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webrtc_m130/media/engine
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Per K 9ece54fa73 Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions 
Bug: webrtc:7135, webrtc:14795
Change-Id: I0242a3600d4a156eae2315966e5e59e03be8aeab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290998
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39139}
2023-01-18 14:06:33 +00:00
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adm_helpers.cc
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adm_helpers.h
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encoder_simulcast_proxy_unittest.cc
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encoder_simulcast_proxy.cc
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encoder_simulcast_proxy.h
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fake_video_codec_factory.cc
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fake_video_codec_factory.h
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fake_webrtc_call.cc
Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions
2023-01-18 14:06:33 +00:00
fake_webrtc_call.h
Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions
2023-01-18 14:06:33 +00:00
fake_webrtc_video_engine.cc
Add API for querying codec support.
2023-01-04 10:04:46 +00:00
fake_webrtc_video_engine.h
Add API for querying codec support.
2023-01-04 10:04:46 +00:00
internal_decoder_factory_unittest.cc
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internal_decoder_factory.cc
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internal_decoder_factory.h
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internal_encoder_factory_unittest.cc
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internal_encoder_factory.cc
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internal_encoder_factory.h
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multiplex_codec_factory_unittest.cc
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multiplex_codec_factory.cc
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multiplex_codec_factory.h
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null_webrtc_video_engine_unittest.cc
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null_webrtc_video_engine.h
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payload_type_mapper_unittest.cc
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payload_type_mapper.cc
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payload_type_mapper.h
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simulcast_encoder_adapter_unittest.cc
Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t.
2022-12-12 06:02:59 +00:00
simulcast_encoder_adapter.cc
Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t.
2022-12-12 06:02:59 +00:00
simulcast_encoder_adapter.h
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webrtc_media_engine_defaults.cc
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webrtc_media_engine_defaults.h
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webrtc_media_engine_unittest.cc
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webrtc_media_engine.cc
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webrtc_media_engine.h
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webrtc_video_engine_unittest.cc
Add SVC fallback.
2023-01-11 16:49:49 +00:00
webrtc_video_engine.cc
replace use of iterators with for loops or auto
2023-01-18 13:55:30 +00:00
webrtc_video_engine.h
Use parsed packet from RtpTransport::DemuxPacket in engine and call
2023-01-10 15:06:50 +00:00
webrtc_voice_engine_unittest.cc
Use parsed packet from RtpTransport::DemuxPacket in engine and call
2023-01-10 15:06:50 +00:00
webrtc_voice_engine.cc
Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions
2023-01-18 14:06:33 +00:00
webrtc_voice_engine.h
Add RTCAudioPlayoutStats to GetStats().
2023-01-16 13:19:45 +00:00