webrtc/modules/rtp_rtcp/source
Niels Möller ab4a530b87 Delete telephone-event handling from RTPReceiverAudio.
Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24172}
2018-08-02 12:55:40 +00:00
..
rtcp_packet Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
byte_io.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
byte_io_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
dtmf_queue.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dtmf_queue.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fec_private_tables_bursty.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fec_private_tables_bursty.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
fec_private_tables_bursty_unittest.cc Fix a couple of nits and update a few comments in forward_error_correction_internal. 2018-04-23 14:29:17 +00:00
fec_private_tables_random.cc Remove part of the FEC code table that covers FEC code for group of 13-48 media packets, instead generate interleaved FEC code at run time. FEC code masks for protection of group of 1 - 12 media packets is not changed. 2018-04-18 14:35:17 +00:00
fec_private_tables_random.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
fec_test_helper.cc Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. 2018-07-10 11:57:46 +00:00
fec_test_helper.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
flexfec_header_reader_writer.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
flexfec_header_reader_writer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
flexfec_header_reader_writer_unittest.cc Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
flexfec_receiver.cc Avoid lifetime issues with FlexfecReceiver packet buffer. 2017-12-12 10:12:47 +00:00
flexfec_receiver_unittest.cc Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
flexfec_sender.cc Add MID sending to FlexfecSender 2018-04-10 16:08:35 +00:00
flexfec_sender_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
forward_error_correction.cc Revert "Fix buffer overflow in ulpfec recovery" 2018-07-16 12:31:57 +00:00
forward_error_correction.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
forward_error_correction_internal.cc Fix a downstream test failure. 2018-06-15 13:30:26 +00:00
forward_error_correction_internal.h Fix a couple of nits and update a few comments in forward_error_correction_internal. 2018-04-23 14:29:17 +00:00
nack_rtx_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
packet_loss_stats.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
packet_loss_stats.h Fix clang style errors in rtp_rtcp and dependant targets 2018-02-07 09:48:28 +00:00
packet_loss_stats_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
playout_delay_oracle.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
playout_delay_oracle.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
playout_delay_oracle_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
receive_statistics_impl.cc Remove StreamStatistician::IsPacketInOrder 2018-06-28 08:44:40 +00:00
receive_statistics_impl.h Remove StreamStatistician::IsPacketInOrder 2018-06-28 08:44:40 +00:00
receive_statistics_unittest.cc Update packetsLost and jitter stats any time a packet is received. 2018-06-25 23:56:39 +00:00
remote_ntp_time_estimator.cc Move timestamp_extrapolator.h to rtc_base/time/ 2018-03-22 14:36:44 +00:00
remote_ntp_time_estimator_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcp_nack_stats.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcp_nack_stats.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcp_nack_stats_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcp_packet.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcp_packet.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_packet_unittest.cc Change RtcpPacket::PacketReadyCallback to rtc::FunctionView 2017-12-07 11:20:08 +00:00
rtcp_receiver.cc Exposing video bitrate allocator into API 2018-07-23 21:23:21 +00:00
rtcp_receiver.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtcp_receiver_unittest.cc Exposing video bitrate allocator into API 2018-07-23 21:23:21 +00:00
rtcp_sender.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtcp_sender.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtcp_sender_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcp_transceiver.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtcp_transceiver.h Allow to turn RtcpTransciever on and off at runtime. 2018-03-06 15:19:11 +00:00
rtcp_transceiver_config.cc Calculate RTT using ExtendedReports in RtcpTransceiver 2017-11-30 14:34:40 +00:00
rtcp_transceiver_config.h Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets 2018-05-17 11:22:56 +00:00
rtcp_transceiver_impl.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtcp_transceiver_impl.h Replace rtc::Optional with absl::optional in modules/rtp_rtcp 2018-06-15 09:53:35 +00:00
rtcp_transceiver_impl_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtcp_transceiver_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtp_fec_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_format.cc Revert "Remove RTPVideoHeader::h264() accessors." 2018-07-16 21:36:12 +00:00
rtp_format.h Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. 2018-07-05 14:29:07 +00:00
rtp_format_h264.cc Revert "Remove RTPVideoHeader::h264() accessors." 2018-07-16 21:36:12 +00:00
rtp_format_h264.h Don't crash in SingleNalu packetization for h264 if no space in packet 2018-03-15 15:42:57 +00:00
rtp_format_h264_unittest.cc Revert "Remove RTPVideoHeader::h264() accessors." 2018-07-16 21:36:12 +00:00
rtp_format_video_generic.cc Add ParsedPayload::video_header() accessor. 2018-07-04 09:31:21 +00:00
rtp_format_video_generic.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_format_video_generic_unittest.cc Reland "Add stereo codec header and pass it through RTP" 2017-11-30 01:44:19 +00:00
rtp_format_vp8.cc Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. 2018-07-05 14:29:07 +00:00
