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3e9af7fe059af739d11bf8693669ff48d50efcfb
platform-external-webrtc/call
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Marina Ciocea 3e9af7fe05 Insert audio frame transformer between depacketizer and decoder. 
The frame transformer is passed from RTPReceiverInterface through the
library to be eventually set in ChannelReceive, where the frame
transformation will occur in the follow-up CL.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I5af06d1431047ef50d00e304cf95e92a832b4220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30956}
2020-04-01 08:15:53 +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
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
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
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
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
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
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
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
call_perf_tests.cc
Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
2020-01-07 13:02:52 +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
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
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
Injecting Clock in video receive.
2019-03-04 21:53:57 +00:00
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
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
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
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
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
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
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
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +00:00
rtp_demuxer.h
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +00:00
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
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
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
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
2019-01-11 17:11:39 +00:00
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
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +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
Insert frame transformer between Encoded and Packetizer.
2020-02-28 07:43:13 +00:00
rtp_video_sender.cc
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +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
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
ssrc_binding_observer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
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