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platform-external-webrtc/call
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Sami Kalliomäki 22c7d69d41 Enable ULPFEC for kVideoCodecGeneric if GenericPictureId is enabled. 
Enable ULPFEC for kVideoCodecGeneric if GenericPictureId field trial
is enabled. GenericPictureId field trial allows kVideoCodecGeneric
to skip FEC packets.

Bug: webrtc:9516
Change-Id: I8de1d947d213dd5c42d7be9d26b9bdfac00ab8cd
Reviewed-on: https://webrtc-review.googlesource.com/97400
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24543}
2018-09-04 08:17:58 +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_unittest.cc
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_estimator_tests.cc
Deprecate two DirectTransport ctors and remove their direct usage.
2018-08-20 11:47:28 +00:00
BUILD.gn
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +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
Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
2018-08-22 11:12:40 +00:00
call_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call.cc
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
call.h
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +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
Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe
2018-08-29 11:05:19 +00:00
flexfec_receive_stream_impl.cc
Delete StreamStatistician::IsRetransmitOfOldPacket
2018-08-30 11:00:13 +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
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
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
Allow send bitrate < start bitrate in RampUpTest.
2018-09-03 07:28:39 +00:00
rampup_tests.h
Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
2018-08-22 11:12:40 +00:00
receive_time_calculator_unittest.cc
Added receive time calculator under field trial.
2018-03-21 15:40:39 +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
rtcp_demuxer_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_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_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_unittest.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +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_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc
Reland "Add spatial index to EncodedImage."
2018-08-31 07:35:52 +00:00
rtp_payload_params.cc
Reland "Add spatial index to EncodedImage."
2018-08-31 07:35:52 +00:00
rtp_payload_params.h
Convert VP8 descriptor to generic descriptor.
2018-08-27 13:03:20 +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_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_stream_receiver_controller_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_transport_controller_send_interface.h
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +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_video_sender_interface.h
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_video_sender_unittest.cc
Reland "Add spatial index to EncodedImage."
2018-08-31 07:35:52 +00:00
rtp_video_sender.cc
Enable ULPFEC for kVideoCodecGeneric if GenericPictureId is enabled.
2018-09-04 08:17:58 +00:00
rtp_video_sender.h
Add shared frame id state to RtpVideoSender.
2018-08-08 15:28:20 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc
Delete StreamStatistician::IsRetransmitOfOldPacket
2018-08-30 11:00:13 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Delete pre_decode_callback.
2018-06-20 07:04:09 +00:00
video_receive_stream.h
Delete pre_decode_callback.
2018-06-20 07:04:09 +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