +00:00
rtcp_utility.h
Add stats for duplicate sent and received NACK requests.
2014-10-29 12:42:30 +00:00
rtp_fec_unittest.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_format_h264_unittest.cc
Remove partially defined WebRtcRTPHeader from Parse().
2014-11-07 11:02:12 +00:00
rtp_format_h264.cc
Remove partially defined WebRtcRTPHeader from Parse().
2014-11-07 11:02:12 +00:00
rtp_format_h264.h
…
rtp_format_video_generic.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_format_video_generic.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +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
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_format_vp8_unittest.cc
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
rtp_format_vp8.cc
Set simulcastIdx field to zero even if it has no meaning.
2014-12-01 15:29:29 +00:00
rtp_format_vp8.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_format.cc
…
rtp_format.h
Remove partially defined WebRtcRTPHeader from Parse().
2014-11-07 11:02:12 +00:00
rtp_header_extension_unittest.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_header_extension.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_header_extension.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_header_parser.cc
…
rtp_packet_history_unittest.cc
Fixed potential crash if rtp packet history is completely full.
2015-02-02 13:08:14 +00:00
rtp_packet_history.cc
Fixed potential crash if rtp packet history is completely full.
2015-02-02 13:08:14 +00:00
rtp_packet_history.h
Fixed potential crash if rtp packet history is completely full.
2015-02-02 13:08:14 +00:00
rtp_payload_registry_unittest.cc
…
rtp_payload_registry.cc
Using << on an int8_t or uint8_t will output a character rather than a number.
2015-01-30 19:54:19 +00:00
rtp_receiver_audio.cc
Set webrtc_rtp category to be disabled by default.
2015-02-16 12:06:48 +00:00
rtp_receiver_audio.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_receiver_impl.cc
Using << on an int8_t or uint8_t will output a character rather than a number.
2015-01-30 19:54:19 +00:00
rtp_receiver_impl.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_receiver_strategy.cc
…
rtp_receiver_strategy.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_receiver_video.cc
Set webrtc_rtp category to be disabled by default.
2015-02-16 12:06:48 +00:00
rtp_receiver_video.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_rtcp_config.h
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
rtp_rtcp_impl_unittest.cc
Allow rtp packet history to dynamically expand in size.
2015-01-29 09:09:41 +00:00
rtp_rtcp_impl.cc
Remove SetCaptureDelay from the RTP module.
2015-02-17 16:36:48 +00:00
rtp_rtcp_impl.h
Remove SetCaptureDelay from the RTP module.
2015-02-17 16:36:48 +00:00
rtp_rtcp.gypi
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
rtp_sender_audio.cc
Set webrtc_rtp category to be disabled by default.
2015-02-16 12:06:48 +00:00
rtp_sender_audio.h
Remove potential deadlock in RTPSenderAudio.
2015-02-12 12:20:50 +00:00
rtp_sender_unittest.cc
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
rtp_sender_video.cc
Set webrtc_rtp category to be disabled by default.
2015-02-16 12:06:48 +00:00
rtp_sender_video.h
Remove potential deadlock in RTPSenderAudio.
2015-02-12 12:20:50 +00:00
rtp_sender.cc
Set webrtc_rtp category to be disabled by default.
2015-02-16 12:06:48 +00:00
rtp_sender.h
Remove potential deadlock in RTPSenderAudio.
2015-02-12 12:20:50 +00:00
rtp_utility.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_utility.h
…
ssrc_database.cc
…
ssrc_database.h
…
tmmbr_help.cc
…
tmmbr_help.h
…
video_codec_information.h
…
vp8_partition_aggregator_unittest.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
vp8_partition_aggregator.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
vp8_partition_aggregator.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00