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webrtc_m130/call
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Marina Ciocea d926cf63b5 [InsertableStreams] Fix simulcast: set frame transformer for all streams 
The transformer was previously moved into the config of the first stream
which resulted in incorrect behavior for simulcast. Use the transformer
in all the streams.

Bug: chromium:1065838
Change-Id: Iea340443da8cd4de32953bb24d3e6a07a275ae2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173026
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31018}
2020-04-07 18:51:18 +00:00
..
adaptation
[Adaptation] Refactor AdaptationTarget. Peek next restrictions.
2020-03-14 11:29:03 +00:00
test
Insert frame transformer between Encoded and Packetizer.
2020-02-28 07:43:13 +00:00
audio_receive_stream.cc
…
audio_receive_stream.h
Insert audio frame transformer between depacketizer and decoder.
2020-04-01 08:15:53 +00:00
audio_send_stream.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_send_stream.h
Insert audio frame transformer between encoder and packetizer.
2020-03-31 11:14:00 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_state.cc
…
audio_state.h
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
bitrate_allocator_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
bitrate_allocator.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
bitrate_allocator.h
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_estimator_tests.cc
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
2019-12-07 00:54:26 +00:00
BUILD.gn
Insert audio frame transformer between encoder and packetizer.
2020-03-31 11:14:00 +00:00
call_config.cc
…
call_config.h
Adds injectable trials from peerconnection down to transport controller.
2019-11-21 12:41:45 +00:00
call_factory.cc
DegradedCall: fake network using TaskQueue instead of ProcessThread
2019-08-06 15:05:30 +00:00
call_factory.h
…
call_perf_tests.cc
Preparation for ReceiveStatisticsProxy lock reduction.
2020-04-06 14:34:38 +00:00
call_unittest.cc
Trials should always be populated in call config.
2019-12-03 10:34:55 +00:00
call.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
call.h
Remove MediaTransport from Call.
2019-08-08 10:58:57 +00:00
degraded_call.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
degraded_call.h
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
DEPS
…
fake_network_pipe_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
fake_network_pipe.cc
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
fake_network_pipe.h
Make fake network degradation work also for sent audio
2019-08-12 15:20:18 +00:00
flexfec_receive_stream_impl.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
flexfec_receive_stream_impl.h
…
flexfec_receive_stream_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
flexfec_receive_stream.cc
…
flexfec_receive_stream.h
Format almost everything.
2019-07-08 13:45:15 +00:00
OWNERS
Add terelius as OWNER in call/
2020-03-23 09:55:34 +00:00
packet_receiver.h
…
rampup_tests.cc
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
rampup_tests.h
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
receive_time_calculator_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
receive_time_calculator.cc
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
receive_time_calculator.h
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtcp_demuxer.h
…
rtcp_packet_sink_interface.h
…
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_config.cc
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
rtp_config.h
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
rtp_demuxer_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
rtp_demuxer.cc
…
rtp_demuxer.h
…
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Ship GenericDescriptor00 by default.
2020-02-18 11:11:48 +00:00
rtp_payload_params.cc
Ship GenericDescriptor00 by default.
2020-02-18 11:11:48 +00:00
rtp_payload_params.h
Populate generic descriptor based on GenericFrameInfo when available.
2020-02-12 10:55:41 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_rtcp_demuxer_helper.cc
…
rtp_rtcp_demuxer_helper.h
…
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Concatenate string literals at compile time.
2020-01-14 14:47:48 +00:00
rtp_stream_receiver_controller.h
…
rtp_transport_controller_send_interface.h
Insert frame transformer between Encoded and Packetizer.
2020-02-28 07:43:13 +00:00
rtp_transport_controller_send.cc
Replace std::string::find() == 0 with absl::StartsWith (part 2).
2020-04-02 14:38:30 +00:00
rtp_transport_controller_send.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_video_sender_interface.h
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +00:00
rtp_video_sender_unittest.cc
Preparation for ReceiveStatisticsProxy lock reduction.
2020-04-06 14:34:38 +00:00
rtp_video_sender.cc
[InsertableStreams] Fix simulcast: set frame transformer for all streams
2020-04-07 18:51:18 +00:00
rtp_video_sender.h
Reland "Reland "Refactors UlpFec and FlexFec to use a common interface.""
2020-03-09 13:41:35 +00:00
rtx_receive_stream_unittest.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.cc
Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
2019-12-03 21:10:53 +00:00
rtx_receive_stream.h
…
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Adds UpdateConfig to SimulatedNetwork
2020-03-16 15:58:43 +00:00
simulated_network.h
Adds UpdateConfig to SimulatedNetwork
2020-03-16 15:58:43 +00:00
simulated_packet_receiver.h
…
ssrc_binding_observer.h
…
syncable.cc
…
syncable.h
Add periodic logging of sync delays.
2020-02-11 09:43:49 +00:00
video_receive_stream.cc
Add commas between codec parameters in VideoReceiveStream logging.
2020-03-09 02:45:34 +00:00
video_receive_stream.h
[InsertableStreams] Set video frame transformer if RTP stream already started.
2020-03-31 14:07:29 +00:00
video_send_stream.cc
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
video_send_stream.h
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00