rtp_format_vp8.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_format_vp8_test_helper.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_format_vp8_test_helper.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_format_vp8_unittest.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_format_vp9.cc Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. 2018-07-05 14:29:07 +00:00
rtp_format_vp9.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_format_vp9_unittest.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_generic_frame_descriptor.cc Discard frame self-dependency when parsing genric frame descriptor 2018-07-03 10:28:05 +00:00
rtp_generic_frame_descriptor.h Add Parsing/Building generic frame descriptor extension 2018-06-19 14:51:27 +00:00
rtp_generic_frame_descriptor_extension.cc Add Generic frame descritpor header extension 2018-06-29 15:02:44 +00:00
rtp_generic_frame_descriptor_extension.h Add Generic frame descritpor header extension 2018-06-29 15:02:44 +00:00
rtp_generic_frame_descriptor_extension_unittest.cc Add Parsing/Building generic frame descriptor extension 2018-06-19 14:51:27 +00:00
rtp_header_extension_map.cc Add Generic frame descritpor header extension 2018-06-29 15:02:44 +00:00
rtp_header_extension_map_unittest.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_header_extensions.cc Pass buffer with size when writing rtp header extension 2018-06-18 13:04:33 +00:00
rtp_header_extensions.h Pass buffer with size when writing rtp header extension 2018-06-18 13:04:33 +00:00
rtp_header_parser.cc Fix clang style errors in rtp_rtcp and dependant targets 2018-02-07 09:48:28 +00:00
rtp_packet.cc Fix infinite loop in rtp packet parsing 2018-02-13 14:42:45 +00:00
rtp_packet.h Pass buffer with size when writing rtp header extension 2018-06-18 13:04:33 +00:00
rtp_packet_history.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtp_packet_history.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_packet_history_unittest.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_packet_received.cc Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these. 2018-02-26 13:25:50 +00:00
rtp_packet_received.h Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these. 2018-02-26 13:25:50 +00:00
rtp_packet_to_send.cc Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these. 2018-02-26 13:25:50 +00:00
rtp_packet_to_send.h Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these. 2018-02-26 13:25:50 +00:00
rtp_packet_unittest.cc Pass buffer with size when writing rtp header extension 2018-06-18 13:04:33 +00:00
rtp_payload_registry.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_payload_registry_unittest.cc Delete enum RtpVideoCodecTypes, replaced with VideoCodecType. 2018-06-04 11:53:17 +00:00
rtp_receiver_audio.cc Delete telephone-event handling from RTPReceiverAudio. 2018-08-02 12:55:40 +00:00
rtp_receiver_audio.h Delete telephone-event handling from RTPReceiverAudio. 2018-08-02 12:55:40 +00:00
rtp_receiver_impl.cc Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. 2018-07-10 11:57:46 +00:00
rtp_receiver_impl.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_receiver_strategy.cc Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 2018-06-19 16:44:19 +00:00
rtp_receiver_strategy.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_receiver_unittest.cc Delete RtpFeedback. The ssrc for a receive stream should be known at 2018-05-28 11:05:19 +00:00
rtp_receiver_video.cc Add ParsedPayload::video_header() accessor. 2018-07-04 09:31:21 +00:00
rtp_receiver_video.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_rtcp_config.h Make RTCP report interval configurable 2018-02-01 10:12:11 +00:00
rtp_rtcp_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_rtcp_impl.h Replace rtc::Optional with absl::optional in modules/rtp_rtcp 2018-06-15 09:53:35 +00:00
rtp_rtcp_impl_unittest.cc Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. 2018-07-05 14:29:07 +00:00
rtp_sender.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtp_sender.h Replace rtc::Optional with absl::optional in modules/rtp_rtcp 2018-06-15 09:53:35 +00:00
rtp_sender_audio.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_sender_audio.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_sender_unittest.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_sender_video.cc Always disable RED when ULPFEC is disabled. 2018-07-12 13:01:20 +00:00
rtp_sender_video.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_utility.cc Add Generic frame descritpor header extension 2018-06-29 15:02:44 +00:00
rtp_utility.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rtp_utility_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_video_header.cc Add members for the codec agnostic descriptor to RTPVideoHeader. 2018-08-02 09:12:31 +00:00
rtp_video_header.h Add members for the codec agnostic descriptor to RTPVideoHeader. 2018-08-02 09:12:31 +00:00
time_util.cc Add TimeMicrosToNtp to calculate current NtpTime without Clock 2017-11-28 10:11:58 +00:00
time_util.h Delete assumption TimeMicrosToNtp can match RealTimeClock 2018-02-22 17:20:25 +00:00
time_util_unittest.cc Delete assumption TimeMicrosToNtp can match RealTimeClock 2018-02-22 17:20:25 +00:00
tmmbr_help.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
tmmbr_help.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
ulpfec_generator.cc Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
ulpfec_generator.h Fix clang style errors in rtp_rtcp and dependant targets 2018-02-07 09:48:28 +00:00
ulpfec_generator_unittest.cc Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
ulpfec_header_reader_writer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ulpfec_header_reader_writer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
ulpfec_header_reader_writer_unittest.cc Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
ulpfec_receiver_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
ulpfec_receiver_impl.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
ulpfec_receiver_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00