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Niels Möller 1f3206cca4 Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.

Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
2018-09-28 12:00:28 +00:00
..
test Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe 2018-08-29 11:05:19 +00:00
audio_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_send_stream.cc replace stringstream in call/ 2018-04-04 16:09:15 +00:00
audio_send_stream.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_state.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_state.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_allocator.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_allocator.h Refactoring PayloadRouter. 2018-07-17 14:46:15 +00:00
bitrate_allocator_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bitrate_estimator_tests.cc Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
BUILD.gn Remove deprecated field_trial_default and metrics_default. 2018-09-28 07:21:07 +00:00
call.cc Use SdpVideoFormat in VideoReceiveStream::Decoder 2018-09-11 15:03:04 +00:00
call.h Delete struct webrtc::PacketTime. 2018-08-07 10:07:15 +00:00
call_config.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
call_config.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
call_perf_tests.cc Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
call_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
callfactory.cc Remove temporary SetConfig method from NetworkSimulatioInterface. 2018-08-23 10:08:35 +00:00
callfactory.h Rename Call::Config to CallConfig, keep old name as alias. 2018-02-14 15:14:39 +00:00
degraded_call.cc Switch webrtc users from deprecated ctors. 2018-08-17 13:54:51 +00:00
degraded_call.h Switch webrtc users from deprecated ctors. 2018-08-17 13:54:51 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe 2018-08-29 11:05:19 +00:00
fake_network_pipe.h Cleanup: remove deprecated class shortcuts. 2018-09-25 09:06:47 +00:00
flexfec_receive_stream.cc Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
flexfec_receive_stream.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
flexfec_receive_stream_impl.cc Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 2018-09-28 12:00:28 +00:00
flexfec_receive_stream_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
OWNERS Makes srte owner in call/ and test/ 2018-07-13 08:39:41 +00:00
packet_receiver.h Delete webrtc::PacketTime and backwards compatibility. 2018-08-17 15:14:03 +00:00
rampup_tests.cc Use SdpVideoFormat in VideoReceiveStream::Decoder 2018-09-11 15:03:04 +00:00
rampup_tests.h Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe 2018-08-22 11:12:40 +00:00
receive_time_calculator.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
receive_time_calculator.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
receive_time_calculator_unittest.cc Added receive time calculator under field trial. 2018-03-21 15:40:39 +00:00
rtcp_demuxer.cc Allow all "token" chars from RFC 4566 when checking for legal mid names. 2018-08-01 18:20:42 +00:00
rtcp_demuxer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_demuxer_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtcp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_bitrate_configurator.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
rtp_bitrate_configurator_unittest.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_config.cc Refactoring PayloadRouter. 2018-07-17 14:46:15 +00:00
rtp_config.h Add shared frame id state to RtpVideoSender. 2018-08-08 15:28:20 +00:00
rtp_demuxer.cc Allow all "token" chars from RFC 4566 when checking for legal mid names. 2018-08-01 18:20:42 +00:00
rtp_demuxer.h Don't check MIDs when demuxing RTP packets in Call 2018-03-29 20:36:08 +00:00
rtp_demuxer_unittest.cc Allow all "token" chars from RFC 4566 when checking for legal mid names. 2018-08-01 18:20:42 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_payload_params.cc Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." 2018-09-26 10:26:43 +00:00
rtp_payload_params.h Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." 2018-09-26 10:26:43 +00:00
rtp_payload_params_unittest.cc Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor." 2018-09-26 10:26:43 +00:00
rtp_rtcp_demuxer_helper.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper_unittest.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_stream_receiver_controller.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
rtp_stream_receiver_controller.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_transport_controller_send.cc Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_transport_controller_send.h Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_transport_controller_send_interface.h Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_video_sender.cc Disable ulpfec when field trial flag is present 2018-09-25 16:07:59 +00:00
rtp_video_sender.h Add shared frame id state to RtpVideoSender. 2018-08-08 15:28:20 +00:00
rtp_video_sender_interface.h Rename PayloadRouter to RtpVideoSender. 2018-07-19 08:50:50 +00:00
rtp_video_sender_unittest.cc Makes new congestion controller work with rtp sender tests. 2018-09-27 15:58:32 +00:00
rtx_receive_stream.cc Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 2018-09-28 12:00:28 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtx_receive_stream_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
simulated_network.cc Move SimulatedNetwork class to separate file. 2018-08-08 09:29:53 +00:00
simulated_network.h Remove temporary SetConfig method from NetworkSimulatioInterface. 2018-08-23 10:08:35 +00:00
simulated_packet_receiver.h Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe 2018-08-22 11:12:40 +00:00
ssrc_binding_observer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
video_receive_stream.cc Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
video_receive_stream.h Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config." 2018-09-28 08:48:02 +00:00
video_send_stream.cc New api function CreateVideoStreamEncoder. 2018-07-24 09:14:26 +00:00
video_send_stream.h Delete VideoSendStream::EnableEncodedFrameRecording. 2018-09-03 13:06:32 +00:00